Cisco Cucm B Feature Configuration Guide For Cisco1251SU1

Feature Configuration Guide for Cisco Unified Communications
Manager, Release 12.5(1)SU1

First Published: 2019-06-19
Last Modified: 2023-12-01
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CONTENTS
PART I Getting Started 43
CHAPTER 1 Feature Configuration Overview 1
About the Feature Configuration Guide 1

Generate a Phone Feature List 1
CHAPTER 2 Configuration Tools 3
About the Feature Configuration Guide 3

Configuration Tools Overview 3
Cisco Unified Communications Manager Administration 3
Log In to Cisco Unified CM Administration 4
Cisco Unified Communications Manager Serviceability 4

Log into Cisco Unified Communications Manager Serviceability 5
PART II Remote Worker Features 7
CHAPTER 3 Cisco Unified Mobility 9
Cisco Unified Mobility Overview 9

Mobility Features 10
Cisco Unified Mobility Prerequisites 11
Cisco Unified Mobility Configuration Task Flow 12
Configure a Mobility User 13
Configure Mobility Users Through Bulk Administration 14
Provision Mobility Users Through LDAP 14
Configure Mobility for IP Phones 15
Feature Configuration Guide for Cisco Unified Communications Manager, Release 12.5(1)SU1
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Configure Softkey Template for Mobility 16

Enable Mobility Within Feature Control Policy 17
Configure IP Phone for Mobility 18
Configure a Remote Destination Profile 18
Configure a Remote Destination 19
Configure an Access List 20
Configure Mobile Voice Access 20
Activate the Cisco Unified Mobile Voice Access Service 22
Enable Mobile Voice Access 22
Configure Directory Number for Mobile Voice Access 23
Restart Cisco CallManager Service 23
Configure an Existing H.323 or SIP Gateway for Remote Access 24
Configure a New H.323 Gateway for Remote Access 25
Configure Enterprise Feature Access 27
Configure Intelligent Session Control 28
Configure Mobility Service Parameters 29
Configure Cisco Jabber Dual-Mode 29
Configure Other Dual-Mode Devices 30
Configure a Mobility Profile 31
Add a Dual-Mode Device for Cisco Jabber 31
Dual-Mode Device Configuration Fields 32
Add Other Dual-Mode Device 33
Configure a Mobility Identity 34
Configure Handoff Number 34
Cisco Unified Mobility Call Flow 35
FMC Over SIP Trunks Without Smart Client 35
Hunt Group Login and Logout for Carrier-Integrated Mobile Devices 36
Cisco Unified Mobility Interactions 37
Cisco Unified Mobility Restrictions 38
Cisco Unified Mobility Troubleshooting 42
Cannot Resume Call on Desktop Phone 42
CHAPTER 4 Device Mobility 43
Device Mobility Overview 43
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Device Pool Assignment 45

Device Mobility Groups Operations Summary 46
Device Mobility Prerequisites 47
Device Mobility Configuration Task Flow 48
Enable Device Mobility Clusterwide 48
Enable Device Mobility for Individual Devices 49
Configure a Physical Location 49
Configure a Device Mobility Group 50
Configure a Device Pool for Device Mobility 50
Configure Device Mobility Information 51
View Roaming Device Pool Parameters 51
Device Mobility Interactions 52
Device Mobility Restrictions 53
CHAPTER 5 Extend and Connect 55

Extend and Connect Overview 55
Extend and Connect Prerequisites 56
Extend and Connect Configuration Task Flow 56
Configure User Account 56
Add User Permissions 57
Create CTI Remote Devices 58
Add Directory Number to a Device 58
Add Remote Destination 59
Verify Remote Destination 60
Associate User with Device 60
CTI Remote Device (CTIRD) Call Flows 61
Extend and Connect Interactions 62
Extend and Connect Restrictions 63
CHAPTER 6 Remote Worker Emergency Calling 65
Remote Worker Emergency Calling Overview 65

Remote Worker Emergency Calling Prerequisites 65
Remote Worker Emergency Calling Configuration Task Flow 66
Configure User As a Remote Worker 66
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Specify Alternate Routing for Emergency Calling 67

Configure the Application Server 67
Configure E911 Messages 67
CHAPTER 7 Configure Mobile and Remote Access 69
Mobile and Remote Access Overview 69

Mobile and Remote Access Prerequisites 71
Mobile and Remote Access Configuration Task Flow 72
Activate Cisco AXL Web Service 73
Configure Maximum Session BitRate for Video 73
Configure a Device Pool for Mobile and Remote Access 74
Configure ICE 74
Configure Phone Security Profile for Mobile and Remote Access 75
Configure Mobile and Remote Access Access Policy for Cisco Jabber Users 76
Configure Users for Mobile and Remote Access 77
Configure Endpoints for Mobile and Remote Access 77
Configure Cisco Expressway for Mobile and Remote Access 77
PART III Remote Network Access 79
CHAPTER 8 Wireless LAN 81
Wireless LAN Overview 81

Wireless LAN Configuration Task Flow 81
Configure a Network Access Profile 82
Configure a Wireless LAN Profile 82
Configure a Wireless LAN Profile Group 82
Link a Wireless LAN Profile Group to a Device or Device Pool 83
Link a Wireless LAN Profile Group to a Device 83
Link a Wireless LAN Profile Group to a Device Pool 83
CHAPTER 9 VPN Client 85
VPN Client Overview 85

VPN Client Prerequisites 85
VPN Client Configuration Task Flow 85
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Complete Cisco IOS Prerequisites 87

Configure Cisco IOS SSL VPN to Support IP Phones 87
Complete ASA Prerequisites for AnyConnect 89

Configure ASA for VPN Client on IP Phone 89
Upload VPN Concentrator Certificates 91
Configure VPN Gateway 92
VPN Gateway Fields for VPN Client 92
Configure VPN Group 93
VPN Group Fields for VPN Client 93
Configure VPN Profile 94
VPN Profile Fields for VPN Client 94
Configure VPN Feature Parameters 95
VPN Feature Parameters 96
Add VPN Details to Common Phone Profile 97
PART IV Licensing 99
CHAPTER 10 Licensing 101
Licensing 101
Unified Communications Manager Licensing 102
License Compliance 103
User Only Licensing 104
Device Only 104
User and Device 104
Maximum Number of Devices Per User 111
TelePresence Room License 111
License Substitution 111
Licensing Scenarios 112
Adding Users 112
Adding Unassociated Devices 112
Adding Users with Associated Devices 113
Number of Devices Per User 114
License Usage Report 114
Cisco Unified Reporting 115
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PART V Monitoring and Recording 117
CHAPTER 11 Silent Monitoring 119
Silent Monitoring Overview 119

Silent Monitoring Prerequisites 120
Configure Silent Monitoring Task Flow 120
Enable Built in Bridge for Phones Clusterwide 121
Enable Built in Bridge for a Phone 121
Enable Monitoring Privileges for Supervisor 122
Assign a Monitoring Calling Search Space 122
Configure Silent Monitoring Notification Tones 123
Configure Secure Silent Monitoring 123
Configure an Encrypted Phone Security Profile 123
Assign Security Profile to Phone 124

Configure Silent Monitoring for Unified Contact Center Express 124
Silent Monitoring Interactions 125
Silent Monitoring Restrictions 126
CHAPTER 12 Recording 127
Recording Overview 127

Multi-Fork Recording 128
Recording Media Source Selection 129
Recording Prerequisites 130
Recording Configuration Task Flow 131
Create a Recording Profile 131
Configure SIP Profile for Recording 132
Configure SIP Trunks for Recording 132
Configure Route Pattern for Recording 133
Configure Agent Phone Line for Recording 133
Enable Built in Bridge for Cluster 134
Enable Built in Bridge for a Phone 134

Enable Gateway for Recording 135
Configure Recording Notification Tones 135
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Configure a Record Feature Button 136

Configure a Phone Button Template for Recording 136
Associate a Phone Button Template with a Phone 137
Configure a Record Softkey 137
Configure a Softkey Template for Recording 138
Associate a Softkey Template with a Phone 138
Associate a Softkey Template with a Common Device Configuration 139
Recording Call Flow Examples 140
Recording Interactions and Restrictions 140
PART VI Call Center Features 143
CHAPTER 13 Agent Greeting 145
Agent Greeting Overview 145

Agent Greeting Prerequisites 145
Agent Greeting Configuration Task Flow 145
Configure Built In Bridge 147
Agent Greeting Troubleshooting 147
CHAPTER 14 Auto-Attendant 149
Auto-Attendant Overview 149

Cisco Unity Connection Configuration 150
Cisco Unity Connection Configuration Task Flow 150
Configure CTI Route Point 151
Configure Auto-Attendant System Call Handler 152
Configure Caller Input Option 152
Configure Extension for Operator Call Handler 153
Modify Standard Call Transfer Rule for Operator 153
Update Default System Transfer Restriction Table 153
Cisco Unity Connection Auto-Attendant Troubleshooting 154
Cisco Unified CCX Configuration 154
Cisco Unified CCX Prerequisites 154
Cisco Unified CCX Auto-Attendant Task Flow 154
Cisco Unity Express Configuration 156
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Cisco Unity Express Auto-Attendant Troubleshooting 156
CHAPTER 15 Manager Assistant 157
Cisco Unified Communications Manager Assistant Overview 157

Manager Assistant Shared Line Overview 158
Manager Assistant Proxy Line Overview 159
Manager Assistant Prerequisites 159
Manager Assistant Task Flow for Proxy Lines 160
Run the Cisco Unified CM Assistant Configuration Wizard 160
Manager Assistant Service Parameters for Proxy Line 162
Configure Manager And Assign Assistant For Proxy Line 166
Configure Assistant Line Appearances for Proxy Line 167
Manager Assistant Task Flow for Shared Lines 168
Configure Partitions for Manager Assistant Shared Line Support 169
Partition Name Guidelines for Manager Assistant Shared Line Support 170
Configure Calling Search Spaces for Manager Assistant Shared Line Support 170
Configure Cisco IP Manager Assistant Service Parameters 171
Configure Intercom Settings 171
Configure an Intercom Partition 172
Configure an Intercom Calling Search Space 173
Configure an Intercom Directory Number 173
Configure an Intercom Translation Pattern 173
Configure Multiple Manager Assistant Pool 174
Configure Secure TLS Connection to CTI for Manager Assistant 175
Configure IPMASecureSysUser Application User 175
Configure CAPF Profile 176
Configure Cisco WebDialer Web Service 177
Configure CTI Route Point 178

Configure IP Phone Services for Manager and Assistant 178
Cisco IP Phone Services Configuration Fields 179
Configure Phone Button Templates for Manager, Assistant, and Everyone 182
Configure a Phone Button Template for Manager Assistant 182
Associate a Manager Assistant Button Template with a Phone 183
Configure Manager and Assign Assistant for Shared Line Mode 183
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Configure Assistant Line Appearances for Shared Line 184

Install Assistant Console Plugin 185
Manager Assistant Interactions 186
Manager Assistant Restrictions 188
Cisco Unified Communications Manager Assistant Troubleshooting 189
Calling Party Gets Reorder Tone 190

Calls Do Not Get Routed When Filtering Is On or Off 191
Cisco IP Manager Assistant Service Unreachable 191
Cannot Initialize Cisco IP Manager Assistant Service 193
Assistant Console Installation from Web Fails 193
HTTP Status 503—This Application Is Not Currently Available 193
Manager Is Logged Out While the Service Is Still Running 194
Manager Cannot Intercept Calls That Are Ringing on the Assistant Proxy Line 194
No Page Found Error 195
System Error - Contact System Administrator 195
Unable to Call Manager When Cisco IP Manager Assistant Service is Down 196
User Authentication Fails 197
PART VII Voice Messaging Features 199
CHAPTER 16 Audible Message Waiting Indicator 201
Audible Message Waiting Indicator Overview 201

Audible Message Waiting Indicator Prerequisites 201
Audible Message Waiting Indicator Configuration Task Flow 201
Configure Audible Message Waiting Indicator Service Parameters 202
Configure Audible Message Waiting Indicator for a Directory Number 202
Configure Audible Message Waiting Indicator for a SIP Profile 203
Audible Message Waiting Indicator Troubleshooting 203
Audible Message Waiting Indicator Is Not Heard on the Phone 203
Localized AMWI Tone Is Not Played in a Specific Locale 204
CHAPTER 17 Immediate Divert 205
Immediate Divert Overview 205

Immediate Divert Prerequisites 206
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Immediate Divert Configuration Task Flow 206

Configure Immediate Divert Service Parameters 207
Configure a Softkey Template for Immediate Divert 208
Add a Softkey Template to the Common Device Configuration 210
Associate a Common Device Configuration with a Phone 210
Immediate Divert Interactions 211
Immediate Divert Restrictions 212
Immediate Divert Troubleshooting 214
Key is not active 214
Temporary Failure 214
Busy 214
PART VIII Conferencing Features 215
CHAPTER 18 Ad Hoc Conferencing 217
Ad Hoc Conferencing Overview 217

Ad Hoc Conferencing Task Flow 217
Configure Softkey Template for Conferencing 218
Associate Softkey Template Common Device 219
Add a Softkey Template to a Common Device Configuration 220
Configure Ad Hoc Conferencing 221
Ad Hoc Conferencing Service Parameters 222
Configure Join Across Lines 224
Conference Interactions 225
Conference Restrictions 225
CHAPTER 19 Meet-Me Conferencing 229
Meet-Me Conferencing Overview 229

Meet-Me Conferencing Task Flow 229
Configure a Softkey Template for Meet-Me Conferencing 230
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Configure a Meet-Me Conferencing Number 233
Meet-Me Number and Pattern Settings 233
Meet-Me Conferencing Restrictions 234
CHAPTER 20 Conference Now 235
Conference Now Overview 235

Conference Now Prerequisites 235
Activate Cisco IP Voice Media Streaming 236
Configure Conference Now Settings 236
Enable Conference Now for User 237
Enable Conference Now via LDAP 237
Conference Now Interactions 238
Conference Now Restrictions 239
PART IX Placing Calls 241
CHAPTER 21 Call Back 243
Call Back Overview 243

Call Back Prerequisites 243
Call Back Configuration Task Flow 244
Configure Softkey Template for CallBack 245
Associate CallBack Softkey Template with a Common Device Configuration 246
Associate CallBack Softkey Template with Phone 247
Configure CallBack Button 248
Configure Phone Button Template for Call Back 248
Associate a Button Template with a Phone 249
Call Back Interactions 249
Call Back Restrictions 250
Call Back Troubleshooting 251
Unplug/Reset Phone After Pressing CallBack Softkey but Before CallBack Occurs 251
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Caller Misses to View Availability Notification Before Phone Reset 251

Call Back Error Messages 251
CallBack Is Not Active 252
CallBack Is Already Active 252
CallBack Cannot Be Activated 252
Key Not Active 253
CHAPTER 22 Hotline 255
Hotline Overview 255

System Requirements for Hotline 256
Hotline Configuration Task Flow 256
Create Custom Softkey Template 256
Configure Hotline on Phones 257
Configure Route Class Signaling Task Flow 258
Enable Route Class Signaling in the Cluster 258
Enable Route Class Signaling on Trunks 259
Enable Route Class Signaling on Gateways 259
Configure Signaling Labels for the Hotline Route Class 260
Configure the Route Class on Hotline Route Patterns 261
Configure the Route Class on Hotline Translation Patterns 261
Configure Hotline to Call Only or Receive Only Task Flow 262
Configure Partitions for Hotline Call Only Receive Only 262
Configure Calling Search Space for Hotline Call Only Receive Only 262
Configure Call Only on Hotline Phone 263
Configure Receive Only on Hotline Phone 263
Configure Call Screening with a Calling Search Space 264
Configure Partitions for Hotline Call Screening 264

Create Calling Search Space for Hotline Call Screening 265
Configure Hotline Phones for Call Screening 266
Hotline Troubleshooting 266
CHAPTER 23 Speed Dial and Abbreviated Dial 269
Speed Dial and Abbreviated Dial Overview 269

Programming Speed Dials with Pauses 269
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Speed Dial and Abbreviated Dial Configuration Task Flow 270

Configure Speed Dial and Abbreviated Dial 270
CHAPTER 24 WebDialer 273
WebDialer Overview 273

WebDialer Prerequisites 273
WebDialer Configuration Task Flow 273
Activate WebDialer 275
Enable WebDialer Tracing 275
Configure WebDialer Servlet 276
Configure Redirector Servlet 276
Configure WebDialer Application Server 277
Configure Secure TLS Connection to CTI 277
Configure WDSecureSysUser Application User 278
Configure CAPF Profile 278
Configure Cisco IP Manager Assistant 280
Configure Language Locale for WebDialer 280
Configure WebDialer Alarms 281

Configure Application Dial Rules 281
Add Users to Standard CCM End User Group 282
Configure Proxy User 282
Add a WebDialer End User 283
Assign Authentication Proxy Rights 283
WebDialer Interactions 284
WebDialer Restrictions 285
WebDialer Troubleshooting 285
Authentication Error 285
Service Temporarily Unavailable 285
Directory Service Down 286
Cisco CTIManager Down 286
Session Expired, Please Login Again 286
User Not Logged In on Any Device 287
Failed to Open Device/Line 287
Destination Not Reachable 287
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CHAPTER 25 Paging 289
Paging Overview 289

InformaCast Basic Paging 289
InformaCast Advanced Notification 289
InformaCast Mobile 290
Paging Prerequisites 290
Cisco Unified Communications Manager Configuration for Basic Paging Task Flow 291
Configure SNMP for Paging 292
Enable SNMP Service 292
Create an InformaCast SNMP Community String 293
Configure Region for Paging 293
Set Default Codec to G.711 293
Configure a Device Pool for Paging 294

Configure Partitions and Calling Search Spaces for Paging 294
Configure Route Partition for InformaCast Paging 295
Configure Calling Search Space for InformaCast Paging 295
Configure CTI Ports for Paging 296
Configure Access Control Group with AXL Access 296
Configure Application User for Paging 297
Enable Web Access for a Phone 298
Enable Web Access for Common Phone Profile 298
Enable Web Access for Enterprise Phone Configuration 299
Configure Authentication URL 299
Set Authentication URL 299
Reset Your Phones 300
Test Your Phones 300
Advanced Notification Paging Configuration Task Flow 301
Install the InformaCast Virtual Appliance 301
Configure Connection to InformaCast 303
Configure Panic Button 304
Configure CallAware Emergency Call Alerting 306
Paging Interactions 307
Advanced Notification Paging Interactions 308
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CHAPTER 26 Intercom 309
Intercom Overview 309

Intercom and Default Devices 309
Intercom Prerequisites 310
Intercom Configuration Task Flow 310
Configure Intercom Partition 310
Configure an Intercom Calling Search Space 311
Configure an Intercom Translation Pattern 312
Configure an Intercom Directory Number 312
Intercom Line and Speed Dial Configuration 313
Intercom Interactions 313
Intercom Restrictions 315
Intercom Troubleshooting 316
Busy Tone When Dialing Out of Intercom Line 316
Intercom Calls cannot use Talkback with Speaker, Handset or Headset 316
Troubleshooting SCCP 316

Intercom Lines Not Showing Up on Phone 316
Intercom Lines Not Showing Up When Phone Falls Back to SRST 317
Troubleshooting SIP 317
Debug Phones That Are Running SIP 317
Configuration of Phones That Are Running SIP 317
Cisco Extension Mobility User Is Logged In But Intercom Line Does Not Display 317
Intercom Line Fails to Display on Phone 318
PART X Receiving Calls 319
CHAPTER 27 Prime Line Support 321
Prime Line Support Overview 321

Prime Line Support Prerequisites 321
Prime Line Support Configuration Task Flow 321
Configure Clusterwide Prime Line Support 322
Configure Prime Line Support for Devices 323
Prime Line Support Interactions 323
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Prime Line Support Troubleshooting 324

Prime Line Support Does Not Work When Set To True 324
Unable To Answer Inbound Calls 324
Inbound Calls Are Answered Automatically 324
CHAPTER 28 Call Forwarding 325
Call Forwarding Overview 325

Call Forward All, Including CFA Loop Prevention and CFA Loop Breakout 326
Call Forwarding Configuration Task Flow 327
Configure Partitions for Call Forwarding 327
Partition Name Guidelines for Call Forwarding 328
Configure Calling Search Space for Call Forwarding 329
Configure Call Forwarding when Hunt List is Exhausted or Hunt Timer Expires 330
Hunt Call Treatment Fields for Call Forwarding 330
Configure Call Forward No Bandwidth 332
Directory Number Configuration Fields for Call Forwarding 332
Configure Call Forward Alternate Destination 333
MLPP Alternate Party And Confidential Access Level Settings Fields for Call Forwarding 334
Configure Other Call Forwarding Types 334
Call Forwarding Fields 335
Enable Destination Override for Call Forwarding 343
Call Forwarding Interactions 343
Call Forwarding Restrictions 347
CHAPTER 29 Call Pickup 349
Call Pickup Overview 349

Group Call Pickup Overview 349
Other Group Pickup Overview 349
Directed Call Pickup Overview 350
BLF Call Pickup Overview 350
Call Pickup Configuration Task Flow 351
Configure a Call Pickup Group 353
Assign a Call Pickup Group to Directory Numbers 353
Configure Partitions for Call Pickup 354
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Configure Calling Search Space 355

Assign a Call Pickup Group to Hunt Pilots 356
Configure Call Pickup Notification 356
Configure Call Pickup Notification for a Call Pickup Group 357
Configure Call Pickup Notification for a Directory Number 358
Configure BLF Call Pickup Notification 358
Configure Directed Call Pickup 360
Configure a Time Period 360
Configure Time Schedule 360
Associate a Time Schedule with a Partition 360
Configure Automatic Call Answering 361
Configure Auto Call Pickup 361
Configure BLF Auto Pickup 362
Configure Call Pickup Phone Buttons 362
Configure Call Pickup Phone Button Template 363
Associate Call Pickup Button Template with Phone 363
Configure BLF Speed Dial Number for the BLF Call Pickup Initiator 364
Configure Softkeys for Call Pickup 364
Configure a Softkey Template for Call Pickup 365
Call Pickup Interactions 368

Call Pickup Restrictions 368
CHAPTER 30 Call Park and Directed Call 371
Call Park Overview 371

Call Park Prerequisites 372
Call Park Configuration Task Flow 372
Configure Clusterwide Call Park 373
Configure a Partition for Call Park 374
Configure a Call Park Number 375
Call Park Configuration Fields 376
Configure a Softkey Template for Call Park 377
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Associate a Softkey with a Phone 379
Configure Call Park Button 380
Configure a Phone Button Template for Call Park 380
Configure Park Monitoring 381
Configure Park Monitoring System Timers 381
Configure Park Monitoring for Hunt Pilots 382
Configure Park Monitoring for a Directory Number 383
Configure Park Monitoring via Universal Line Template 384
Call Park Interactions 386
Call Park Restrictions 387
Troubleshooting Call Park 387
User Cannot Park Calls 387
Call Park Number is Not Displayed Long Enough 388
Directed Call Park Overview 388
Directed Call Park Prerequisites 388
Directed Call Park Configuration Task Flow 388
Configure ClusterWide Directed Call Park 389
Configure a Directed Call Park Number 389
Directed Call Park Configuration Settings 390
Configure BLF/Directed Call Park Buttons 391
BLF/Directed Call Park Configuration Fields 391
Synchronize Directed Call Park with Affected Devices 392
Directed Call Park Interactions 392
Directed Call Park Restrictions 394
Troubleshooting Directed Call Park 394
User Cannot Retrieve Parked Calls 394
User Cannot Park Calls 395
User Receives a Reorder Tone After the Reversion Timer Expires 395
User Receives a Reorder Tone or Announcement 395
User Cannot Park a Call at a Number Within The Range 395
Parked Calls Revert Too Quickly 395
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Park Slot Unavailable 395

Parked Calls Do Not Revert to the Parked Call Number 396
Number or Range Cannot Be Deleted Because It Is in Use 396
CHAPTER 31 Extension Mobility 397
Extension Mobility Overview 397

Extension Mobility Prerequisites 397
Extension Mobility Configuration Task Flow 398
Activate Extension Mobility Services 398
Configure the Cisco Extension Mobility Phone Service 399
Create an Extension Mobility Device Profile for Users 400
Associate a Device Profile to a User 400
Subscribe to Extension Mobility 401
Configure the Change Credential IP Phone Service 401
Configure Service Parameters for Extension Mobility 402
Extension Mobility Service Parameters 403
Cisco Extension Mobility Interactions 406
Cisco Extension Mobility Restrictions 407
Extension Mobility Troubleshooting 408
Troubleshoot Extension Mobility 408
Blank User ID or PIN 409
Busy Please Try Again 409
Database Error 409
Dev Logon Disabled 409
Device Name Empty 409
EM Service Connection Error 410
Extension Mobility Performance During Upgrade 410
Host Not Found 410
HTTP Error 410
Phone Resets 410
Phone Services Unavailable After Login 411
Phone Services Unavailable After Logout 411
User Logged in Elsewhere 411
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User Profile Absent 411
CHAPTER 32 Extension Mobility Cross Cluster 413
Extension Mobility Cross Cluster Overview 413

Extension Mobility Cross Cluster Prerequisites 413
Extension Mobility Cross Cluster Configuration Task Flow 413
Configure Extension Mobility 415
Activate Services for Extension Mobility Cross Cluster 416
Configure the Extension Mobility Phone Service 416
Configure a Device Profile for Extension Mobility Cross Cluster 417
Enable Extension Mobility Cross Cluster for a User 423
Subscribe Devices to Extension Mobility 423
Configure Certificates for Extension Mobility Cross Cluster 423
Activate the Bulk Provisioning Service 424
Configure Bulk Certificate Management and Export Certificates 424
Consolidate the Certificates 425
Import the Certificates into the Clusters 426
Configure Extension Mobility Cross Cluster Devices and Templates 427
Create a Common Device Configuration 427
Configure an Extension Mobility Cross Cluster Template 428
Set the Default Template 428
Add Extension Mobility Cross Cluster Devices 429
Configure a Geolocation Filter for Extension Mobility Cross Cluster 429
Configure Feature Parameters for Extension Mobility Cross Cluster 429
Feature Parameter Fields for Extension Mobility Cross Cluster 430
Configure Intercluster SIP Trunk for Extension Mobility Cross Cluster 433
Configure an Intercluster Service Profile for Extension Mobility Cross Cluster 433
Configure Remote Cluster Services 434
Extension Mobility Cross Cluster Interactions 434
Extension Mobility Cross Cluster Restrictions 435
Extension Mobility Cross Cluster and Security Mode for Different Cluster Versions 437
Extension Mobility Cross Cluster Troubleshooting 439
Extension Mobility Application Error Codes 439
Extension Mobility Service Error Codes 440
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CHAPTER 33 Extension Mobility Roaming Across Clusters 445
Extension Mobility Roaming Across Clusters Overview 445

System Requirements for Extension Mobility Roaming Across Clusters 446
Extension Mobility Roaming Across Clusters Login 446
ILS Interaction 449
Extension Mobility Roaming Across Clusters Task Flow 449
Activate Extension Mobility Services 450
Configure the Cisco Extension Mobility Phone Service 450
Create an Extension Mobility Device Profile for Users 451
Associate a Device Profile to a User 452
Subscribe to Extension Mobility 453
Configure Roaming for Extension Mobility Users 453
Extension Mobility Roaming Across Clusters Interactions and Restrictions 454
Different Types of Extension Mobility 454
Extension Mobility Roaming Across Clusters Troubleshooting 455
Blank User ID or PIN 455
Busy Please Try Again 455
Database Error 455
Dev Logon Disabled 456
Device Name Empty 456
EM Service Connection Error 456
Host Not Found 456
HTTP Error 456
Phone Resets 456
Phone Services Unavailable After Login 457
Phone Services Unavailable After Logout 457
User Logged in Elsewhere 457
User Profile Absent 457
CHAPTER 34 Hold Reversion 459
Hold Reversion Overview 459
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Hold Reversion Prerequisites 459

Hold Reversion Configuration Task Flow 460
Configure Call Focus Priority for Hold Reversion 460
Configure Hold Reversion Timer Defaults for Cluster 461
Configure Hold Reversion Timer Settings for Phone 462
Hold Reversion Interactions 463
Hold Reversion Restrictions 464
CHAPTER 35 Accessing Hunt Groups 465
Hunt Group Overview 465

Hunt Group Prerequisites 466
Hunt Group Configuration Task Flow 466
Configure a Softkey Template for Hunt Group 467
Configure Phones for Hunt Group 470
Configure Hunt Group Service Parameter 470
Hunt Group Interactions 471
Hunt Group Restrictions 472
CHAPTER 36 Malicious Call Identification 473
Malicious Call Identification Overview 473

Malicious Call Identification Prerequisites 473
Malicious Call Identification Configuration Task Flow 474
Set Malicious Call ID Service Parameter 475
Configure Malicious Call ID Alarms 475
Configure a Softkey Template for Malicious Call Identification 476
Configure Malicious Call Identification Button 478
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Configure Malicious Call ID Phone Button Template 479

Malicious Call Identification Interactions 480

Malicious Call Identification Restrictions 481
Malicious Call ID Troubleshooting 482
CHAPTER 37 Call Transfer 483
Call Transfer Overview 483

Call Transfer Configuration Task Flow 484
Configure Consult and Blind Transfer 484
Configure a Softkey Template for Transfer 484
Configure Transfer Button 487
Configure Transfer On-Hook 489
Configure Direct Transfer 489
Configure a Softkey Template for Direct Transfer 489
Configure Direct Transfer Button 492
Call Transfer Interactions 494
Call Transfer Restrictions 495
CHAPTER 38 External Call Transfer Restrictions 497
External Call Transfer Restrictions Overview 497

Configure External Call Transfer Restrictions Task Flow 498
Configure the Service Parameter for Call Transfer Restrictions 498
Configure Incoming Calls Task Flow 499
Configure the Clusterwide Service Parameter 499
Configure Gateways for Call Transfer Restrictions 500
Configure Trunks for Call Transfer Restrictions 500
Configure Outgoing Calls 501
External Call Transfer Restrictions Interactions 502
External Call Transfer Restrictions Restrictions 502
PART XI Presence and Privacy Features 505
CHAPTER 39 Barge 507
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Barge Overview 507

Built-In Conference 508
Shared Conference 508
Built-In and Shared Conference Differences 508
Barge Configuration Task Flow 509
Configure Softkey Template for Built-In Conferencing 510
Configure Softkey Template for Shared Conferencing 511
Associate Softkey Template with Phone 512
Associate a Softkey Template with Common Device Configuration 512
Add a Softkey Template to Common Device Configuration 513
Associate Common Device Configuration with Phone 514
Configure Barge for Built-In Conferencing 514
Configure Barge for Shared Conferencing 515
Associate User with Device 515
Barge Interactions 516
Barge Restrictions 516
Barge Troubleshooting 517
No Conference Bridge Available 517
Error: Past Limit 517
CHAPTER 40 BLF Presence 519
BLF Presence Overview 519

BLF Presence Prerequisites 519
BLF Presence Configuration Task Flow 520
Configure/Synchronize Cluster-Wide Enterprise Parameters for BLF 521
Configure Cluster-Wide Service Parameters for BLF 522
Configure BLF Presence Groups 522
BLF Presence Group Fields for BLF 523
BLF Presence Group Association with Devices and Users 524
Associate BLF Presence Groups with Phone 524
Associate BLF Presence Groups with SIP Trunk 525
Associate BLF Presence Groups with End User 526
Associate BLF Presence Groups with Application User 527
Accept BLF Presence Requests from External Trunks and Applications 527
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Contents
Configure a Calling Search Space for Presence Requests 528

Configure a Phone Button Template for BLF and SpeedDial Buttons 529
Associate Button Template with a Device 530
Configure User Device Profile 530
BLF Presence Interactions 531
BLF Presence Restrictions 531
CHAPTER 41 Call Display Restrictions 533
Call Display Restrictions Overview 533

Call Display Restrictions Configuration Task Flow 533
Configure Partitions for Call Display Restrictions 534
Partition Name Guidelines 535
Configure Calling Search Spaces for Call Display Restrictions 535
Configure the Service Parameter for Connected Number Display Restriction 536
Configure Translation Patterns 537
Translation Pattern Fields for Call Display Restrictions 537
Configure Phones for Call Display Restrictions 538
Configure the PSTN Gateway for Call Display Restrictions 540
Configure Call Display Restrictions on SIP Trunks 540
SIP Trunk Fields for Call Display Restrictions 541
Call Display Restrictions Interactions 542
Call Display Restrictions Feature Restrictions 544
CHAPTER 42 Do Not Disturb 545
Do Not Disturb Overview 545

Do Not Disturb Configuration Task Flow 546
Configure Busy Lamp Field Status 547
Configure Do Not Disturb on a Common Phone Profile 547
Apply Do Not Disturb Settings to the Phone 548
Configure a Do Not Disturb Feature Button 549
Configure Phone Button Template for Do Not Disturb 549
Associate Button Template with Phone 550
Configure a Do Not Disturb Softkey 550
Configure Softkey Template for Do Not Disturb 551
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Contents

Associate Softkey Template with a Phone 553
Do Not Disturb Interactions and Restrictions 554
Interactions 554
Restrictions 555
Do Not Disturb Troubleshooting 556
CHAPTER 43 Privacy 557
Privacy Overview 557

Privacy on Hold 557
Privacy Configuration Task Flow 558
Enable Privacy Cluster-wide 558
Enable Privacy for a Device 558
Configure Privacy Phone Button Template 559
Associate Privacy Phone Button Template with a Phone 559
Configure Shared Line Appearance 560
Configure Privacy on Hold 561
Privacy Restrictions 561
CHAPTER 44 Private Line Automatic Ringdown 563
Private Line Automatic Ringdown Overview 563

Private Line Automatic Ringdown Configuration Task Flow for SCCP Phones 563
Create Partition 564
Assign Partitions to Calling Search Spaces 564
Assign Partition to the Private Line Automatic Ringdown Destination 565
Configure Translation Pattern for Private Line Automatic Ringdown on Phones 565
Private Line Automatic Ringdown Configuration Task Flow for SIP Phones 566
Create SIP Dial Rule for Private Line Automatic Ringdown 566
Assign Private Line Automatic Ringdown Dial Rule to SIP Phone 566
Private Line Automatic Ringdown Troubleshooting 567
CHAPTER 45 Secure Tone 569
Secure Tone Overview 569

Protected Device Gateways 570
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Contents
Secure Tone Prerequisites 570

Secure Tone Configuration Task Flow 570
Configure Phone As a Protected Device 571
Configure Directory Number for Secure Tones 571
Configure Secure Tone Service Parameters 572
Configure MGCP E1 PRI Gateway 572
Secure Tone Interactions 573
Secure Tone Restrictions 573
PART XII Custom Features 575
CHAPTER 46 Branding Customizations 577
Branding Overview 577

Branding Prerequisites 577
Branding Task Flow 578
Enable Branding 578
Disable Branding 579
Restart the Tomcat Service 580

Branding File Requirements 580
CHAPTER 47 Client Matter Codes and Forced Authorization Codes 585

Client Matter Codes and Forced Authorization Codes Overview 585
Client Matter Codes and Forced Authorization Codes Prerequisites 585
Client Matter Codes and Forced Authorization Codes Configuration Task Flow 586
Configure Client Matter Codes 586
Add Client Matter Codes 587
Enable Client Matter Codes 587
Configure Forced Authorization Codes 587
Add Forced Authorization Codes 588
Enable Forced Authorization Codes 588
Client Matter Codes and Forced Authorization Codes Interactions 589
Client Matter Codes and Forced Authorization Codes Restrictions 590
CHAPTER 48 Custom Phone Rings and Backgrounds 591
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Contents
Custom Phone Rings Overview 591

Custom Phone Rings Prerequisites 591
Custom Phone Rings Configuration Task Flow 592
Prepare Custom Phone Rings for Upload 592
Upload Custom Phone Rings to TFTP Server 592
Restart TFTP Service 593
PCM File Format Requirements 593
Ringlist.xml File Format Requirements 593
Custom Backgrounds 594
Custom Backgrounds Configuration Task Flow 594
Create Phone Background Images 595
Edit the List.xml file 596
Upload Backgrounds to TFTP Server 597
Restart the TFTP Server 597
Assign Phone Background for Phone Users 597
CHAPTER 49 Music On Hold 599
Music On Hold Overview 599

Caller-Specific Music On Hold 599
Increased Capacity of IP Voice Media Streaming Application and Expanded MOH Audio Source 600
Performance Impact of Media Devices with Services 600
Configuration Limitations for Capacity Planning 602
Interwork External Multicast MOH to Unicast MOH 603
Music On Hold Prerequisites 604
Music On Hold Configuration Task Flow 604
Activate Cisco IP Voice Media Streaming 605
Configure Music On Hold Server 606
Upload Audio File for Music On Hold 606
Configure Music On Hold Audio Source 607
Configure Fixed Music On Hold Audio Source 608
Add MOH to Media Resource Group 608
Configure Media Resource Group List 609
Add Media Resources to Device Pool 609
Configure MOH Service Parameters 610
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Contents
View Music on Hold Audio File 610

Unicast and Multicast Audio Sources 611
Music On Hold Interactions 613
Music On Hold Restrictions 614
Music On Hold Troubleshooting 616
Music On Hold Does Not Play on Phone 616
CHAPTER 50 Self Care Portal 617
Self Care Portal Overview 617

Self Care Portal Task Flow 617
Grant User Access to the Self Care Portal 618
Configure the Self Care Portal Options 618
Self Care Portal Interactions and Restrictions 619
CHAPTER 51 Emergency Call Handler 621
Emergency Call Handler Overview 621

Emergency Call Handler Prerequisites 622
Emergency Call Handler Task Flow 622
Enable Emergency Call Handler 623
Configure Emergency Location Groups 624
Add a Device Pool to an Emergency Location Group 624
Add Device to an Emergency Location Group 625
Enable Route Patterns and Translation Patterns 626
Bulk Administration of Emergency Location Groups and Phones 626
Bulk Administration of Emergency Location Groups and Phones Task Flow 626
Interactions 629
Emergency Call Handler Troubleshooting 631
About Emergency Call Handler Troubleshooting Scenarios 631
Configuration Scenarios 631
Emergency Calls Get Busy Signals and Are Not Routed 631
Emergency Location Numbers Are Dialed from Outside Running a Reorder Tone 631
Outgoing Calls Scenarios 632
Outgoing Emergency Call Does Not Contain Calling Party as Emergency Location Number 632
Outgoing Emergency Call Contains Modified Emergency Location Number 632
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Contents
Incoming Calls Scenarios 632

Incoming PSAP Callback Call Fails 632
Incoming PSAP CallBack Call is Not Routed as Expected 633
CHAPTER 52 Enterprise Groups 635
Enterprise Groups Overview 635

Enterprise Groups Prerequisites 636
Enterprise Groups Configuration Task Flow 636
Verify Group Sync from LDAP Directory 637
Enable Enterprise Groups 637
Enable Security Groups 638
Create Security Group Filter 638
Synchronize Security Groups from LDAP Directory 638
Configure Cisco Jabber for Security Groups 639
View User Groups 640
Enterprise Groups Deployment Models (Active Directory) 640
Enterprise Groups Limitations 642
PART XIII Device Management 647
CHAPTER 53 Headset Management 649

Headset Management Overview 649
Feature Compatibility for Headset Management 649
Third-Party Headset Support 651
Workflow: Configure Headset Serviceability 651
Activate Cisco Headset Service 652
Prepare Your Headset COP Files 653
Configure User Profiles for Headset Users 653
Apply User Profiles to End Users 655
Headset Template Management 655
Configure a Headset Template 659
Firmware Management 660
Headset Inventory Management 660
Headset Inventory Settings 661
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Contents
View Headset Inventory 661

Headset Inventory Summary 662
Get an Aggregate Summary of Your Deployed Headsets 662
Headset Troubleshooting and Diagnostics 663
Generate PRT for Endpoints on Unified CM 663

Generate PRT for Endpoints on RTMT 664
CHAPTER 54 Video Endpoints Management 665
Video Endpoints Management Overview 665

Video Endpoints Management Feature Compatibility 666
Migration Considerations for Video Endpoints Provisioning 667
Video Endpoints Migration Report 668
Provisioning and Migration Scenarios 669
Add Migrating Video Endpoint to Unified CM 670
PART XIV Advanced Call Processing 673
CHAPTER 55 Configure Call Control Discovery 675
Call Control Discovery Overview 675

Call Control Discovery Prerequisites 675
Call Control Discovery Configuration Task Flow 675
Configure SAF Security Profile 677
Configure SAF Forwarders 678
Configure SIP or H.323 Intercluster Trunks 678
Configure Hosted DN Groups 679
Configure Hosted DN Patterns 679
Configure the Advertising Service 680
Configure the Partition for Call Control Discovery 680
Configure the Requesting Service 680
Block Learned Patterns 681
Call Control Discovery Interactions 682
Call Control Discovery Restrictions 683
CHAPTER 56 Configure External Call Control 685
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Contents
External Call Control Overview 685

External Call Control Prerequisites 686
External Call Control Configuration Task Flow 686
Configure a Calling Search Space for External Call Control 687
Configure an External Call Control Profile 688
Assign a Profile to a Translation Pattern 688
Import the Route Server Certificate into the Trusted Store 689
Export the Self-Signed Certificate to the Route Server 689
Configure the Chaperone Function 690
Configure Customized Announcements 691
External Call Control Interactions 692
External Call Control Restrictions 694
CHAPTER 57 Configure Call Queuing 695
Call Queuing Overview 695

Call Queuing Prerequisites 696
Call Queuing Task Flow 697
Configure Announcements 697
Configure Music On Hold 698
Audio Source Fields for Music On Hold 699
Configure Hunt Pilot Queuing 701
Automatically Logout Hunt Member on No Answer 703
Call Queuing Interactions 703
Call Queuing Restrictions 704
Performance and Scalability for Hunt Pilots with Call Queuing 704
CHAPTER 58 Configure Call Throttling 707
Call Throttling Overview 707

Call Throttling Configuration Task Flow 708
Configure Call Throttling 708
Configure Memory Throttling 708
CHAPTER 59 Configure Logical Partitioning 711
Logical Partitioning Overview 711
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Contents
Logical Partitioning Configuration Task Flow 711

Enable Logical Partitioning 712
Configure Geolocations 712
Create Geolocations 713
Assign Geolocations 713
Set the Default Geolocation 714
Configure a Logical Partitioning Default Policy 714
Configure Devices to Avoid Logical Partitioning Checks 714
Configure Geolocation Filters 715
Create Geolocation Filter Rules 715
Assign Geolocation Filters 716
Set the Default Geolocation Filter 716
Define a Set of Logical Partitioning Policy Records 717
Enable Location Conveyance 717
Logical Partitioning Interactions 718
Logical Partitioning Restrictions 719
CHAPTER 60 Configure Location Awareness 721
Location Awareness Overview 721

Wireless Network Updates 722
Supported Endpoints for Location Awareness 722
Location Awareness Prerequisites 723
Location Awareness Configuration Task Flow 723
Start Services for Wireless Infrastructure Synchronization 724
Configure Wireless Access Point Controller 724
Insert Infrastructure Devices 725
Deactivate Infrastructure Device from Tracking 726
Related Documentation 726
Location Awareness Restrictions 727
CHAPTER 61 Configure Flexible DSCP Marking and Video Promotion 729
Flexible DSCP Marking and Video Promotion Overview 729

Custom QoS Settings for Users 730
Traffic Class Label 731
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Contents
DSCP Settings Configuration Task Flow 731

Configure Flexible DSCP Marking and Video Promotion Policy 731
Flexible DSCP Marking and Video Promotion Service Parameters 732
Configure Custom QoS Policy for Users 733
Configure Custom QoS Settings in SIP Profile 733
Apply Custom QoS Policy to a Phone 734
Flexible DSCP Marking and Video Promotion Interactions 735
Flexible DSCP Marking and Video Promotion Restrictions 735
CHAPTER 62 Separate Calling Party Number and Billing Number in SIP 737
External Presentation Name and Number Overview 737

Configuration Overview 737
Call Processing 738
Incoming Call Process 738
Outgoing Call Process 739
External Presentation Number Mask Operation 739
Directory Number Overview 740
Directory Number Configuration Tasks 740
Import an End User from LDAP 740
Add an End User Manually 741
Add New Phone for End User 742
Move an Existing Phone to a End User 743

Configure External Presentation Information on DN 743
SIP Profile Overview 744
SIP Profile Configuration Tasks 745
Configure SIP Profiles 745
Configure External Presentation Information on SIP Profile 745
SIP Trunk Overview 746
Trunk Configuration Tasks 747
Configure SIP Trunk Security Profile 747
Configure Common Device Configuration 748
Configure SIP Trunks 749
Configure Presentation Information on SIP Trunks 750
Intercluster SME Call Flows 751
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CHAPTER 63 SIP OAuth Mode 753
SIP OAuth Mode Overview 753

SIP OAuth Mode Prerequisites 754
SIP OAuth Mode Configuration Task Flow 754
Upload CA Certificate to the Phone Edge Trust 755
Configure Refresh Logins 755
Configure OAuth Ports 756
Configure OAuth Connection to Expressway-C 757
Enable SIP OAuth Mode 757
Restart Cisco CallManager Service 757
Configure Device Security Mode in Phone Security Profile 758
Configure SIP Oauth Registered Phones for MRA Mode 758
PART XV QoS Management 761
CHAPTER 64 Configure QoS with APIC-EM Controller 763
APIC-EM Controller Overview 763

APIC-EM Controller Prerequisites 764
APIC-EM Controller Configuration Task Flow 764
Configure the APIC-EM Controller 765
Upload APIC-EM Controller Certificate 765
Configure HTTPS Connection to APIC-EM Controller 766
Enable External QoS Service for System 766
Configure External QoS Service at SIP Profile Level 766
Assign SIP Profile to Phones 767
CHAPTER 65 Configure AS-SIP Endpoints 769
AS-SIP Overview 769

Third-Party AS-SIP Phones 769
AS-SIP Conferencing 771
AS-SIP Prerequisites 771
AS-SIP Enpdoint Configuration Task Flow 772
Configure a Digest User 773
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Contents
Configure SIP Phone Secure Port 773

Restart Services 773
Configure SIP Profile for AS-SIP 774
Configure Phone Security Profile for AS-SIP 775
Configure AS-SIP Endpoint 775
Associate Device with End User 776
Configure SIP Trunk Security Profile for AS-SIP 777
Configure SIP Trunk for AS-SIP 777
Configure AS-SIP Features 778
CHAPTER 66 Configure Multilevel Precedence and Preemption 781
Multilevel Precedence and Preemption Overview 781

Multilevel Precedence and Preemption Prerequisites 781
Multilevel Precendence and Preemption Task Flow 781
Configure Domains and Domain Lists 783
Configure a Multilevel Precedence and Preemption Domain 784
Configure a Resource Priority Namespace Network Domain 784
Configure a Resource Priority Namespace Network Domain List 785
Configure a Common Device Configuration for Multilevel Precedence and Preemption 785
Configure the Enterprise Parameters for Multilevel Precedence and Preemption 786
Enterprise Parameters for Multilevel Precedence and Preemption 786
Configure a Partition for Multilevel Precedence and Preemption 787
Partition Naming Guidelines 788
Configure a Calling Search Space for Multilevel Precedence and Preemption 788
Configure a Route Pattern for Multilevel Precedence and Preemption 789
Route Pattern Configuration Fields for Multilevel Precedence and Preemption 789
Configure a Translation Pattern for Multilevel Precedence and Preemption 790
Configure Multilevel Precedence and Preemption for Gateways 791
Configure Multilevel Precedence and Preemption for Phones 792
Multilevel Precedence and Preemption Settings for Phones 792
Configure a Directory Number to Place Multilevel Precedence and Preemption Calls 794
Configure a User Device Profile for Multilevel Precedence and Preemption 794
Configure the Default Device Profile for Multilevel Precedence and Preemption 795
Multilevel Precedence and Preemption Interactions 796
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Contents
Multilevel Precedence and Preemption Restrictions 797
PART XVI SIP Interoperability 801
CHAPTER 67 Configure SIP Normalization and Transparency 803
SIP Normalization and Transparency Overview 803

Default Scripts for SIP Normalization and Transparency 804
SIP Normalization and Transparency Prerequisites 804
SIP Normalization and Transparency Configuration Task Flow 804
Create New SIP Normalization and Transparency Scripts 805
Apply Normalization or Transparency Script to SIP Trunk 806
Apply Normalization or Transparency to SIP Devices 806
CHAPTER 68 Configure SDP Transparency Profiles 809
SDP Transparency Profile Overview 809

SDP Transparency Profile Restrictions 809
SDP Transparency Profile Prerequisites 810
Configure SDP Transparency Profile 810
CHAPTER 69 Configure Presentation Sharing using BFCP 811

Binary Floor Control Protocol Overview 811
BFCP Architecture 811
BFCP Limitations 812
Presentation Sharing using BFCP Prerequisites 812
Presentation Sharing using BFCP Configuration Task Flow 813
Enable BFCP Support for SIP Trunks 813
Enable Presentation Sharing using BFCP for Third-Party Phones 814
CHAPTER 70 Video Telephony 815

Video Telephony Overview 815
Video Telephony Support 815
Video Calls 816
Real-Time Transport Control Protocol Pass-Through in MTP Topologies 816
Video Codecs 817
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Contents
Video Network 818

Video Telephony Configuration Task Flow 820
H.323 Video 820
H.239-Extended Video Channels in H.323 Call 821
Support for Third-Party H.323 Devices 821
H.323 Devices Invoke Presentation Feature 821
Opening Second Video Channels 822
Call Admission Control (CAC) on Second Video Channels 823
Number of Video Channels Allowed 824
H.239 Commands and Indication Messages 824
Topology and Protocol Interoperability Limitation 824
Midcall Feature Limitation 824
Video Support 825
Skinny Client Control Protocol Video 825
SIP Video 825
Configuring SIP Devices for Video Calls 825
Cisco Video Conference Bridges 826
Cisco TelePresence MCU Video Conference Bridge 826
Cisco TelePresence Conductor Video Conference Bridge 826
Cisco Meeting Server 826
Video Encryption 827
Configure Interop with VCS 827
Video Features 828
Endpoint Support for the Binary Floor Control Protocol 828

Encrypted iX Channel 828
Encryption Modes 829
Non-Encrypted Modes 829
Far End Camera Control Protocol Support 830
QoS for Video Networks 830
Bandwidth Management 830

Enhanced Locations Call Admission Control 830
Session Level Bandwidth Modifiers 831
Video Resolution Support for SIP Phones 832
Alternate Routing 832
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Contents
Flexible DSCP Markings 833
Phone Configuration for Video Calls 833

Conference Control for Video Conferencing 833
Video Telephony and Cisco Unified Serviceability 834
Performance Counters 834
Video Bridge Counters 835
Call Detail Records (CDRs) 836
Call Management Records (CMRs) 836
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xlii
PA R T I
Getting Started
• Feature Configuration Overview, on page 1
• Configuration Tools, on page 3
CHAPTER 1
Feature Configuration Overview
• About the Feature Configuration Guide, on page 1
• Generate a Phone Feature List, on page 1
About the Feature Configuration Guide

This guide provides information about the tasks that you need to complete in order to configure features on
the Unified Communications Manager system. Use this guide after you have configured the call control system,
which includes "day 1" configurations such as inbound and outbound calling, dial plans, and network resources.
For information about configuring the call control system, see System Configuration Guide for Cisco Unified
Communications Manager.
Generate a Phone Feature List

Generate a phone feature list report to determine which devices support the feature that you want to configure.
Procedure
Step 1 From Cisco Unified Reporting, choose System Reports.

Step 2 From the list of reports, click Unified CM Phone Feature List.
Step 3 Perform one of the following steps:
• Choose Generate New Report (the bar chart icon) to generate a new report.
• Choose Unified CM Phone Feature List if a report exists.
Step 4 From the Product drop-down list, choose All.

Step 5 Click the name of the feature that you want to configure.
Step 6 Click Submit, to generate the report.
1
Getting Started
2
CHAPTER 2
Configuration Tools
• About the Feature Configuration Guide, on page 3
• Configuration Tools Overview, on page 3
• Generate a Phone Feature List, on page 5
About the Feature Configuration Guide

This guide provides information about the tasks that you need to complete in order to configure features on
the Unified Communications Manager system. Use this guide after you have configured the call control system,
which includes "day 1" configurations such as inbound and outbound calling, dial plans, and network resources.
For information about configuring the call control system, see System Configuration Guide for Cisco Unified
Configuration Tools Overview

The procedures in this guide require you to use the following two configuration tools:
• Cisco Unified Communications Manager Administration
• Cisco Unified Serviceability
This chapter provides a brief description of the tools and how to access them.
Cisco Unified Communications Manager Administration

Cisco Unified Communications Manager Administration Administration is a web-based application that allows
you to make individual, manual configuration changes to the Unified Communications Manager nodes. The
procedures in this guide describe how to configure features using this application.
If you need to perform bulk configuration tasks and want to automate the configuration process, you can use
the Unified Communications Manager Bulk Administration Tool (BAT) to make a large number of configuration
changes at the same time. For more information, see Bulk Administration Guide for Cisco Unified
3
Getting Started
Log In to Cisco Unified CM Administration
Log In to Cisco Unified CM Administration

Use the following procedure to log in to Cisco Unified Communications Manager Administration. After you
log in to Cisco Unified Communications Manager Administration, messages may display that indicate the
current state of licenses for Unified Communications Manager in the main window. For example, Unified
Communications Manager may identify the following situations:
• Unified Communications Manager currently operates with starter (demo) licenses, so upload the
appropriate license files.
• Unified Communications Manager currently operates with an insufficient number of licenses, so upload
additional license files.
• Unified Communications Manager does not currently use the correct software feature license. In this
case, the Cisco CallManager service stops and does not start until you upload the appropriate software
version license and restart the Cisco CallManager service.
Use the following procedure to browse into the server and log in to Cisco Unified CM Administration.
Procedure
Step 1 Start your preferred operating system browser.

Step 2 In the address bar of the web browser, enter the following case-sensitive URL:
https://<Unified CM-server-name>:{8443}/ccmadmin/showHome.do
where: <Unified CM-server-name> equals the name or IP address of the server
Note You can optionally specify a port number.
Step 3 A Security Alert dialog box displays. Click the appropriate button.
Step 4 At the main Cisco Unified CM Administration window, enter the username and password that you specified
during Unified Communications Manager installation and click Login. (If you want to clear the content of
both fields, click Reset.)
Note For security purposes, Cisco Unified Communications Manager Administration logs you out
after 30 minutes of inactivity, and you must log back in.
Cisco Unified Communications Manager Serviceability

Some procedures in this guide require you to use the Cisco Unified Serviceability application to start or restart
services on the Unified Communications Manager nodes.
Cisco Unified Serviceability is a web-based troubleshooting tool that provides the following functionality:
• Saves alarms and events for troubleshooting and provides alarm message definitions.
• Saves trace information to log files for troubleshooting.
• Monitors real-time behavior of components through the Cisco Unified Real-Time Monitoring Tool
(Unified RTMT).
4
Getting Started
Log into Cisco Unified Communications Manager Serviceability
• Provides audit capability by logging configuration changes to the system by a user or due to result of the
user action. This functionality supports the Information Assurance feature of Unified Communications
Manager and Cisco Unity Connection.
• Provides feature services that you can activate, deactivate, and view through the Service Activation
window.
• Generates and archives daily reports; for example, alert summary or server statistic reports.
• Allows Unified Communications Manager, IM and Presence Service and Cisco Unity Connection to
work as a managed device for Simple Network Management Protocol (SNMP) remote management and
troubleshooting.
• Monitors the disk usage of the log partition on a node (or all nodes in the cluster).
• Monitors the number of threads and processes in the system; uses cache to enhance the performance.
• Unified Communications Manager only: Generates Unified Communications Manager reports for
Quality of Service, traffic, and billing information through Cisco Unified Communications Manager
CDR Analysis and Reporting.
Log into Cisco Unified Communications Manager Serviceability

Use the following procedure to log in to Cisco Unified Serviceability.
Procedure
Step 1 Start your preferred operating system browser.

Step 2 In the address bar of the web browser, enter the following case-sensitive URL:
https://<Unified CM-server-name>:{8443}/ccmadmin/showHome.do
where: <Unified CM-server-name> equals the name or IP address of the server
Step 3 A Security Alert dialog box displays. Click the appropriate button.
Step 4 From Cisco Unified CM Administration, choose Cisco Unified Serviceability from the Navigation menu
drop-down list and click Go.
Step 5 Enter the username and password that you specified during Unified Communications Manager installation
and click Login.
Note For security purposes, the system logs you out after 30 minutes of inactivity, and you must log
back in.

5
Getting Started
Procedure


6
PA R T II
Remote Worker Features
• Cisco Unified Mobility, on page 9
• Device Mobility, on page 43
• Extend and Connect, on page 55
• Remote Worker Emergency Calling, on page 65
• Configure Mobile and Remote Access, on page 69
CHAPTER 3
Cisco Unified Mobility
• Cisco Unified Mobility Overview, on page 9
• Cisco Unified Mobility Prerequisites, on page 11
• Cisco Unified Mobility Configuration Task Flow, on page 12
• Cisco Unified Mobility Call Flow, on page 35
• FMC Over SIP Trunks Without Smart Client, on page 35
• Hunt Group Login and Logout for Carrier-Integrated Mobile Devices, on page 36
• Cisco Unified Mobility Interactions, on page 37
• Cisco Unified Mobility Restrictions, on page 38
• Cisco Unified Mobility Troubleshooting, on page 42
Cisco Unified Mobility Overview

Cisco Unified Mobility offers a set of mobility-related features that allow users to interact with Unified
Communications applications no matter where they may be, or which device they are using. Whether the
device you are using is a home office phone, a dual-mode Cisco Jabber on iPhone or Android client over a
WiFi connection, or a mobile phone from another cellular provider, you can still access Unified Communications
features and have the call be anchored in the enterprise.
For example, you can answer a call that is directed to your enterprise number from any of your configured
phones and then transfer the call to your mobile phone, allowing you to continue an in-progress conversation
as you are leaving the office.
Benefits of Cisco Unified Mobility

Most of the mobility features offer call anchoring within the enterprise–even if the call is placed to or from a
mobile device, the call is routed through an enterprise gateway.
This provides the following benefits:
• Single enterprise phone number and voicemail for all business calls, regardless of which device you are
using, and whether you are in the office or out of the office.
• Ability to extend business calls to a mobile device and have the call still be handled as if it were your
office phone.
• Calls placed from mobile devices are anchored to the enterprise and routed through an enterprise gateway.
This provides access to UC mid-call features, centralized billing and call detail records, and potential
cost savings from avoiding expensive cellular networks.
9
Mobility Features
• Ability to roam from one network to another and have the call not be dropped.
Mobility Features
Cisco Unified Mobility offers the following mobility-related features:
Mobility Feature Description
Single Number Reach Provides you with a single enterprise phone number and voicemail by which a
caller can reach you, regardless of whether you are in the office or outside the
office. When someone dials your enterprise number, you can answer the call
from your desk phone, or from any of your configured remote destinations (for
example, a home office phone, a dual-mode Cisco Jabber on iPhone or Android
client, and even a mobile phone from another provider) .
Move to Mobile Allows you to transfer an active call from your desk-phone to a mobile device
that is configured as a remote destination by pressing the Mobility softkey on
your Cisco IP Phone. It is associated with Single Number Reach as a part of the
Remote Destination configuration.
Similar to the Move to Mobile option is the Desk Pickup option, which fits the
example where you are on a mobile call and are just arriving at the office. You
can hang up on the call on your mobile device and immediately resume the call
by picking up your desk phone before the Maximum Wait Time for Desk Pickup
timer expires (the default is 10 seconds). This option is enabled as part of your
Single Number Reach configuration.
• Ensure that you set the Enforce Privacy Setting on Held Calls Service
Parameter to False.
• You can also use the Enterprise Feature Access code and the Session
Handoff codes to transfer calls between your remote destinations and desk
phone.
Mobile Voice Access Allows you to place calls from any remote phone and have the call be anchored
in the enterprise and presented to the called party as if you had called from your
office phone. When using this feature, you must dial in to a system interactive
voice response from your mobile device. After authenticating you, and prompting
you for the call destination, the system places the call as if you had called from
your enterprise phone.
You can also use Mobile Voice Access prompts to enable or disable Single
Number Reach for a remote destination.
Enterprise Feature Access Provides two-stage dialing from a configured remote destination. Also, ensures
that the call that is presented to the called party appears as if it originated from
.
your desk phone. Unlike Mobile Voice Access, to use Enterprise Feature
Access, you must be dialing from one of your configured remote destinations.
Enterprise Feature Access also allows you to access mid-call features while on
a call from a remote destination. You can access mid-call features by sending
DTMF digits that represent the codes for the various features such as Hold,
Exclusive Hold, Transfer.
10
Cisco Unified Mobility Prerequisites
Mobility Feature Description
Intelligent Session Enables automatic call anchoring for enterprise-originated calls that are placed
Control directly to configured remote destination numbers (for example, an
enterprise-originated call to a cell phone number that is configured as a remote
destination). By configuring a service parameter, you can have the system redirect
those calls automatically to the associated enterprise number, providing cost
savings and added UC functionality.
Dual-Mode Phones Cisco Jabber on iPhone and Android clients can be provisioned as dual-mode
devices. Dual-Mode phones have the capability of connecting over Wi-Fi or
through cellular networks. When the client is within the enterprise network, Cisco
Jabber can register to Unified Communications Manager over Wi-Fi, and has
UC calling and instant messaging functionality. If you configure a mobile identity
with the phone number of the mobile device, allowing the call to be transferred
from Jabber to the cellular device when leaving the enterprise network.
Note An added feature that is available to Cisco Jabber mobile clients is
Mobile and Remote Access, which allows Cisco Jabber clients to
connect to data networks when outside of the enterprise network.
For more information, see "Configure Mobile and Remote Access"
section in Feature Configuration Guide for Cisco Unified
Cisco Unified Mobility Prerequisites

Refer to the following prerequisites:
• Enabling Mobility features requires proper planning to ensure that your dial plan and call routing
configuration can handle the deployment needs. For more information, see "Mobile Collaboration"
section in the Cisco Collaboration System Solution Reference Network Designs guide.
• For information on which Cisco IP Phones support Mobility feature, see Generate a Phone Feature List,
on page 1.
• For a list of Cisco IP Phones that support the Mobility softkey, run a report for the Mobility feature.
• For a list of supported dual-mode phones, run a report for the Dual-Mode feature.
• If you are deploying Mobile Voice Access and you want to make additional locales available to your
system (if you want to use non-English phone locales or country-specific tones), you can download the
locale installers from cisco.com and install them through the Cisco Unified OS Administration interface.
For more information on installing locales, see Installation Guide for Cisco Unified Communications
Manager and the IM and Presence Service.
• Configure Self-Provisioning so that phone users can provision their own Cisco Jabber clients and remote
destinations. For more information, see "Configure Self Provisioning" and "Provisioning End Users"
section in the System Configuration Guide for Cisco Unified Communications Manager.
11
Cisco Unified Mobility Configuration Task Flow
Caution The Cisco mobility solution is verified with only Cisco equipment. This solution may also work with other
third-party PSTN gateways and Session Border Controllers (SBCs), but the features might not work as
described here. If you are using this solution with third-party PSTN gateways or SBCs, Cisco technical support
may not be able to resolve problems that you encounter.
Cisco Unified Mobility Configuration Task Flow

Complete these tasks to configure Mobility features for your deployment.
Procedure
Command or Action Purpose

Step 1 Perform one of the following: Adds mobility features for an individual end
user.
• Configure a Mobility User, on page 13
• Configure Mobility Users Through Bulk Configures Mobility features for a large number
Administration, on page 14 of existing end users, use the Bulk
• Provision Mobility Users Through LDAP, Administration Tool.
on page 14 Provisions new users with mobility
functionality, you can use a feature group
template and LDAP sync.
Step 2 Configure Mobility for IP Phones, on page 15 Configures Cisco IP Phones for Mobility
including setting up the Single Number Reach
(SNR) and Move to Mobile features. This
allows enterprise phone users to extend
enterprise calls to a wide range of mobile
devices, including a home office phone or a
mobile phone.
Step 3 Configure Mobile Voice Access, on page 20 Optional. Provides a system IVR so that mobile
users can call from any mobile device and have
the call that is presented to the called party as
if the caller were dialing from their enterprise
desk phone.
Step 4 Configure Enterprise Feature Access, on page Optional. Provides two-stage dialing from a
27 configured remote destination and have the call
that is presented to the called party as if it
originated from a desk phone. This feature also
allows you to access mid-call features while on
a call from a remote destination.
Step 5 Configure Intelligent Session Control, on page Configure the system so that inbound calls to a
28 remote destination are rerouted to an associated
enterprise, if one is available. This provides
automatic call anchoring within the enterprise
12
Configure a Mobility User

for mobility calls, providing cost savings and
added Unified Communications functionality.
Step 6 Configure Mobility Service Parameters, on page Optional. Configure optional mobility-related
29 service parameters if you want to change the
behavior of Cisco Unified Mobility.
Step 7 Configure Cisco Jabber Dual-Mode, on page Configure Cisco Jabber for mobility so your
29 users can access enterprise communications
features through a Jabber client on their
smartphone.
Step 8 Configure Other Dual-Mode Devices, on page Complete this task flow if you want to deploy
30 other dual-mode devices, such as FMC or IMS
clients that can connect through Wi-Fi.
Configure a Mobility User

Use this procedure to configure an end user with the mobility feature.
Procedure
Step 1 From Cisco Unified CM Administration, choose User Management > End User.
Step 2 In Find and List Users window, perform one of the following tasks:
• Click Find and select an existing user to modify the settings.
• Click Add New to configure a new user.
Step 3 Configure values for the following fields:

• User ID
• Last Name
Step 4 In the Mobility Information area, complete the following fields:

a) Check the Enable Mobility check box.
b) Optional. Check the Enable Mobile Voice Access check box to allow this user to use Mobile Voice
Access.
c) In the Maximum Wait Time for Desk Pickup field, enter a value in milliseconds. After hanging up a
call from a remote destination, this timer represents the amount of time where the user still has the option
of resuming the call from a deskphone.
d) In the Remote Destination Limit field, enter the number of remote destinations that a user is permitted
to have for single number reach (SNR) targets.
Step 5 Complete the remaining fields in the End User Configuration window. For more information on the fields
and their configuration options, see Online Help.
Step 6 Click Save.
13
Configure Mobility Users Through Bulk Administration
Configure Mobility Users Through Bulk Administration

Use this procedure to use Bulk Administration's Update Users menu to add the Mobility feature to existing
end users by bulk.
Note Bulk Administration contains other features that allow you to update existing users by bulk. For example,
you can use the Export and Import functions to import a CSV file with the new Mobility settings. For more
information, see the Bulk Administration Guide for Cisco Unified Communications Manager.
Procedure
Step 1 From Cisco Unified CM Administration, choose Bulk Administration > Users > Update Users > Query.
Step 2 Apply the filter and click Find to select the users whom you want to assign as mobility users.
Step 3 Click Next.
Step 4 In the Mobility Information area, modify the following four fields by first checking the check box on the
far left to indicate that this field is to be updated, and then configuring the setting on the right as follows:
• Enable Mobility—Check this check box to enable the users provisioned with this template for Mobility
features.
• Enable Mobile Voice Access—Check this check box for provisioned users to be able to use Mobile
Voice Access.
• Maximum Wait Time for Desk Pickup—This field represents the amount of time, after hanging up a
call on a mobile phone, that you have to resume the call on your desk phone.
• Remote Destination Limit—This field represents the number of Remote Destinations and Mobile
Identities that you can assign to users whom are provisioned through this template.
Step 5 Under Job Information, check Run Immediately.

Step 6 Click Submit.
Provision Mobility Users Through LDAP

If you have not yet synced your LDAP directory, you can use this procedure to configure synced end users
with mobility capability through the Feature Group Template configuration. Newly synced users inherit the
mobility settings from the template.
Note This method works only if you have not yet synced your LDAP directory. You cannot assign new feature
group template configurations to an LDAP directory sync after the initial sync has occurred.
14
Configure Mobility for IP Phones
Procedure
Step 1 From Cisco Unified CM Administration, choose User Management > User/Phone Add > Feature Group
Template.
Step 2 In the Find and List Feature Group Templates window, perform one of the following:
• Click Add New to configure a new template.
• Click Find and select an existing template to configure.
Step 3 Assign a Name to the template.

Step 4 Configure the following Mobility fields:
• Enable Mobility—Check this check box to enable the users provisioned with this template for Mobility
features.
• Enable Mobile Voice Access—Check this check box for provisioned users to be able to use Mobile
Voice Access.
• Maximum Wait Time for Desk Pickup—This field represents the amount of time in milliseconds, after
hanging up a call on a mobile phone, that you have to resume the call on your deskphone.
• Remote Destination Limit—This field represents the number of Remote Destinations and Mobile
Identities that you can assign to users whom are provisioned through this template.
Step 5 Configure the remaining fields in the Feature Group Template Configuration window. For more information
on the fields and their configuration options, see Online Help.
Step 6 Click Save.
Note Assign the configured Feature Group Template to an LDAP Directory that has not yet been
synced. Newly synced users have Mobility enabled. For more information, on provisioning users
through LDAP see "Provisioning End Users" chapter in System Configuration Guide for Cisco
Unified Communications Manager.
Configure Mobility for IP Phones

Complete these tasks to configure mobility features for Cisco IP Phones. This includes setting up Single
Number Reach (SNR) and the Move To Mobile feature. This provides users with a single enterprise number
that rings all their devices, in addition to an enterprise-level voicemail that can be reached no matter which
device rings. And also, users are able to transfer active calls between their deskphone and mobile device.
Procedure

Step 1 Configure Softkey Template for Mobility, on Configures a mobility softkey template for
page 16 Cisco IP Phones that includes the Mobility
softkey. Users can transfer calls from their
deskphone to a mobile phone by pressing the
softkey.
15
Configure Softkey Template for Mobility

Step 2 Configure IP Phone for Mobility, on page 18 Configures an IP phone for mobility so that
incoming calls to an enterprise number are
extended to remote destinations.
Step 3 Configure a Remote Destination Profile, on Configures common settings that you want to
page 18 apply to all the remote destination numbers for
a user.
Step 4 Configure a Remote Destination, on page 19 Configures a remote destination that is a virtual
device that represents a mobile device where
the user can be reached (for example, a home
office phone, or a mobile phone on a cellular
network). The remote destination carries many
of the same settings as the user's desk phone.
Step 5 Configure an Access List, on page 20 Optional. Controls which calls can ring which
remote destinations, and at which times of day.
The access list filters callers based on the Caller
ID and can either allow calls or block calls from
the caller during that remote destination's ring
schedule.
Configure Softkey Template for Mobility

Use this procedure to configure a softkey template that includes the Mobility softkey. The softkey will be
enabled for all phones that use this template.
Procedure
Step 1 From Cisco Unified CM Administration, choose Device > Device Settings > Softkey Template.
Step 2 To create a new softkey template do the following. Otherwise, proceed to the next step.
a) Click Add New.
b) Select a default template and click Copy.
c) In the Softkey Template Name field, enter a new name for the template.
d) Click Save.
Step 3 To add mobility softkeys to an existing template.
a) Enter search criteria and click Find.
b) Choose an existing template.
Step 4 (Optional) Check the Default Softkey Template check box if you want to designate this softkey template as
the default softkey template.
Note If you designate a softkey template as the default softkey template, you cannot delete it unless
you first remove the default designation.
Step 5 Click Save.

Step 6 From the Related Links drop-down list, choose Configure Softkey Layout and click Go.
16
Enable Mobility Within Feature Control Policy
Step 7 From the Select a Call State to Configure drop-down list, choose the call state for which you want to add
the softkey. Typically, you will want to add the softkey for both the OnHook and Connected call states.
Step 8 From the Unselected Softkeys list, choose the Mobility softkey and use the arrows to move the softkey to
the Selected Softkeys list. Use the up and down arrows to change the position of the new softkey.
Step 9 To display the softkey in additional call states, repeat the previous step.
Step 10 Click Save.
Note If you created a new softkey template, you can assign the template to a phone through the Phone
Configuration window or to a group of phones through Bulk Administration's Update Phones
menu.
There are several methods to assign softkey template to phones during provisioning. For example,
you can use the Universal Device Template configuration, or you can assign it as the default
device profile for a specific model.
Enable Mobility Within Feature Control Policy

If you have configured feature control policies to enable or disable features for Cisco IP Phones, then you
will also have to enable Mobility within the policy that is used by your Cisco IP Phones. If the feature is
disabled within the feature control policy configuration that is used by your phones, then the Mobility softkey
will be disabled for all Cisco IP Phones that use that policy.
Procedure
Step 1 From Cisco Unified CM Administration, choose Device > Device Settings > Feature Control Policy.
Step 2 Click Find and choose the applicable policy.
Note You can also choose Add New if you want to create a new feature control policy that you assign
to your phones to enable mobility, along with other associated features. You can assign the policy
to phones through the Phone Configuration window, or to a set of phones through the Common
Phone Profile Configuration. You can also assign the policy to a universal device template to
assign the policy to phones as you provision them.
Step 3 In the Name field, enter a name for the feature control policy. The name can include up to 50 alphanumeric
characters and can contain any combination of spaces, periods (.), hyphens (-), and underscore characters (_).
Ensure that each feature control policy name is unique to the system.
Step 4 In the Description field, enter a brief description for the feature control policy. The description can include
up to 50 alphanumeric characters and can contain any combination of spaces, periods (.), hyphens (-), and
underscore characters (_).
Step 5 In the Feature Control area, check both the Override Default check box and the Enable Setting check box
that corresponds to the Mobility softkey.
Step 6 Click Save.
17
Configure IP Phone for Mobility
Configure IP Phone for Mobility

If you have Single Number Reach or Move to Mobility configured, use this procedure to configure your desk
phone with the Mobility feature so that enterprise calls can be redirected to a remote destination.
Procedure
Step 1 From Cisco Unified CM Administration, choose Device > Phone.

Step 2 Perform one of the following tasks:
• Click Find and select an existing phone to modify the settings.
• Click Add New and choose a phone from the Phone type drop-down list to add a new phone.
Step 3 Click Next.

Step 4 From the SoftKey Template drop-down list, choose the mobility softkey template that you configured.
Step 5 From the Owner User ID drop-down list, choose the user account on which you enabled mobility.
Note You can configure either the Owner User ID or Mobility User ID field. Mobility users are
configured for mobility-enabled devices and Owner users are configured for Non-Mobility devices.
Configuring both users for the same device is not recommended.
Step 6 (Optional) If you are using a Feature Control Policy to enable features, choose the policy from the drop-down
list.
Step 7 Click Save.
Configure a Remote Destination Profile

Configures common settings that you want to apply to all the remote destination numbers for a user.
Procedure
Step 1 From Cisco Unified CM Administration, choose Device > Device Profile > Remote Destination Profile.
Step 2 Click Add New.
Step 3 Enter a Name for the profile.
Step 4 From the User ID drop-down list, choose the end user to whom this profile applies.
Step 5 From the Device Pool drop-down list, select the device pool where this profile should reside.
Step 6 Configure the remaining fields in the Remote Destination Profile Configuration window. For more
information on the fields and their configuration options, see Online Help.
Step 7 Click Save.
Step 8 Under Association Information, click Add a New DN.
Step 9 In the Directory Number field, add the directory number of the user's desk phone.
18
Configure a Remote Destination
Configure a Remote Destination

A remote destination is a virtual device that represents a mobile device where the user can be reached (for
example, a home office phone, a mobile phone on a cellular network, or a PSTN phone). The remote destination
carries many of the same settings as the user's desk phone.
Note • When an enterprise user initiates a call from a remote destination to Cisco Jabber, Unified Communications
Manager tries to establish a data call with Cisco Jabber by sending an INVITE message to Cisco
TelePresence Video Communication Server (VCS). The call is established regardless of receiving a
response from VCS.
• If you have Self-Provisioning enabled, your end users can provision their own phones from the Self-Care
Portal. See the System Configuration Guide for Cisco Unified Communications Manager and the
"Configure Self-Provisioning" chapter for details on configuring the system for self-provisioning and
the "Provisioning End Users" part for details on enabling self-provisioning for users as a part of a User
Profile.
Procedure
Step 1 From Cisco Unified CM Administration, choose Device > Remote Destination.
Step 3 In the Destination field, enter the number of the remote destination. For example, this could be a cellular
number or PSTN number.
Step 4 From the Mobility User ID field, select the mobility-enabled end user who uses this remote destination.
Step 5 Check the Enable Unified Mobility features check box.
Step 6 From the Remote Destination Profile drop-down list, choose the profile that you set up for the user who
owns this remote destination.
Step 7 Use the Single Number Reach Voicemail Policy drop-down list to configure the voicemail policy.
a) Check the Enable Single Number Reach check box.
b) Check the Enable Move to Mobile check box to include this remote destination to the list of available
destinations when the user presses the Mobility softkey on their desk phone.
Step 8 (Optional) If you want to limit enterprise calls to this remote destination to specific periods such as office
hours, configure a Ring Schedule.
Step 9 In the When receiving a call during the above ring schedule area, apply the list that is configured for this
remote destination.
Step 10 Configure the remaining fields on the Remote Destination Configuration window. For more information
Step 11 Click Save.
19
Configure an Access List
Configure an Access List

An access list is an optional remote destination configuration if you want to control which calls can ring which
remote destinations, and at which times of day. The access list filters callers based on the Caller ID and can
either allow calls or block calls during that remote destination's ring schedule.
Procedure
Step 1 From Cisco Unified CM Administration, choose Call Routing > Class of Control > Access List.
Step 2 Click Add New to create an access list.
Step 3 Enter a name and description to identify the new access list.
Step 4 Associate the access list to a user by choosing an ID from the Owner drop-down list.
Step 5 Choose one of the following options:
• Allowed—All numbers in the access list are allowed.
• Blocked—All numbers in the access list are blocked.
Step 6 Click Save.

Step 7 From the Filter Mask drop-down list, choose the filters that you want to apply to the access list:
• Not Available—All callers that advertise a not available status are added to the access list.
• Private—All callers that advertise a private status are added to the access list.
• Directory Number—All directory numbers or directory strings that you specify are added to the access
list. If you choose this option, add a number or number string in the DN Mask field.
Step 8 Choose Save.

Step 9 Apply the access list to a remote destination:
a) From Cisco Unified CM Administration, choose Device > Remote Destination and reopen the remote
destination that you created.
b) Configure the Ring Schedule for this access list and do either of the following:
• If you created an allowed access list, click the Ring this destination only if caller is in radio button
and choose the access list that you created from the drop-down list.
• If you created a blocked access list, click the Do not ring this destination if caller is in radio button
and choose the access list that you created from the drop-down list.
c) Click Save.
Configure Mobile Voice Access

Complete the following tasks to configure the system for Mobile Voice Access, which lets users place
enterprise-anchored calls from any device. Users dial a system IVR for authentication, following which the
call is sent out as an enterprise call that will appear to the end user as if the call were sent from the office
phone.
20
Configure Mobile Voice Access
Before you begin

To use Mobile Voice Access:
• Users must be enabled as mobility users with the Enable Mobile Voice Access option checked within
End User Configuration For details, see Configure a Mobility User, on page 13.
• Interactive Voice Response service must be active, and included in a Media Resource Group List that
the trunk uses.
Procedure

Step 1 Activate the Cisco Unified Mobile Voice In Cisco Unified Serviceability, make sure that
Access Service, on page 22 the Cisco Unified Mobile Voice Access feature
service is activated.
Step 2 Enable Mobile Voice Access, on page 22 Enable the Mobile Voice Access feature and
specify a directory number that users can dial
to reach the enterprise.
Step 3 Configure Directory Number for Mobile Voice Configure mobile voice access (MVA) to assign
Access, on page 23 sets of localized prompts for users who dial in
from outside the enterprise.
Step 4 Restart Cisco CallManager Service, on page 23 After you activate Mobile Voice Access, restart
the Cisco CallManager service.
Step 5 Configure a gateway for legacy MVA or Note Gateway configuration is no

enterprise feature access (EFA) by performing longer mandatory for Mobile
one of the following tasks: Voice Access. This is an optional
configuration only if you want to
• Configure an Existing H.323 or SIP
configure legacy Mobile Voice
Gateway for Remote Access, on page 24
Access through an ISR G2 router.
• Configure a New H.323 Gateway for
Remote Access, on page 25 Depending on your system requirements, you
can add a new gateway or configure an existing
gateway to handle calls that come from outside
the enterprise through MVA or EFA.
If you have an existing H.323 or SIP PSTN
gateway in your system, you can configure it
for MVA. This function is accessed by calling
a system-configured DID number that is
answered and handled by an H.323 or SIP
VoiceXML (VXML) gateway. After you
configure your gateway, it uses a vxml script
on the publisher node to pull the interactive
voice response (IVR) prompts that are played
to the MVA users. These prompts request user
authentication and input of a number that users
must dial on their phone keypad.
21
Activate the Cisco Unified Mobile Voice Access Service

If you do not have an existing H.323 or SIP
PSTN gateway and you want to configure
mobile voice access, you must add a new H.323
gateway and configure it for MVA functionality
by using the hairpinning method. From a
technical standpoint, this method refers to using
a second gateway to receive the inbound call,
apply the MVA service and then the inbound
call leg returns to the PSTN gateway (original
source) after the system applies the MVA
service.
Activate the Cisco Unified Mobile Voice Access Service

Use the following procedure to activate this service in your publisher node.
Procedure
Step 1 From Cisco Unified Serviceability, choose Tools > Service Activation.
Step 2 From the Server drop-down list, choose the publisher node.
Step 3 Click Go.
Step 4 Under CM Services, check the Cisco Unified Mobile Voice Access Service check box.
Step 5 Click Save.
Enable Mobile Voice Access

Configure service parameters to enable Mobile Voice Access (MVA) and to specify the directory number or
PSTN DID number that users can dial in order to reach the IVR.
Before you begin

The Cisco Unified Mobile Voice Access feature service must be activated for Mobile Voice Access to work.
Procedure
Step 1 From Cisco Unified CM Administration, choose System > Service Parameters.
Step 2 From the Server drop-down list, choose publisher node.
Step 3 From the Service drop-down list, choose Cisco CallManager.
Step 4 Configure the following service parameters:
• Enable Mobile Voice Access—Set this parameter to True.
• Mobile Voice Access Number—Enter the access number that you want users to dial when they access
the enterprise.
22
Configure Directory Number for Mobile Voice Access
Step 5 Click Save.
Configure Directory Number for Mobile Voice Access

Configure mobile voice access (MVA) to assign sets of localized prompts for users who dial in from outside
the enterprise.
Procedure
Step 1 From Cisco Unified CM Administration, choose Media Resources > Mobile Voice Access.
Step 2 In the Mobile Voice Access Directory Number, enter the internal directory number (DN) to receive Mobile
Voice Access calls from the gateway.
Enter a value between 1-24 digits in length. Valid values are 0-9.
Step 3 In the Localization pane, use the arrows to move the locales that you want to select to or from this pane.
Note Mobile Voice Access uses the first locale that appears in the Selected Locales pane in the Mobile
Voice Access window. For example, if English United States appears first in the Selected Locales
pane, the Cisco Unified Mobility user hears English when the IVR is used during a call.
Step 4 Click Save.
Restart Cisco CallManager Service

After you enable the Mobile Voice Access feature, restart the Cisco CallManager service.
Procedure
Step 1 From Cisco Unified Serviceability, choose Tools > Control Center - Feature Services
Step 2 From the Server drop-down list, choose the Cisco Unified Communications Manager publisher node.
Step 3 Under CM Services, select the radio button that corresponds to the Cisco CallManager service.
Step 4 Click Restart.
What to do next
You have now completed all the tasks that are required to configure Unified Communications Manager with
native Mobile Voice Access support. However, if you want to configure legacy Mobile Voice Access where
an ISR G2 router provides the IVR and voice prompts, you can complete either of the following two optional
tasks:
• Configure an Existing H.323 or SIP Gateway for Remote Access, on page 24
• Configure a New H.323 Gateway for Remote Access, on page 25
23
Configure an Existing H.323 or SIP Gateway for Remote Access
Configure an Existing H.323 or SIP Gateway for Remote Access

If you have an existing H.323 or SIP PSTN gateway in your system, you can configure it for MVA. This
function is accessed by calling a system-configured DID number that is answered and handled by an H.323
or SIP VoiceXML (VXML) gateway. After you configure your gateway, it uses a vxml script on the publisher
node to pull the interactive voice response (IVR) prompts that are played to the MVA users. These prompts
request user authentication and input of a number that users must dial on their phone keypad.
Before you begin

Configure Directory Number for Mobile Voice Access, on page 23
Procedure
Step 1 Configure the T1/E1 controller for PRI from the PSTN.
Example:
controller T1 1/0
framing esf
linecode b8zs
pri-group timeslots 1-24
Step 2 Configure the serial interface for the PRI (T1/E1).

Example:
interface Serial 1/0:23
ip address none
logging event link-status none
isdn switch-type primary 4ess
isdn incoming-voicevoice
isdn bchan-number-order ascending
no cdp enable
Step 3 Load the VXML application from the publisher node.

Example:
Sample configuration for IOS Version 12.3 (13) and later:
application service CCM
http://<Unified CM Publisher IP Addr>:8080/ccmivr/pages/IVRMainpage.vxml
Example:
Sample configuration before IOS Version 12.3(12):
call application voice Unified CCM
Caution Although VXML was added in Version12.2(11), other versions such as 12.3(8), 12.3(9),
12.3(14)T1, and 12.2(15) have VXML issues.
Step 4 Configure the dial peer to associate the Cisco Unified Mobility application with system remote access.
Example:
Sample configuration for IOS 12.3(13) and later:
24
Configure a New H.323 Gateway for Remote Access
dial-peer voice 58888 pots

service CCM (Cisco Unified Mobility VXML application)
incoming called-number 58888
Example:
Sample configuration for IOS 12.3(12) and earlier:
dial-peer voice 100 pots
application CCM (Cisco Unified Mobility VXML application)
(58888 represents the mobile voice access (MVA) number)

Step 5 Add a dial peer to transfer the calls to the MVA DN.
Example:
Sample configuration for primary Unified Communications Manager:
dial-peer voice 101 voip
preference 1
destination-pattern <Mobile Voice Access DN>
session target ipv4:10.1.30.3
codec g711ulaw
dtmf-relay h245-alphanumeric
no vad
Example:
Sample configuration for secondary Unified Communications Manager (if needed):
preference 2
codec g711ulaw
no vad
Note If a generic dial peer is already configured to terminate the calls and is consistent with the MVA
DN, you do not need to perform this step.
Example:
Sample configuration for SIP gateway VoIP dial-peer:
rtp payload-type nse 99
session protocol sipv2
incoming called-number .T
dtmf-relay rtp-nte
codec g711ulaw

If you do not have an existing H.323 or SIP PSTN gateway and you want to configure mobile voice access,
you must add a new H.323 gateway and configure it for MVA functionality by using the hairpinning method.
From a technical standpoint, this method refers to using a second gateway to receive the inbound call, apply
25
the MVA service and then the inbound call leg returns to the PSTN gateway (original source) after the system
applies the MVA service.
Note If you use Mobile Voice Access with hairpinning, users calling into your system will not be identified
automatically by their caller ID. Instead, users must enter their remote destination number manually before
they enter their PIN. The reason is that the PSTN gateway must first route the call to Unified Communications
Manager to reach the hairpinned Mobile Voice Access gateway. Because of this route path, the conversion
of the calling number from a mobile number to an enterprise directory number occurs before the Mobile Voice
Access gateway handles the call. As a result, the gateway is unable to match the calling number with a
configured remote destination, and therefore the system prompts users to enter their remote destination number.
Before you begin

Configure Directory Number for Mobile Voice Access, on page 23
Procedure
Step 1 Load the VXML application from the publisher node.

Example:
Sample configuration for IOS Version 12.3 (13) and later:
application service CCM
Example:
Sample configuration before IOS Version 12.3(12):
call application voice CCM
Caution Although VXML was added in Version 12.2(11), other versions such as 12.3(8), 12.3(9),
12.3(14)T1, and 12.2(15) have VXML issues.
Step 2 Configure the dial-peer to associate the Cisco Unified Mobility application with system remote access.
Example:
Sample configuration for IOS 12.3(13) and later:
service CCM
codec g711u
session target ipv4:<ip_address of call manager>
Example:
Sample configuration for IOS 12.3(12) and earlier:
application CCM
codec g711u
session target ipv4:<ip_address of call manager>
26
Configure Enterprise Feature Access
Step 3 Add a dial-peer for transferring calls to the Mobile Voice Access (MVA) DN.
Example:
Sample configuration for primary Unified Communications Manager:
preference 1
voice-class h323 1
codec g711ulaw
novad
Example:
Sample configuration for secondary Unified Communications Manager (if needed):
preference 2
voice-class h323 1
codec g711ulaw
novad
Note If a generic dial peer is already configured to terminate the calls and is consistent with the MVA
DN, you do not need to perform this step.
Step 4 Configure hairpin.

voice service voip
allow-connections h323 to h323
Step 5 On the Unified Communications Manager, create a new route pattern to redirect the incoming MVA number
to the H.323 gateway that has the vxml script loaded. Ensure that the incoming CSS of the gateway can access
the partition in which the new route pattern is created.
Configure Enterprise Feature Access

Use the following procedure to configure Enterprise Feature Access from a remote destination for:
• Two-stage dialing to place enterprise calls from a configured remote destination. Calls appear to the
called party as if they were placed from an associated desk phone.
• Remote destination access to mid-call features through EFA codes that are sent using DTMF digits sent
from the remote destination.
Note Unlike Mobile Voice Access, with Enterprise Feature Access you must be calling from a configured remote
destination.
27
Configure Intelligent Session Control
Procedure
Step 1 From Cisco Unified CM Administration, choose Call Routing > Mobility > Enterprise Feature Access
Number Configuration.
Step 2 In the Number field, enter the unique DID number that mobile users will dial from a remote destination in
order to access the Enterprise Feature Access feature.
Step 3 From the Route Partition drop-down list, choose the partition where the DID resides.
Step 4 (Optional) Check the Default Enterprise Feature Access Number check box to make this EFA number the
default for this system.
Step 5 Click Save.
Step 6 Configure the Enteprise Feature Access service parameters:
a) From Cisco Unified CM Administration, choose System > Service Parameters.
b) From the Server drop-down list, choose the publisher node.
c) From the Service drop-down list, choose Cisco CallManager.
d) Set the Enable Enterprise Feature Access service parameter to True.
e) (Optional) In the Clusterwide Parameters (System - Mobility) area, edit the DTMF digits that you must
enter to access midcall features through Enterprise Feature Access. For example, you could edit the
Enterprise Feature Access Code for Hold service parameter, which has a default value of *81. The
default values are as follows:
• Hold: *81
• Exclusive Hold: *82
• Resume: *83
• Transfer: *84
• Conference: *85
• Session Handoff: *74
• Starting Selective Recording: *86
• Stopping Selective Recording: *87
• Hunt group login—enter a new code
• Hunt group logout—enter a new code
f) Click Save.
Configure Intelligent Session Control

Configure the system so that inbound calls to a remote destination are rerouted to an associated enterprise
number, if one is available. This provides automatic call anchoring within the enterprise for mobility calls,
providing cost savings and added Unified Communications functionality.
28
Configure Mobility Service Parameters
Procedure
Step 2 From the Server drop-down list, choose a Cisco Unified Communications Manager node.
Step 4 Under Clusterwide Parameters (Feature - Reroute Remote Desination Calls to Enterprise Number) set
the following service parameters:
• Reroute Remote Destination Calls to Enterprise Number—To enable Intelligent Session Control, set
this parameter to True.
• Ring All Share Lines—Set the value of the parameter to True. If Intelligent Session Control is enabled,
and this service parameter is also enabled, the system anchor calls to remote destinations within the
enterprise, and will also ring all the user's shared lines.
• Ignore Call Forward All on Enterprise DN—This parameter applies only to outgoing calls to a remote
destination when Intelligent Session Control is enabled. By default, this parameter is set to True.
Step 5 Click Save.
Configure Mobility Service Parameters

Use this procedure to configure optional Mobility-related service parameters.
Procedure
Step 4 Configure any service parameters that you want to edit. The Mobility-related parameters are listed under the
following headings. For help descriptions, click the parameter name:
• Clusterwide Parameters (System - Mobility)
• Clusterwide Parameters (System - Mobility Single Number Reach Voicemail)
• Clusterwide Parameters (Feature - Reroute Remote Destination Calls to Enterprise Number)
Step 5 Click Save.
Configure Cisco Jabber Dual-Mode

Complete these tasks to configure Cisco Jabber on iPhone or Android as dual-mode mobile devices that can
connect over WiFi. Cisco Jabber registers to Unified Communications Manager over WiFi and can be reached
through an enterprise number if Single Number Reach is enabled in the user's mobile identity.
29
Configure Other Dual-Mode Devices
Procedure

Step 1 Configure a Mobility Profile, on page 31 Configure a mobility profile to send consistent
caller ID to Jabber mobile clients that are
placing Dial through Office calls.
Step 2 Add a Dual-Mode Device for Cisco Jabber, on Configure a dual-mode device type for Cisco
page 31 Jabber on iPhone or Android clients.
Step 3 Configure a Mobility Identity, on page 34 Add a Mobility Identity to the Jabber mobile
client that points to the device phone number
(that is, the iPhone number) to provide calling
when Jabber roams out of WiFi range. Enable
Single Number Reach destination for the Mobile
Identity.
Step 4 Required: Configure Handoff Number, on page Configure a handoff number for dual-mode
34 devices that are leaving enterprise. Even when
the device disconnects from the enterprise WiFi
network the call can be maintained without
interruption by reconnecting to a remote mobile
or cellular network.
Configure Other Dual-Mode Devices

Complete these tasks to configure other dual-mode mobile devices that can place calls over the cellular network
and can also connect over WiFi. For example:
• Carrier-Integrated Mobile Devices that connect over Fixed Mobile Convergence (FMC) networks.
• IMS-integrated Mobile Devices over IP Multimedia networks
Procedure

Step 1 Add a Dual-Mode Device for Cisco Jabber, on Configure an IMS or FMC dual-mode device.
page 31
Step 2 Configure a Mobility Identity, on page 34 Add a Mobility Identity that points to the phone
number of the actual device.
Step 3 Required: Configure Handoff Number, on page Configure a handoff number for dual-mode
34 devices that are leaving the enterprise. Even
when the device disconnects from the enterprise
WiFi network the call can be maintained
without interruption by reconnecting to a remote
mobile or cellular network.
30
Configure a Mobility Profile
Configure a Mobility Profile

Configure a mobility profile for dual-mode Cisco Jabber on iPhone and Android clients. The profile configures
the client with a consistent caller ID for dial via office calls.
Note From a technical standpoint, this caller ID is sent during the dial via office reverse (DVO-R) callback portion
of a call to the mobility identity or alternate callback number. DVO-R call feature uses enbloc dialing. If no
mobility profile is assigned to the mobility identity or if the Callback Caller ID field is left blank, the system
sends the default enterprise feature access number.
Procedure
Step 1 From Cisco Unified CM Administration, choose Call Routing > Mobility > Mobility Profile.
Step 3 Enter a Name for the profile.
Step 4 From the Mobile Client Calling Option drop-down list, select Dial via Office Reverse.
Note Despite the field options, Dial via Office Forward is not available.
Step 5 Configure a Callback Caller ID for Dial-via-Office Reverse.

Step 6 Configure the fields in the Mobility Profile Configuration window. For more information on the fields and
their configuration options, see Online Help.
Step 7 Click Save.
Add a Dual-Mode Device for Cisco Jabber

Use the following procedure to configure a dual-mode device type for Cisco Jabber on iPhone or Android
clients.
Before you begin

Make sure that your end users are mobility-enabled. Also, if you want to add remote destinations to your
Jabber client, make sure that you have a softkey template that includes the Mobility softkey.
Procedure

Step 2 Perform one of the following:
• Click Find to edit an existing device.
• Click Add New and select either Cisco Dual Mode for Android or Cisco Dual Mode for iPhone as
the phone model, to add a new device. Click Next.
Step 3 Configure the fields in the Phone Configuration window.
31
Dual-Mode Device Configuration Fields
For detailed information about product specific configuration layout fields, see your Jabber client documentation
at http://www.cisco.com/go/jabber.
Step 4 Configure the following mandatory fields:

• Device Name
• Device Pool
• Softkey Template
• Owner User ID—The user must have mobility enabled.
• Mobility User ID—The user must have mobility enabled.
• Device Security Profile
• SIP Profile
Step 5 Click Save.

Step 6 Add a directory number:
a) In the left Association area, click Add a New DN.
b) Enter a new Directory Number and click Save.
c) Complete any fields that you want in the Directory Number Configuration window and click Save. For
more information on the fields and their configuration options, see Online Help.
d) Click Associate End Users .
e) Click Find and select the mobility-enabled end user whom owns this DN.
f) Click Add Selected.
g) Click Save.
What to do next
Add a Mobility Identity that points to the phone number of the iPhone or Android device. This allows you to
transfer the call to the phone if you move out of Wi-Fi range. You can also add the device as a Single Number
Reach destination. For details, Configure a Mobility Identity, on page 34.
Optionally, add Remote Destinations and Single Number Reach to your Cisco Jabber client. When someone
calls the Jabber client, the remote destination rings as well. Configure a Remote Destination, on page 19.
Dual-Mode Device Configuration Fields

Table 1: Dual-Mode Device Configuration Fields
Field Description
Softkey Template Choose the Mobility Softkey template.
Owner User ID Choose the user ID of the assigned phone user. The user ID is recorded in the
call detail record (CDR) for all calls made from this device.
Mobility User ID Choose the user ID of the person to whom this dual-mode phone is assigned.
32
Add Other Dual-Mode Device
Field Description
Device Security Profile Choose the security profile to apply to the device.
You must apply a security profile to all phones that are configured in Cisco
Unified Communications Manager Administration. To enable security features
for a phone, you must configure a new security profile for the device type and
protocol, and then apply it to the phone.
Rerouting Calling Search Choose a calling search space for routing calls to configured remote destinations
Space and mobility identities that are configured for this device.
SIP Profile Choose Standard SIP Profile for Mobile Device.
Add Other Dual-Mode Device

Use this procedure to add another dual-mode device (for example, a Carrier-integrated Mobile Device for
network-based FMC, or an IMS-integrated Mobile Device.
Before you begin

Make sure that your end users are mobility-enabled. Refer to topics earlier in this chapter for details on how
to enable mobility for users.
Procedure

Step 3 From the Phone Model drop-down list Carrier-integrated Mobile Device or IMS-integrated Mobile Device.
Step 4 Configure the following mandatory fields:
• Device Name
• Device Pool
• Owner User ID—The user must have mobility enabled.
• Mobility User ID—The user must have mobility enabled.
Step 5 Configure the remaining fields in the Phone Configuration window. For more information on the fields and
Step 6 Click Save.
Step 7 Add a directory number:
a) In the left Association area, click Add a New DN.
b) Enter a new Directory Number and click Save.
c) Complete any fields that you want in the Directory Number Configuration window and click Save. For
d) Click Associate End Users.
e) Click Find and select the mobility-enabled end user whom owns this DN.
f) Click Add Selected.
33
Configure a Mobility Identity
g) Click Save.
Configure a Mobility Identity

Add a Mobility Identity that points to the phone number of the device if you want to enable the device as a
Single Number Reach that can be reached through the enterprise number.
Procedure

Step 2 Enter search criteria if needed, click Find, and choose the dual-mode device that you created.
Step 3 Click Add New Mobility Identity.
Step 4 In the Destination field, enter the phone number of the mobile device. For example, for a Cisco Jabber on
iPhone client, this would be the phone number of the iPhone.
Step 5 Cisco Jabber only. Select the Mobility Profile that you configured.
Step 6 If you want to make this Mobile Identity available from an enteprise phone number:
a) Check the Enable Single Number Reach check box.
b) Configure a Single Number Reach Voicemail policy
Step 7 Configure a Dial-via-Office Reverse Voicemail policy.
Step 8 Configure the fields on the Mobility Identity Configuration window. For more information on the fields
Step 9 Click Save.
Note If you want to apply a Ring Schedule and access list to limit calls to this mobile identity to specific
times and users, Configure an Access List, on page 20.
Configure Handoff Number

Configure handoff mobility for dual-mode phones if you want your system to preserve a call while the user
moves out of the enterprise. Even when a user's device disconnects from the enterprise WiFi network and
reconnects to the mobile voice or cellular network, an in-progress call is maintained without interruption.
Procedure
Step 1 From Cisco Unified CM Administration, choose Call Routing > Mobility > Handoff Configuration.
Step 2 In the Handoff Number field, enter the direct inward dialing (DID) number for handoff between the Wi-Fi
and mobile voice or cellular network.
For numbers that start with the international escape character (+), you must precede the + with a backslash
(\). Example: \+15551234.
Step 3 From the Route Partition drop-down list, choose the partition to which the handoff DID number belongs.
34
Cisco Unified Mobility Call Flow
Step 4 Click Save.
Cisco Unified Mobility Call Flow

This section describes the incoming and outgoing call flows of Cisco Unified Mobility commonly known as
Single Number Reach (SNR). Unified Communications Manager supports the separate calling party number
and billing number feature when SNR is configured for users to allow desk phones to extend calls to mobile
devices.
For example, User-A calls from a PSTN network to User-B whose directory number configured to SNR. If
Enable External Presentation Name and Number check box is checked in SIP profile and Display External
Presentation Name and Number service parameter value set to True, then Unified Communications Manager
displays the FROM header information on both the User-B’s desk phone and the configured remote destination
device. In the same way, if any one option is disabled, Unified Communications Manager displays
P-Asserted-Identity (PAID) header information on the called device.
Similarly, in outgoing call scenario User B (SNRD line) configured with External Presentation Information
on Directory Number configuration page initiates a call to a PSTN network through a SIP trunk. If Enable
External Presentation Name and Number is configured in its SIP profile, then, Unified Communications
Manager send the External Presentation Information in the FROM header of the outgoing SIP message to
display on the called device.
If Enable External Presentation Name and Number check box is disabled, then Unified Communications
Manager sends the directory number information in the FROM and PAID to display on the called device and
configured External Presentation Information in the X-Cisco-Presentation header.
If you check the Anonymous External Presentation check box, the configured External Presentation Name
and External Presentation Number are removed from the respective fields and external presentation displayed
as anonymous on the called device.
For more details on Configuring External Presentation Information, see Configure Directory Number chapter
in the System Configuration Guide for Cisco Unified Communications Manager.
FMC Over SIP Trunks Without Smart Client

Unified Communications Manager allows service providers to provide base PBX-extension features such as
Enterprise Dialing, SNR, Single VM, Call move, and Mid-call features through the trunk without a smart
client on the mobile. Basic mobile features such as SNR, Deskphone Pickup, Send Call to Mobile, Mobile
Voice Access, and Mid-call DTMF features are supported. Extension dialing is supported if it is implemented
in the network and the network is integrated with Unified Communications Manager. These features can be
provided by any type of trunk.
Unified Communications Manager can be configured in the Ring All Shared Lines service parameter so that
the shared-line is rung when mobile DN is dialed.
35
Hunt Group Login and Logout for Carrier-Integrated Mobile Devices
Note The Reroute Remote Destination Calls to Enterprise Number feature must be enabled for Ring All Shared
Lines to take effect. Reroute Remote Destination Calls to Enterprise Number is disabled by default.
IMS shared lines will ring solely based on the value of the Ring All Shared Lines parameter.
You can also migrate from the Remote Destination feature used in previous versions to this new device type.
Hunt Group Login and Logout for Carrier-Integrated Mobile

Devices
When configuring the device type Carrier-Integrated Mobile, set the Owner User ID value to the mobile user
identity. The mobile user identity does not appear on the configuration. Only users with mobility enabled will
appear in the Owner User ID drop-down on the end user page and one line (DN) can be associated with an
FMC device. Users should associate a mobile identity with the FMC. This can be done on the FMC device
configuration page after the device has been added. For calls to be extended to the number of the mobile
identity, users must enable Cisco Unified Mobility on the Mobile Identity window.
Carrier-integrated mobile devices can be configured to support hunt group login and logout through Enterprise
Feature Access codes. Make sure that you've configured the following:
• Enterprise Feature Access must be configured in Call Routing > Mobility > Enterprise Feature Access
Number Configuration.
• Make sure you assign values for the Enterprise Feature Access Number for Hunt Group Login and
Enterprise Feature Access Number for Hunt Group Logout fields in Service Parameters.
After you configure these, the user can log in or log out of Hunt groups from Carrier-Integrated Mobile devices
by dialing the configured Enterprise Feature Access Number. If the user dials the given Hunt login access
code number, the Carrier-Integrated Mobile device allows them to be part of the Hunt group list. If the Hunt
logout access code is dialed, then the user is moved out of the Hunt group list and calls do not reach them.
Note Users on Carrier-Integrated Mobile devices can invoke midcall features via Enterprise Feature Access codes.
For details on how to configure and use Enterprise Feature Access, see Configure Enterprise Feature Access
section.
36
Cisco Unified Mobility Interactions
Cisco Unified Mobility Interactions

Table 2: Cisco Unified Mobility Interactions
Feature Interaction
Auto Call Pickup Cisco Unified Mobility interacts with auto call pickup depending on how you
configured the service parameter. When the Auto Call Pickup Enabled service
parameter is set to True, users must press only the PickUp softkey to pick up a
call.
If the service parameter is set to False, users must press the PickUp, GPickUp,
or OPickUp softkey and then the Answer softkey.
Automatic Alternate Cisco Unified Mobility supports automatic alternate routing (AAR) as follows:
Routing
• If a rejection occurs because of a lack of bandwidth for the location-based
service, the rejection triggers AAR and reroutes the call through the PSTN
so the caller does not need to hang up and redial.
• If a rejection occurs because of resource reservation protocol (RSVP),
however, AAR is not triggered for calls to remote destinations and the call
stops.
Extend and Connect Users who need the capabilities of both Cisco Unified Mobility and Extend and
Connect can configure the same remote destination on the remote device profile
and CTI remote device types when the owner ID of both device types is the same.
This configuration allows Cisco Unified Mobility features to be used concurrently
with Extend and Connect.
For more information, see the “Extend and Connect” chapter.
External Call Control If external call control is configured, Unified Communications Manager follows
the route decision from the adjunct route server for these Cisco Unified Mobility
features:
• Cisco Unified Mobility
• Mobile voice access
• Enterprise feature access
• Dial via office
Unified Communications Manager does not send a routing query for the following
Cisco Unified Mobility features:
• Cell pickup
• Desk pickup
• Session handoff
37
Cisco Unified Mobility Restrictions
Feature Interaction
Intelligent Session For direct calls to remote destinations that are anchored to the enterprise number,
Control and Session mobile users can use the session handoff feature to hand off the call to their
Handoff deskphones.
You must enable Cisco Unified Mobility before you implement intelligent session
control.
Licensing Cisco Unified Mobility is included in all user-based licenses from basic to
professional.
Local Route Groups For single number reach calls to a remote destination, the device pool of the
originating calling party determines the selection of the standard local route group.
Note Local Route Group is not supported when the AgentGreeting with
BiB (Built in Bridge) is invoked.
Number of Supported Each remote destination supports a maximum of six active calls. However, the
Calls number of supported calls depends on the Unified Communications Manager
configuration.
For example, the Cisco Unified Mobility user receives a call while the user already
has six calls for the remote destination or while the user is using DTMF to transfer
or conference a call from the remote destination.
The received call is sent to the enterprise voice mail when:
• The number of calls with user exceeds Busy trigger configuration
• CFB is configured
• All shared lines are busy
Note The calls sent to the enterprise voice mail is not based on the
maximum supported calls.
SIP Trunks with Cisco Cisco Unified Mobility supports the Cisco Unified Mobility feature without
Unified Border Element midcall features over SIP trunks with Cisco Unified Border Element (CUBE).

Table 3: Cisco Unified Mobility Interactions
Restriction Description
Auto Answer A remote destination call does not work when auto answer is enabled.
Note Auto Answer is not supported with Dual-Mode phones.
38
Call Forwarding Call Forward Unregistered (CFUR) support for Cisco Jabber on iPhone and
Unregistered Android is as follows:
• CFUR is supported if Cisco Jabber on iPhone or Android does not have
either a mobile identity or remote destination configured.
• CFUR is not supported, and will not work, if a Remote Destination is
configured
• CFUR is not supported, and will not work, if a Mobile Identity is configured
with a mobile phone number and Single Number Reach is enabled.
If you have a mobile identity or remote destination configured, use Call Forward
Busy and Call Forward No Answer instead.
Call Queuing Unified Communications Manager does not support call queuing with Cisco
Unified Mobility.
Conferencing Users cannot initiate a meet-me conference as conference controller by using

mobile voice access, but they can join a meet-me conference.
If an existing conference call is initiated from a shared-line IP phone or dual-mode
phone or smartphone that is a remote destination, no new conference party can
be added to the existing conference after the call is sent to a mobile phone or a
dual-mode handoff action occurs.
To permit the addition of new conference parties, use the Advanced Ad Hoc
Conference Enabled service parameter.
Dialing + Character from Users can dial a + sign through dual-tone multifrequency (DTMF) on a mobile
Mobile Phones phone to specify the international escape character.
Cisco Unified Mobility does not support + dialing through DTMF for IVR to
make an outgoing call from a mobile phone to an enterprise IP phone for which
the directory number contains the + character.
Cisco Unified Mobility does not support + dialing through DTMF for two-stage
dialing to make an outgoing call from a mobile phone to an enterprise IP phone
for which the directory number contains the + character.
Do Not Disturb on the If do not disturb (DND) is enabled on a desk phone, the desk phone cannot be
Desk Phone and Direct placed in the remote In use state and the call is not anchored in the following
Calls to Remote scenarios:
Destination
• DND is enabled with the call reject option.
• DND is activated by pressing the DND softkey on the desk phone.
If DND is enabled with the ring off option, however, the call is anchored.
39
Dual-Mode Phones Dual-Mode Handoff and Caller ID

The handoff DN method of dual-mode handoff requires a caller ID in the
cellular network. The mobility softkey method does not require caller ID.
Dual-Mode Phones and CTI Applications
While a dual-mode phone is in Wi-Fi enterprise mode, no CTI applications
control it nor monitor it.
The In Use Remote indicator for dual-mode phones on a shared line call in
the WLAN disappears if the dual-mode phone goes out of WLAN range.
Dual-Mode Phones and SIP Registration Period
For dual-mode phones, Unified Communications Manager determines the
registration period by using the value in the Timer Register Expires
(seconds) field of the SIP profile that associates with the phone, not the value
that the SIP Station KeepAlive Interval service parameter specifies. The
standard SIP profile for mobile devices determines the registration period as
defined by the Time Register Expires field in that profile.
Enterprise Features From Enterprise features from cellular networks require out-of-band DTMF.
Cellular Networks
When using intercluster DNs as remote destinations for an IP phone over a SIP
trunk (either intercluster trunk or gateway), check the Require DTMF Reception
check box when configuring the IP phone. This allows DTMF digits to be received
out of band, which is crucial for enterprise feature access midcall features.
Gateways and Ports Both H.323 and SIP VoIP gateways are supported for mobile voice access.
Cisco Unified Mobility features are not supported for T1 CAS, FXO, FXS and
BRI.
SNR(Single Number Reach) is not supported with MGCP(Media Gateway
Controlled Protocol).
Jabber Devices When initially configured, Jabber devices count as registered devices. These
devices increase the count of registered devices in a node, set by the Maximum
Number of Registered Devices service parameter.
Locales Cisco Unified Mobility supports a maximum of nine locales. If more than nine
locales are installed, they appear in the Available Locales pane, but you can only
save up to nine locales in the Selected Locales pane.
If you attempt to configure more than nine locales for Cisco Unified Mobility,
the following message appears: “Update failed. Check constraint
(informix.cc_ivruserlocale_orderindex) failed.”
40
Maximum Wait Timer for If a user presses the *81 DTMF code from a remote destination (either a
Desktop Call Pickup smartphone or any other phone) to put a call on hold, the user desk phone displays
the Resume softkey. However, the desk phone does not apply a timer for Desktop
Call Pickup. The Resume key continues to be displayed even after the timeout
that is configured for the end user to pick up the call elapses and the call is not
dropped.
Instead, users should hang up the call on the remote phone, which triggers the
desk phone to apply the timer for desktop call pickup. (Use the Maximum Wait
Time for Desk Pickup field on the End User Configuration window to change
this setting.)
Multilevel Precedence and Cisco Unified Mobility does not work with multilevel precedence and preemption
Preemption (MLPP). If a call is preempted with MLPP, Cisco Unified Mobility features are
disabled for that call.
Overlap Sending Overlap sending patterns are not supported for the Intelligent Session Control
feature.
Q Signaling Mobility does not support Q signaling (QSIG).
QSIG Path Replacement QSIG path replacement is not supported.
Service Parameters Enterprise feature access service parameters apply to standard phones and
smartphones; however, smartphones generally use one-touch keys to send the
appropriate codes. You must configure any smartphones that will be used with
Cisco Unified Mobility to use either the default codes for enterprise feature access
or the codes that are specified in the smartphone documentation.
Session Handoff The following limitations apply to the session handoff feature:
• Session handoff can take place only from mobile phone to desk phone. For
session handoff from desk phone to mobile phone, the current remote
destination pickup method specifies that you must use send call to mobile
phone.
• Only audio call session handoff is supported.
Single Number Reach If you have a hunt group configured and one or more of the directory numbers
with Hunt Groups that the hunt group points toward also has Single Number Reach (SNR) enabled,
the call does not extend to the SNR remote destinations unless all devices in the
hunt group are logged in.
For each device within the hunt group, the Logged Into Hunt Group check box
must be checked within the Phone Configuration window for that device.
SIP Trunks The Cisco Unified Mobility feature is supported only for primary rate interface
(PRI) public switched telephone network (PSTN) connections.
For SIP trunks, Cisco Unified Mobility is supported over IOS gateways or
intercluster trunks.
41
Cisco Unified Mobility Troubleshooting
SIP URI and Direct Calls The Intelligent session control feature does not support direct URI dialing.
to Remote Destination Therefore, calls that are made to a SIP URI cannot be anchored to an enterprise
number.
Unified Communications In a cluster environment, the publisher must be reachable in order to enable or
Manager publisher disable Single Number Reach. Some features may not function if the publisher
dependent features is not actively running.
Mobile voice access is not available when the publisher node is not reachable;
IVR prompts for Mobile Voice Access are stored only on the publisher.
Video Calls Cisco Unified Mobility services do not extend to video calls. A video call that is
received at the desk phone cannot be picked up on the mobile phone.
Mobile Voice Access Cisco 4000 Series Integrated Services Routers do not support Voice XML
(MVA) (VXML). Hence, when these routers function as Unified Communications
gateways with Cisco Unified Communications Manager, they do not support
Mobile Voice Access (MVA) application.
Related Topics
Ad Hoc Conferencing Service Parameters, on page 222
Cisco Unified Mobility Troubleshooting

Cannot Resume Call on Desktop Phone
Problem When a remote destination (mobile phone) is not a smartphone and a call to this mobile phone is
anchored through Cisco Unified Communications Manager, the user can hang up the mobile phone and expect
to see a Resume softkey on the user desktop phone to resume the call. The user cannot resume this call on
the user desktop phone.
Possible Cause If the calling party receives a busy, reorder, or disconnect tone when the mobile phone
hangs up, the mobile phone provider probably did not disconnect the media. No disconnect signals came
from the provider. To verify this possibility, let the calling party wait for 45 seconds. After this wait, the
service provider will time out and send disconnect signals, at which time Cisco Unified Communications
Manager can provide a Resume softkey to resume the call.
• Add the following command to the gateway:
voice call disc-pi-off
• For the Cisco CallManager service, set the Retain Media on Disconnect with PI for Active Call service
parameter to False.
42
CHAPTER 4
Device Mobility
• Device Mobility Overview, on page 43
• Device Mobility Prerequisites, on page 47
• Device Mobility Configuration Task Flow, on page 48
• Device Mobility Interactions, on page 52
• Device Mobility Restrictions, on page 53
Device Mobility Overview

Device mobility lets mobile users roam between sites, taking on the site-specific settings of the local site.
When this feature is configured, Cisco Unified Communications Manager matches the IP address of a roaming
device to IP subnets in the Device Mobility configuration to determine the physical location of the device so
that an appropriate device pool can be assigned. The settings from this dynamically-assigned device pool
override the settings in the Phone Configuration for that device and ensure that voice quality and allocation
of resources are appropriate for the new phone location.
For roaming mobile devices, this feature provides a more efficient use of network resources:
• When a mobile user moves to another location, call admission control (CAC) can ensure video and audio
quality with the appropriate bandwidth allocations for that location.
• When a mobile user makes a PSTN call, the phone is routed to the local gateway. Otherwise, PSTN calls
would first be routed back to the home site over IP WAN connections, and then on to a PSTN gateway
at the home site.
• When a mobile user calls the home location, Cisco Unified Communications Manager can assign the
appropriate codec for the region.
Site-Specific Settings
For roaming devices, Cisco Unified Communications Manager overwrites the following device pool parameters
from the device configuration with values from the dynamically assigned device pool:
• Date/Time Group
• Region
• Location
• Network Locale
43
Device Mobility Overview
• SRST Reference
• Connection Monitor Duration
• Physical Location
• Device Mobility Group
• Media Resource Group List
When networks span geographic locations outside the United States, you can configure device mobility groups
to allow phone users to use their configured dial plan no matter where they roam. When a device is roaming
but remains in the same device mobility group, Cisco Unified Communications Manager also overwrites the
following device pool parameters:
• AAR Group
• AAR Calling Search Space
• Device Calling Search Space
When the phone returns to its home location, the system disassociates the roaming device pool, downloads
the configuration settings for home location, and resets the device. The device registers with the home location
configuration settings.
Note Cisco Unified Communications Manager always uses the Communications Manager Group setting from the
phone record. The device always registers to its home location Cisco Unified Communications Manager server
even when roaming. When a phone is roaming, only network location settings such as bandwidth allocation,
media resource allocation, region configuration, and AAR group get changed.
Configuration
This feature needs to be enabled at both the system-level, and at the device level. At the system level, this
feature uses the following components:
• Physical Location—The physical location of the device pool. During registration, the system matches
the device registration location to a subnet in the Device Mobility Info in order to assign an appropriate
device pool.
• Device Pool—Location-specific device settings such as media resources, regions, and SRST references.
For roaming devices, the system assigns the device pool that matches that device’s physical location.
• Device Mobility Group—A logical group of sites with similar dialing patterns. For example, an enterprise
with a worldwide network might set up groups that represent individual countries. The device mobility
group setting determines whether the device is moved within the same geographical entity, primarily to
allow users to keep their own dial plans.
• Device Mobility Info—This info contains the subnets that the system provides for roaming devices, and
the device pools that the system can assign to roaming devices that register to one of those subnets.
At the device level, the feature must be turned on for devices to use this feature.
44
Device Pool Assignment
Figure 1: Device Mobility-Related Configurations
Device Pool Assignment

This section describes how Unified Communications Manager assigns device pools when device mobility is
enabled. Depending on whether the device is roaming, the device may be assigned a device pool in the local
site, or it may use the device pool from its home site.
Following initialization, the device mobility feature operates according to the following process:
1. A phone device record gets created for an IP phone that is provisioned to be mobile, and the phone gets
assigned to a device pool. The phone registers with Unified Communications Manager, and an IP address
gets assigned as part of the registration process.
2. Unified Communications Manager compares the IP address of the device to the subnets that are configured
for device mobility in the Device Mobility Info Configuration window. The best match uses the largest
number of bits in the IP subnet mask (longest match rule). For example, the IP address 9.9.8.2 matches
the subnet 9.9.8.0/24 rather than the subnet 9.9.0.0/16.
3. If the device pool in the phone record matches the device pool in the matching subnet, the system considers
the phone to be in its home location, and the phone retains the parameters of its home device pool.
4. If the device pool in the phone record does not match the device pools in the matching subnet, the system
considers the phone to be roaming. The following table describes possible scenarios for device mobility
and the system responses.
45
Device Mobility Groups Operations Summary
Table 4: Device Mobility Scenarios
Scenario System Response
The physical location setting in the phone device The system does not consider the phone to be
pool matches the physical location setting in a roaming, and the system uses the settings in the
device pool that is associated with the matching home location device pool.
subnet.
Note Although the phone may have moved
from one subnet to another, the
physical location and associated
services have not changed.
The matching subnet has a single device pool that The system considers the phone to be roaming. It
is assigned to it; the subnet device pool differs from reregisters with the parameters of the device pool
the home location device pool, and the physical for the matching subnet.
locations differ.
The physical locations differ, and the matching The system considers the phone to be roaming. The
subnet has multiple device pools assigned to it. new device pool gets assigned according to a
round-robin rule. Each time that a roaming device
comes in to be registered for the subnet, the next
device pool in the set of available device pools gets
assigned.
Physical location gets defined for the home device The physical location has not changed, so the phone
pool but is not defined for the device pools that are remains registered in the home device pool.
associated with the matching subnet.
Physical location that is not defined for the home The system considers the phone to be roaming to
device pool gets defined for the device pools that the defined physical location, and it registers with
are associated with the matching subnet. the parameters of the device pool for the matching
subnet.
A subnet gets updated or removed. The rules for roaming and assigning device pools
get applied by using the remaining subnets.
Note If no device mobility information entries match the device IP address, the device uses the home location device
pool settings.
Device Mobility Groups Operations Summary

You can use device mobility groups to determine when a device moves to another location within a geographic
entity, so a user can use its own dial plan. For example, you can configure a device mobility group for the
United States and another group for the United Kingdom. If a phone moves into a different mobility group
(such as from the United States to the United Kingdom), Unified Communications Manager uses the Calling
Search Space, AAR Group and AAR CSS from the phone record, and not from the roaming location.
46
Device Mobility Prerequisites
If the device moves to another location with same mobility group (for example, Richardson, USA, to Boulder,
USA), the CSS information gets taken from the roaming device pool settings. With this approach, if the user
is dialing PSTN destinations, the user reaches the local gateway.
The following table describes the device pool parameters that the system uses for various scenarios.
Table 5: Device Mobility Group Scenarios
Scenario Parameters Used
A roaming device moves Roaming Device Pool: yes

to another location in the
Location: Roaming device pool setting
same device mobility
group. Region: Roaming device pool setting
Media Resources Group List: Roaming device pool setting
Device CSS: Roaming device pool setting (Device Mobility CSS)
AAR Group: Roaming device pool setting
AAR CSS: Roaming device pool setting
A roaming device moves Roaming Device Pool: yes

to another location in a
Location: Roaming device pool setting
different device mobility
group. Region: Roaming device pool setting
Media Resources Group List: Roaming device pool setting
Device CSS: Home location settings
AAR Group: Home location settings
AAR CSS: Home location settings
A device roams, and a Because the device is roaming, it takes the roaming device pool settings, including
device mobility group the Device Mobility Calling Search Space, AAR Calling Search Space, and AAR
does not get defined for Group.
the home or roaming
device pool.
Device Mobility Prerequisites

• The phone must have a dynamic IP address to use device mobility. If a phone with a static IP address
roams, Unified Communications Manager uses the configuration settings from its home location.
• The Device Mobility feature requires you to set up device pools with site-specific settings. This chapter
describes only the device pool settings that relate to device mobility. For more detailed information on
configuring device pools, see the "Configure Device Pools" chapter in the System Configuration Guide
for Cisco Unified Communications Manager.
• Cisco Database Layer Monitor service must be running on the same node as the Cisco CallManager
service.
• Cisco TFTP service must be running on at least one node in the cluster.
47
Device Mobility Configuration Task Flow
• Cisco Unified Communications Manager Locale Installer (if you want to use non-English phone locales
or country-specific tones).
• Any phone that runs either SCCP or SIP.
Device Mobility Configuration Task Flow

Complete these tasks to configure device mobility.
Procedure

Step 1 Enable device mobility at the device level by Enables device support through a clusterwide
completing either of these tasks: service parameter, or within the Phone
Configuration window of an individual device.
• Enable Device Mobility Clusterwide, on
page 48
• Enable Device Mobility for Individual
Devices, on page 49
Step 2 Configure a Physical Location, on page 49 Set up the physical locations that you will assign
to your device pools.
Step 3 Configure a Device Mobility Group, on page A device mobility group is a logical grouping
50 of sites with similar dialing patterns.
Step 4 Configure a Device Pool for Device Mobility, Assign the physical location, device mobility
on page 50 group, and other device mobility-related
information to device pools that will be used
for device mobility.
Step 5 Configure Device Mobility Information, on Assign the IP subnets where roaming devices
page 51 can register and the device pools that can be
assigned to those roaming devices.
Enable Device Mobility Clusterwide

Use the following procedure to configure a service parameter that sets the default device mobility setting to
On for all phones clusterwide except where there is an overriding configuration in that phone's Phone
Configuration.
Procedure
Step 2 From the Server drop-down list, choose the node that is running the Cisco CallManager service.
Step 3 From the Service drop-down list, choose Cisco CallManager Service.
Step 4 Under Clusterwide Parameters (Device - Phone), set the Device Mobility Mode service parameter to On.
48
Enable Device Mobility for Individual Devices
Step 5 Click Save.

For devices that are already registered, you must restart the Cisco CallManager service for this new setting
to be enabled.
What to do next
If you want to configure device mobility settings for an individual device, go to Enable Device Mobility for
Individual Devices, on page 49.
Otherwise, you can begin configuring the system for device mobility. Go to Configure a Physical Location,
on page 49.
Enable Device Mobility for Individual Devices

Use this procedure to enable device mobility for an individual device. This configuration overrides the setting
of the Device Mobility Mode clusterwide service parameter.
Procedure

Step 2 Click Find and select the device that you want to configure.
Step 3 From the Device Mobility Mode drop-down list, choose one of the following:
• On—Device mobility is enabled for this device.
• Off—Device mobility is disabled for this device.
• Default—The device uses the setting of the Device Mobility Mode clusterwide service parameter. This
is the default setting.
Step 4 Click Save.
Configure a Physical Location

Use this procedure to configure a physical location that you will assign to a device pool. Device Mobility uses
the location of the device registration to assign an appropriate device pool.
Procedure
Step 1 From Cisco Unified CM Administration, choose System > Physical Location.
Step 3 Enter a Name for the location.
Step 4 Enter a Description for the location.
Step 5 Click Save.
49
Configure a Device Mobility Group
Configure a Device Mobility Group

Use the following procedure to configure device mobility group is a logical grouping of sites with similar
dialing patterns. For example, a company with a worldwide network may want to set up device mobility
groups that represent individual countries.
Procedure
Step 1 From Cisco Unified CM Administration, choose System > Device Mobility > Device Mobility Group.
Step 3 Enter a Name for the device mobility group.
Step 4 Enter a Description for the device mobility group.
Step 5 Click Save.
Configure a Device Pool for Device Mobility

Use this procedure to set up a device pool with parameters that you configured for device mobility.
Procedure
Step 1 From Cisco Unified CM Administration, choose System > Device Pool.
Step 2 Do either of the following:
• Click Find and select an existing device pool.
• Click Add New to create a new device pool.
Step 3 Under Roaming Sensitive Settings, assign the parameters that you set up in the previous device mobility
tasks:
• Physical Location—From the drop-down list, select the physical location that you set up for this device
pool. Device mobility uses this location to assign a device pool for a roaming device.
• Device Mobility Group—From the drop-down list, select the device mobility group that you set up for
this device pool.
Step 4 Under Device Mobility Related Information, configure the following device mobility-related fields. For
• Device Mobility Calling Search Space—Select the CSS to be used by a roaming device that uses this
device pool.
• AAR Calling Search Space—Select the calling search space for the device to use when automated
alternate routing (AAR) is performed.
• AAR Group—If AAR is configured, select the AAR Group for this device.
• Calling Party Transformation CSS—Select the calling party transformation CSS for roaming devices
that use this device pool.
50
Configure Device Mobility Information
Note • The Calling Party Transformation CSS overrides the device level configuration for roaming
devices, even if the Use Device Pool Calling Party Transformation CSS check box is
unchecked in the Phone Configuration window.
• The Called Party Transformation CSS setting is applied to the gateway rather than to the
roaming device.
Step 5 Configure any remaining fields in the Device Pool Configuration window. For more information on the
fields and their configuration options, see the system Online Help.
Step 6 Click Save.
Configure Device Mobility Information

Use this procedure to configure Device Mobility Info, representing the IP subnets to which roaming devices
can register and the corresponding device pools that the system can assign to roaming devices.
Procedure
Step 1 From Cisco Unified CM Administration, choose System > Device Mobility > Device Mobility Info.
Step 3 Enter a Name for the Device Mobibility Info.
Step 4 Enter the IP subnet details for roaming device registrations.
• If you are using IPv4 addresses for your mobile devices, complete the IPv4 subnet details.
• If you are using IPv6 addresses for your mobile devices, complete the IPv6 subnet details.
Step 5 Select the device pools that you want the system to assign for roaming devices that register to one of these
subnets. Use the arrows to move the appropriate device pools from the Selected Device Pools list box to the
Available Device Poolss list box.
Step 6 Click Save.
For more information on the fields and their configuration options, see Online Help.
View Roaming Device Pool Parameters

Use the following procedure to view and verify the current device mobility settings for a device.
Procedure

Step 2 Enter search criteria and click Find to find the device with device mobility mode enabled.
Step 3 Click View Current Device Mobility Settings next to the Device Mobility Mode field.
51
Device Mobility Interactions
The roaming device pool settings appear. If the device is not roaming, the home location settings appear.
Device Mobility Interactions

Table 6: Device Mobility Interactions
Feature Interaction
Calling Party Calling party normalization enhances the dialing capabilities of some phones and
Normalization improves call-back functionality when a call is routed to multiple geographical
locations; that is, the feature ensures that the called party can return a call without
the need to modify the directory number in the call log directories on the phone.
Additionally, calling party normalization allows you to globalize and localize
phone numbers, so the appropriate calling number presentation is displayed on
the phone.
Roaming When a device is roaming in the same device mobility group, Unified
Communications Manager uses the device mobility CSS to reach the local
gateway. If a user sets call forward all (CFA) at the phone, the CFA CSS is set
to None, and the CFA CSS activation policy is set to With Activating Device/Line
CSS, then the following behaviors will occur, depending on the device location:
• The Device CSS and Line CSS are used as the CFA CSS when the device
is in its home location.
• If the device is roaming within the same device mobility group, the device
mobility CSS from the roaming device pool and the line CSS are used as
the CFA CSS.
• If the device is roaming within a different device mobility group, the Device
CSS and Line CSS are used as the CFA CSS.
52
Device Mobility Restrictions

Table 7: Device Mobility Restrictions
IP Address The device mobility feature depends on the IPv4 address or IPv6 address of the
device that registers with Unified Communications Manager.
• The phone must have a dynamic IPv4 address or IPv6 address to use the
device mobility.
• If the device is assigned an IP address by using network address translation
(NAT) or port address translation (PAT), the IP address that is provided
during registration may not match the actual IP address of the device.
• If the Cisco IP phone supports IPv4-Only Stack or IPv6-Only Stack, then
the phone gets re-associated either with IPv4 or IPv6 Device Mobility
Information, based on the IP addressing mode preference defined. For
example, when a phone is defined with IPv6 preference but has no matching
Device Mobility Information (IPv6 subnet and mask size), then it is
associated with IPv4. When you add matching IPv6 Device Mobility
Information, then the phone gets re-associated with IPv6 Device Mobility
Information.
53
54
CHAPTER 5
Extend and Connect
• Extend and Connect Overview, on page 55
• Extend and Connect Prerequisites, on page 56
• Extend and Connect Configuration Task Flow, on page 56
• CTI Remote Device (CTIRD) Call Flows, on page 61
• Extend and Connect Interactions, on page 62
• Extend and Connect Restrictions, on page 63
Extend and Connect Overview

The Extend and Connect feature allows administrators to deploy Unified Communications Manager (UC)
Computer Telephony Integration (CTI) applications that interoperate with any endpoint. With Extend and
Connect, users can access UC applications from any location using any device.
The Extend and Connect feature for Unified Communications Manager provides the following UC features:
• Receive incoming enterprise calls
• Make Call
• Disconnect
• Hold and Retrieve
• Redirect and Forward
• Call Forward All
• Call Forward Busy
• Call Forward No Answer
• Do Not Disturb
• Play Dual Tone Multi Frequency (DTMF) (out-of-band and in-band)
• Consult Transfer, Conference
• Add, edit, and delete remote destinations
• Set remote destination as Active or Inactive
55
Extend and Connect Prerequisites
• Persistent Connection
• Play Whisper Announcement
Extend and Connect Prerequisites

• Cisco Jabber, Release 9.1(1) or later
• Cisco Unified Workspace License (CUWL) Standard, CUWL Professional, or Cisco User Connect
License (UCL) - Enhanced
Extend and Connect Configuration Task Flow

This section describes the procedures that you must complete to provision Unified Communications Manager
users with Extend and Connect capabilities. For information about provisioning Cisco Jabber for Windows
users with Extend and Connect, see the Cisco Jabber for Windows Installation and Configuration Guide.
Before you begin
Procedure

Step 1 Configure User Account, on page 56 Enable mobility for users so that they can use
CTI remote devices. CTI devices are off-cluster
phones that work with Cisco UC applications.
Step 2 Add User Permissions, on page 57 Add access control group permissions.
Step 3 Create CTI Remote Devices, on page 58 Configure off-cluster phones that users can use
with Cisco UC applications.
Step 4 Add Directory Number to a Device, on page Associate a directory number with the CTI
58 remote device.
Step 5 Add Remote Destination, on page 59 Add a numerical address or directory URI that
represents the other phones that the user owns.
Step 6 Verify Remote Destination, on page 60 Verify if the remote destination is successfully
added for a user.
Step 7 Associate User with Device, on page 60 Associate an end user account to the CTI remote
device.
Configure User Account

Use the following procedure to configure a new or existing user in Unified Communications Manager, you
must enable user mobility so that they can use CTI remote devices. If you do not enable mobility for users,
you cannot assign those users as owners of CTI remote devices.
56
Add User Permissions
Procedure
Step 2 Perform either of the following:
• Click Add New, to configure a new user.
• Apply the filters using the Find User Where field and click Find to retrieve a list of users.
Note You may add the new end user account through LDAP integration or local configuration.
Step 3 Locate the Mobility Information section.

Step 4 Check the Enable Mobility check box.
Step 5 Click Save.
Add User Permissions

After the end user is active in Unified Communications Manager, add access control group permissions.
Procedure
Step 2 Specify the appropriate filters in the Find User Where field, and then select Find to retrieve a list of users.
Step 3 Select the user from the list.
Step 4 Locate the Permissions Information section.
Step 5 Click Add to Access Control Group.
The Find and List Access Control Groups window appears.
Step 6 Click Find.

The Access Control Group list for Standard Users appears.
Step 7 Check the check boxes next to the following permissions:

• Standard CCM End-Users
• Standard CTI Enabled
Step 8 Click Add Selected.

Step 9 Click Save.
57
Create CTI Remote Devices
Create CTI Remote Devices

Use the following procedure to create a CTI remote device is a device type that represents off-cluster phones
that users can use with Cisco UC applications. The device type is configured with one or more lines (directory
numbers) and one or more remote destinations.
Unified Communications Manager provides Extend and Connect capabilities to control calls on devices such
as public switched telephone network (PSTN) phones and private branch exchange (PBX) devices.
Procedure

Step 3 Select CTI Remote Device from the Phone Type drop-down list and then click Next.
Step 4 Select the appropriate user ID from the Owner User ID drop-down list.
Note Only users for whom you enable mobility are available from the Owner User ID drop-down list.
Unified Communications Manager populates the Device Name field with the user ID and a CTRID prefix,
for example, CTRIDusername.
Step 5 Edit the default value in the Device Name field, if appropriate.
Step 6 Enter a meaningful description in the Description field.
Note Cisco Jabber displays device descriptions to users. If Cisco Jabber users have multiple devices
of the same model, the descriptions from Unified Communications Manager help users tell the
difference between them.
Step 7 Ensure that you select an appropriate option from the Rerouting Calling Search Space drop-down list in the
Protocol Specific Information section.
The Rerouting Calling Search Space drop-down list defines the calling search space for rerouting and ensures
that users can send and receive calls from the CTI remote device.
Step 8 Configure the remaining fields in the Phone Configuration window. For more information on the fields and
Step 9 Click Save.
The fields to associate directory numbers and add remote destinations are displayed in the Phone Configuration
window.
Add Directory Number to a Device

A directory number (DN) is a numerical address that is configured as a line on the CTI remote device. A DN
typically represents the primary work number of a user (for example, 2000 or +1 408 200 2000).
58
Add Remote Destination
Note • The Calling Search Space (CSS) and partition of DN are mandatory on devices.
• The CTI Remote Device should not block its own DN. The CSS is important for the CTIRD device to
reach its own DN.
Follow these steps to add a directory number to a CTI remote device.
Procedure
Step 1 Locate the Association Information section in the Phone Configuration window.
Step 2 Click Add a new DN.
Step 3 Specify a directory number in the Directory Number field.
Step 4 Configure all other required fields. For more information on the fields and their configuration options, see
Online Help.
Step 5 Click Save.
Add Remote Destination

Use the following procedure to add a remote destination is a numerical address or directory URI that represents
the other phones that the user owns (for example, a home office line or other PBX phone). A remote destination
may be any off-cluster device. Unified Communications Manager automatically applies application dial rules
to all remote destination numbers for CTI remote devices. By default, four remote destinations are supported
per device. You can set the maximum number to 10 remote destinations per device in End User Configuration
window.
Note You can determine which remote destination the Jabber client has set as Active by opening the Phone
Configuration window from the Cisco Unified Communications Manager Administration interface.
Note Unified Communications Manager users can add remote destinations through the Cisco Jabber interface. For
more information, see the Cisco Jabber for Windows Installation and Configuration Guide.
• Unified Communications Manager automatically verifies whether it can route calls to remote destinations
that Cisco Jabber users add through the client interface.
• Unified Communications Manager does not verify whether it can route calls to remote destinations that
you add through the Cisco Unified Communications Manager Administration interface.
59
Verify Remote Destination
Procedure

Step 2 Specify the appropriate filters in the Find Phone Where field to and then click Find to retrieve a list of
phones.
Step 3 Select the CTI remote device from the list.
Step 4 Locate the Associated Remote Destinations section.
Step 5 Click Add a New Remote Destination.
Step 6 Enter the destination number in the Destination Number field.
To use the remote destination with Cisco Jabber clients, you must configure the destination name as JabberRD.
Step 7 Configure the remaining fields in the Remote Destination Information window. For more information on
the fields and their configuration options, see Online Help.
Step 8 Click Save.
Verify Remote Destination

Perform these steps to verify if the remote destination is successfully added for a user.
Procedure

Step 2 Specify the appropriate filters in the Find Phone Where field to and then click Find to retrieve a list of
phones.
Step 3 Select the CTI remote device from the list.
Step 4 Locate the Associated Remote Destinations section and verify that the remote destination is available.
Step 5 Click Apply Config.
Note The Device Information section on the Phone Configuration window indicates when a remote
destination is active or controlled by Cisco Jabber.
Associate User with Device

Procedure
Step 2 Specify the appropriate filters in the Find User Where field to and then click Find to retrieve a list of users.
Step 4 Locate the Device Information section.
60
CTI Remote Device (CTIRD) Call Flows
Step 5 Click Device Association.

Step 6 Find and select the CTI remote device.
Step 7 To complete the association, click Save Selected/Changes.
Step 8 From Related Links drop-down list, choose Back to User, and then click Go.
The End User Configuration window appears, and the associated device that you chose appears in the
Controlled Devices pane.
CTI Remote Device (CTIRD) Call Flows

Unified Communications Manager supports the separate calling party number and billing number feature
when users are added as CTI Remote Device. Each CTI Remote Device is configured with the user directory
number (DN) (for example, 2000) and a remote destination that represents any off-cluster device (for example,
a PBX phone with the number +1 408 111 1111).
When a call is initiated from the PSTN network to a CTIRD line, Unified Communications Manager looks
for FROM and PAID header information. The FROM header contains the external presentation name and
number and PAID contains the identity of the user (that is a user’s DN or DDI).
If FROM and PAID headers have different numbers and Enable External Presentation Name and Number
check box is checked in its SIP profile and Display External Presentation Name and Number service
parameter value set to True, then Unified Communications Manager displays the FROM header information
on the called device. In the same way, if any one option is disabled,Unified Communications Manager displays
PAID header information on the called device.
Similarly, in the outgoing call scenario a user calls from Remote Destination (CTIRD line) configured with
External Presentation Name and External Presentation Number on Directory Number configuration page
to a PSTN network through a SIP trunk with Enable External Presentation Name and Number configured
in its SIP profile. Then, Unified Communications Manager send the External Presentation Information
configured on the Directory Number Configuration page in the FROM header of the outgoing SIP message
to display on the called device.
If Enable External Presentation Name and Number check box is unchecked, then Unified Communications
Manager sends the directory number information in the FROM and PAID to display on the called device and
configured External Presentation Information in the X-Cisco-Presentation header.
If you check the Anonymous External Presentation check box, the configured External Presentation Name
and Number are removed from the respective fields and external presentation displayed as anonymous on the
called device.
For more details on Configuring External Presentation Information, see the Configure Directory Number
chapter in the System Configuration Guide for Cisco Unified Communications Manager.
61
Extend and Connect Interactions
Extend and Connect Interactions

Table 8: Extend and Connect Interactions
Feature Interaction
Directory URI Dialing Configure a Directory URI as the DN, remote destination, or both for the CTI
remote device.
Unified Mobility Extend and Support does not support moving active calls between a Cisco Unified
IP Phone and a remote destination.
If you want the capabilities of both Unified Mobility and Extend and Connect,
you can configure the same remote destination on the Remote Device Profile and
CTI Remote Device types when the Owner ID of both device types is the same.
This configuration allows Cisco Mobility features to be used concurrently with
Extend and Connect. The ability to configure the same remote destination on
both device types is supported using Cisco Unified Communications Manager
Release 10.0(1) or later.
Do not configure remote destinations that are used with the Cisco Extend and
Connect feature on Cisco Dual-mode for iPhone, Cisco Dual-mode for Android,
and Carrier-integrated Mobile device types. Do not use prefixes to differentiate
the same remote destination address. For example, 91-4085555555 and
+1-4085555555 are treated as the same number.
Hunt List The Extend and Connect feature allows users to receive hunt calls on remote
destination phones under the following conditions:
• The user has a Cisco Unified IP Phone.
• The Cisco Unified IP Phone is available to answer hunt calls
(logged-in/HLog).
• Cisco Jabber is running in Extend and Connect mode.
CallerID Information • The incoming caller ID information (name and number) is displayed on the
Jabber client.
• This information may also be displayed on the device, depending on your
carrier and trunk configuration.
• Outbound Dial Via Office calls to the remote destination display Voice
Connect as the name and the trunk DID as the number.
• Configure the trunk DID in the Unified CM Trunk Pattern, Route Pattern,
or Cisco Gateway. This configuration may also be assigned by the carrier.
The number field may display as blank if the trunk DID is not configured.
• Outbound calls to the desired party display the CTI Remote Device Display
Name and Directory Number (DN) as configured in Unified Communications
Manager.
• Remote destination numbers are never displayed to the called party.
62
Extend and Connect Restrictions

Table 9: Extend and Connect Restrictions
Maximum number of You can configure up to ten remote destinations for each CTI remote device.
remote destinations
Note By default, four remote destinations are supported per device. You
can set the maximum number to 10 remote destinations per device.
Off-cluster devices • Remote destination numbers must represent off-cluster devices.

• Remote destinations can be off-cluster URIs.
Directory numbers You cannot configure directory numbers as remote destination numbers.
Cisco Jabber Before you save the remote destinations that are configured using Cisco Jabber,
verify if the remote destinations can be routed by the configured dial plan.
Application dial rules Application Dial Rules are applied to all remote destinations that are configured
on the CTI remote device through the Cisco Unified Communications Manager
Administration interface and Cisco Jabber.
Note Advise end users which number formats the Application Dial Rules
are configured to support (for example, nn-nnn-nnnn, E.164, both).
Remote destination Each remote destination number must be unique within the cluster.
number
Note Two or more users cannot use the same remote destination numbers.
Remote destination • Remote destination numbers are validated using the CTI remote device
validation reroute calling search space.
• Remote destinations that are configured using the Cisco Unified
Communications Manager Administration interface and AXL interface are
not validated.
Consult transfer limitation When a consult transfer is initiated from a CTI Remote Device to an internal IP
phone or another extend and connect enabled device, no ringback is heard on the
remote destination associated to the CTI Remote device which is initiating the
transfer.
Call Forward Extend and Connect does not support Call Forward Unregistered Internal or Call
Unregistered Forward Unregistered External.
Route Next Hop By Extend and Connect does not support Translation Patterns when the "Route Next
Calling Party Number Hop By Calling Party Number" option is enabled.
63
64
CHAPTER 6
Remote Worker Emergency Calling
• Remote Worker Emergency Calling Overview, on page 65
• Remote Worker Emergency Calling Prerequisites, on page 65
• Remote Worker Emergency Calling Configuration Task Flow, on page 66
Remote Worker Emergency Calling Overview

The Remote Worker Emergency Calling feature enables customers to provide reliable emergency calling
support to remote workers by using remote Virtual Private Network (VPN) connections. Emergency calls
from off-premises users are routed to the Public Safety Answering Point (PSAP), and user-provided location
information is delivered with each call.
To use this feature, remote workers must confirm or update their location whenever their device registration
is interrupted. A customizable disclaimer notice is first displayed on the devices that are designated for
off-premises (connected remotely to the customer network), which advises the users to provide correct location
information. After the location information is provided, the off-premises location that is currently associated
with the designated device is displayed. Users can confirm their current location or select another previously
stored location from their device display; if their location is new, they are directed to the Cisco Emergency
Responder Off-Premises User web page to create a new location.
Before completing this process, the administrator may restrict the device to calling a single configured
destination. This action ensures that the device user has acknowledged the disclaimer and provided current
location information before the device is enabled for normal use.
Remote Worker Emergency Calling Prerequisites

You must configure Intrado (a third party application) on the Cisco Emergency Responder before you configure
the Remote Worker Emergency Calling feature. For information about configuring Intrado on the Cisco
Emergency Responder, see Cisco Emergency Responder Administration Guide
65
Remote Worker Emergency Calling Configuration Task Flow
Remote Worker Emergency Calling Configuration Task Flow

Before you begin
Procedure

Step 1 Configure User As a Remote Worker, on page Associate the off-premises device with the
66 owner of the device.
Step 2 Specify Alternate Routing for Emergency These parameters specify the calling search
Calling, on page 67 space and destination number that are used to
restrict the routing of any call that is made from
a registered off-premises device where the user
chose not to set a location. If these parameters
are not configured, calls are routed normally.
Step 3 Configure the Application Server, on page 67 Direct end users to the application server where
they enter the location of the device.
Step 4 Configure E911 Messages, on page 67 Configure the E911 messages that appear on an
off-premises end-user phone.
Configure User As a Remote Worker

Before you begin
Ensure that you have configured Intrado on the Cisco Emergency Responder. For more information about
configuring Intrado on the Cisco Emergency Responder, see Cisco Emergency Responder Administration
Guide.
Procedure

Step 2 Enter the appropriate search criteria to find the phone and click Find.
A list of phones that match the search criteria is displayed.
Step 3 Choose the phone for which you want to configure Remote Worker Emergency Calling.
The Phone Configuration window is displayed.
Step 4 From the Device Information section, select the appropriate user ID from the Owner User ID drop-down
list and check the Require off-premise location check box.
Step 5 Click Save.
66
Specify Alternate Routing for Emergency Calling
Specify Alternate Routing for Emergency Calling

Perform the following steps to configure calling search space and destination number. These parameters are
used to restrict the routing of any call made from a registered off-premises device where the user has not set
a location. If you do not configure these parameters, the calls are routed normally.
Procedure
Step 2 From the Server drop-down list, choose a server.
The Service Parameter Configuration window appeares.
Step 4 In the Clusterwide Parameters (Emergency Calling for Required Off-premise Location) section, specify
Alternate Destination for Emergency Call.
Step 5 Specify Alternate Calling Search Space for Emergency Call.
Step 6 Click Save.
Configure the Application Server

You must configure the application server to enable the E911 Proxy to communicate with the Cisco Emergency
Responder. E911 proxy is used to direct the users to the application server where they enter the location of
the device.
Procedure
Step 1 From Cisco Unified CM Administration, choose System > Application Server.
The Application Server window appears.
Step 3 From the Application Server Type drop-down list, select CER Location Management.
Step 4 Click Next.
Step 5 In the Name field, specify a name to identify the application server that you are configuring.
Step 6 In the IP address field, specify the IP address of the server that you are configuring.
Step 7 From the list of Available Application Users, select the application user and click the Down arrow.
Step 8 In the End User URL field, enter a URL for the end users that are associated with this application server.
Step 9 Click Save.
Configure E911 Messages

Use the following procedure to select and edit E911 messages for off-premises devices.
67
Configure E911 Messages
Procedure
Step 1 From Cisco Unified CM Administration, choose System > E911 Messages.
Step 2 Select the required language link of the E911 messages.
The E911 Messages Configuration page displays the Agreement, Disclaimer, and Error messages.
Step 3 (Optional) Edit the E911 messages to be displayed on off-premises devices.

Step 4 Click Save.
68
CHAPTER 7
Configure Mobile and Remote Access
• Mobile and Remote Access Overview, on page 69
• Mobile and Remote Access Prerequisites, on page 71
• Mobile and Remote Access Configuration Task Flow, on page 72
Mobile and Remote Access Overview

Unified Communications Manager Mobile and Remote Access is a core part of the Cisco Collaboration Edge
Architecture. It allows endpoints such as Cisco Jabber to have their registration, call control, provisioning,
messaging, and presence services that are provided by Unified Communications Manager when the endpoint
is not within the enterprise network. Cisco Expressway connects the mobile endpoint to the on-premises
network, providing secure firewall traversal and line-side support for Unified CM registrations.
The overall solution provides:
• Off-premises access: a consistent experience outside the network for Jabber and EX/MX/SX Series
clients
• Security: secure business-to-business communications
• Cloud services: enterprise grade flexibility and scalable solutions providing rich Webex integration and
Service Provider offerings
• Gateway and interoperability services: media and signaling normalization, and support for non-standard
endpoints
69
Mobile and Remote Access Overview
Figure 2: Unified Communications: Mobile and Remote Access
Third-party SIP or H.323 devices can register to the Expressway-C and, if necessary, interoperate with Unified
CM-registered devices over a SIP trunk.
Figure 3: Typical Call Flow: Signaling and Media Paths
• Unified CM provides call control for both mobile and on-premises endpoints.
• Signaling traverses the Expressway solution between the mobile endpoint and Unified CM.
• Media traverses the Expressway solution and is relayed between endpoints directly; all media is encrypted
between the Expressway-C and the mobile endpoint.
Configuring Mobile and Remote Access

To enable Cisco Jabber users with Mobile and Remote Access functionality, set up an Mobile and Remote
Access User Policy within the User Profile Configuration window of Unified Communications Manager.
The Mobile and Remote Access User Policy is not required for non-Jabber endpoints.
In addition, you must configure Cisco Expressway with Mobile and Remote Access. For details, see Mobile
and Remote Access via Cisco Expressway Deployment Guide .
70
Mobile and Remote Access Prerequisites
Mobile and Remote Access Prerequisites

Cisco Unified Communications Manager Requirements
The following requirements apply:
• If you are deploying multiple Unified Communications Manager clusters, set up an ILS network.
• Mobile and Remote Access requires that you set up NTP servers for your deployment. Make sure that
you have NTP servers deployed for your network and Phone NTP References for SIPendpoints.
• If you are deploying ICE for media path optimization, you will need to deploy a server that can provide
TURN and STUN services.
DNS Requirements
For the internal connection to Cisco Expressway, configure the following locally resolvable DNS SRV that
points to Unified Communications Manager:
_cisco-uds._tcp<domain>
You must create internal DNS records, for both forward and reverse lookups, for all Unified Communications
nodes used with Mobile and Remote Access. This allows Expressway-C to find the nodes when IP addresses
or hostnames are used instead of FQDNs. Make sure that the SRV record is not resolvable outside of the local
network.
Cisco Expressway Requirements

This feature requires you to integrate Unified Communications Manager with Cisco Expressway. For Cisco
Expressway configuration details for Mobile and Remote Access, refer to the Mobile and Remote Access
Through Cisco Expressway Deployment Guide.
The minimum Expressway release for Mobile and Remote Access Access Policy support with Cisco Jabber
is X8.10.
Certificate Prerequisites
You must exchange certificates between Unified Communications Manager, the IM and Presence Service,
and Cisco Expressway-C. Cisco recommends that you use CA-signed certificates with the same CA for each
system. In this case:
• Install the CA root certificate chain on each system (for Unified Communications Manager and the IM
and Presence Service Service install the certificate chain to the tomcat-trust store).
• For Unified Communications Manager, issue a CSR to request CA-signed tomcat (for AXL and UDS
traffic) and Cisco CallManager (for SIP) certificates.
• For the IM and Presence Service Service, issue a CSR to request CA-signed tomcat certificates.
Note If you use different CAs, you must install each CA's root certificate chain on Unified Communications Manager,
IM and Presence Service Service, and Expressway-C.
71
Mobile and Remote Access Configuration Task Flow
Note You can also use self-signed certificates for both Unified Communications Manager and the IM and Presence
Service Service. In this case, you must upload onto Expressway-C the tomcat and Cisco CallManager certificates
for Unified Communications Manager and a tomcat certificate for the IM and Presence Service Service.
Mobile and Remote Access Configuration Task Flow

Complete these tasks in Unified Communications Manager if you want to deploy Mobile and Remote Access
endpoints.
Procedure

Step 1 Activate Cisco AXL Web Service, on page 73 Make sure that the Cisco AXL Web Service is
activated on the publisher node.
Step 2 Configure Maximum Session BitRate for Video, Optional. Configure Region-specific settings
on page 73 for your Mobile and Remote Access endpoints.
For example, if you expect Mobile and Remote
Access endpoints to use video, you may want
to increase the Maximum Session Bit Rate for
Video Calls setting as the default setting of 384
kbps may be too low for some video endpoints.
Step 3 Configure a Device Pool for Mobile and Assign your Date/Time Group and Region
Remote Access, on page 74 configuration to the device pool that your
Mobile and Remote Access endpoints use.
Step 4 Configure ICE, on page 74 Optional. ICE is an optional deployment that

uses STUN and TURN services to analyze the
available media paths for an Mobile and Remote
Access call and then to select the best path. ICE
adds potentially to the call setup time, but
increases the reliability of Mobile and Remote
Access calls.
Step 5 Configure Phone Security Profile for Mobile Use this procedure to set up a phone security
and Remote Access, on page 75 profile to be used by Mobile and Remote Access
endpoints.
Step 6 Configure Mobile and Remote Access Access Cisco Jabber only. Set up an Mobile and
Policy for Cisco Jabber Users, on page 76 Remote Access Access Policy for Cisco Jabber
users. Cisco Jabber users must be enabled with
Mobile and Remote Access access within their
user profiles in order to use the Mobile and
Remote Access feature.
72
Activate Cisco AXL Web Service

Step 7 Configure Users for Mobile and Remote For Cisco Jabber users, the User Policy that you
Access, on page 77 set up must be applied to their End User
Configurations.
Step 8 Configure Endpoints for Mobile and Remote Configure and provision endpoints that use the
Access, on page 77 Mobile and Remote Access feature.
Step 9 Configure Cisco Expressway for Mobile and Configure Cisco Expressway for Mobile and
Remote Access, on page 77 Remote Access.
Activate Cisco AXL Web Service

Make sure that the Cisco AXL Web Service is activated on the publisher node.
Procedure
Step 2 From the Server drop-down list, select the publisher node and click Go.
Step 3 Under Database and Admin Services, confirm that the Cisco AXL Web Service is Activated.
Step 4 If the service is not activated, check the corresponding check box and click Save to activate the service.
Configure Maximum Session BitRate for Video

Configure Region settings for your Mobile and Remote Access endpoints. The default settings may be sufficient
in many cases, but if you expect Mobile and Remote Access endpoints to use video, you may want to increase
the Maximum Session Bit Rate for Video Calls within your Region Configuration. The default setting of
384 kbps may be too low for some video endpoints, such as the DX series.
Procedure
Step 1 From Cisco Unified CM Administration, choose System > Region Information > Region.
Step 2 Perform any one of the following:
• Click Find and select the region to edit the bit rates within an existing region.
• Click Add New to create a new region.
Step 3 In the Modify Relationship to other Regions area, configure a new setting for the Maximum Session Bit
Rate for Video Calls. For example, 6000 kbps.
Step 4 Configure any other fields in the Region Configuration window. For more information on the fields and
their configuration options, see Online Help..
Step 5 Click Save.
73
Configure a Device Pool for Mobile and Remote Access
Configure a Device Pool for Mobile and Remote Access

When you created a new region, assign your region to the device pool that your Mobile and Remote Access
endpoints use.
Procedure
• Click Find and select the existing device pool to edit.
• Click Add New to create a new device pool.
Step 3 Enter a Device Pool Name.

Step 4 Select a redundant Cisco Unified Communications Manager Group.
Step 5 Assign the Date/Time Group that you set up. This group includes the Phone NTP references that you set up
for Mobile and Remote Access endpoints.
Step 6 From the Region drop-down list, select the region that you configured for Mobile and Remote Access.
Step 7 Complete the remaining fields in the Device Pool Configuration window. For more information on the fields
Step 8 Click Save.
Configure ICE
Use this procedure if you want to deploy ICE to handle call setup for Mobile and Remote Access calls. ICE
is an optional deployment that uses STUN and TURN services to analyze the available media paths for an
Mobile and Remote Access call and to select the best path. ICE adds potentially to the call setup time, but
increases the reliability of Mobile and Remote Access calls.
Before you begin

Decide how you are going to deploy ICE. You can configure ICE for groups of phones via the Common Phone
Profile Configuration, to individual Cisco Jabber desktop devices, or through system-wide defaults that apply
to all phones.
As a fallback mechanism, ICE can use a TURN server to relay media. Make sure that you have deployed a
TURN server.
Procedure
Step 1 From Cisco Unified CM Administration:

• Choose System > Enterprise Phone to configure system defaults for ICE.
• Choose Device > Device Settings > Common Phone Profile to configure ICE for groups of endpoints
and select the profile you want to edit.
• Choose Device > Phone to configure ICE for an individual Cisco Jabber desktop endpoint and select
the endpoint that you want to edit.
74
Configure Phone Security Profile for Mobile and Remote Access
Step 2 Scroll down to the Interactive Connectivity Establishment (ICE) section.

Step 3 Set the ICE drop-down list to Enabled.
Step 4 Set the Default Candidate Type:
• Host—A candidate obtained by selecting the IP address on the host device. This is the default.
• Server Reflexive—An IP address and port candidate obtained by sending a STUN request. In many
cases, this may represent the public IP address of the NAT.
• Relayed—An IP address and port candidate obtained from a TURN server. The IP address and port are
resident on the TURN server such that media is relayed through the TURN server.
Step 5 From the Server Reflexive Address drop-down list, select whether you want to enable STUN-like services
by setting this field to Enabled or Disabled. You must set this field to enabled if you configured Server
Relexive as the Default Candidate.
Step 6 Enter the IP address or hostname for the Primary and Secondary TURN Servers.
Step 7 Set the TURN Server Transport Type to Auto (default setting), UDP, TCP, or TLS.
Step 8 Enter the Username and Password of the TURN Server.
Step 9 Click Save.
Note If you configured ICE for a Common Phone Profile, you must associate phones to that Common
Phone Profile for phones to be able to use the profile. You can apply the profile to a phone through
the Phone Configuration window.
Configure Phone Security Profile for Mobile and Remote Access

Use this procedure to set up a phone security profile to be used by Mobile and Remote Access endpoints.
Procedure
Step 1 From Cisco Unified CM Administration, choose System > Security > Phone Security Profile.
Step 3 From the Phone Security Profile Type drop-down list, select your device type. For example, you could select
Cisco Unified Client Service Framework for a Jabber application.
Step 4 Click Next.
Step 5 Enter a Name for the profile. For Mobile and Remote Access, the name must be in FQDN format and must
include the enterprise domain.
Step 6 From the Device Security Mode drop-down list, select Encrypted.
Note This field must be set to Encrypted. Otherwise, Expressway rejects communications.
Step 7 Set the Transport Type to TLS.

Step 8 Leave the TFTP Encrypted Config check box unchecked for the following phones as Mobile and Remote
Access will not work for these phones with this option enabled: DX Series, IP Phone 7800, or IP Phone 8811,
8841, 8845, 8861 and 8865
Step 9 Complete the remaining fields in the Phone Security Profile Configuration window. For more information
75
Configure Mobile and Remote Access Access Policy for Cisco Jabber Users
Step 10 Click Save.

Note You must apply this profile to the Phone Configuration for each of your Mobile and Remote
Access endpoints.
Configure Mobile and Remote Access Access Policy for Cisco Jabber Users
Use this procedure to set up an Mobile and Remote Access Access Policy for Cisco Jabber users. Cisco Jabber
users must be enabled with Mobile and Remote Access access within their user profiles in order to use the
Mobile and Remote Access feature. The minimum Expressway release for Mobile and Remote Access Policy
support with Cisco Jabber is X8.10.
Note The Mobile and Remote Access Policy is not required for non-Jabber users.
For more information on user profiles, see "User Profile Overview" section in System Configuration Guide
Procedure
Step 1 From Cisco Unified CM Administration, choose User Management > User Settings > User Profile.
Step 3 Enter a Name and Description for the user profile.
Step 4 Assign a Universal Device Template to apply to users' Desk Phones, Mobile and Desktop Devices, and
Remote Destination/Device Profiles.
Step 5 Assign a Universal Line Template to apply to the phone lines for users in this user profile.
Step 6 If you want the users in this user profile to be able to use the self-provisioning feature to provision their own
phones, do the following:
a) Check the Allow End User to Provision their own phones check box.
b) In the Limit Provisioning once End User has this many phones field, enter a maximum number of
phones the user is allowed to provision. The maximum is 20.
Step 7 If you want Cisco Jabber users who are associated with this user profile, to be able to use the Mobile and
Remote Access feature, check the Enable Mobile and Remote Access check box.
Note • By default, this check box is selected. When you uncheck this check box, the Client Policies
section is disabled, and No Service client policy option is selected by default.
• This setting is mandatory only for Cisco Jabber users whom are using OAuth Refresh Logins.
Non-Jabber users do not need this setting to be able to use Mobile and Remote Access.
Mobile and Remote Access feature is applicable only for the Jabber Mobile and Remote
Access users and not to any other endpoints or clients.
Step 8 Assign the Jabber policies for this user profile. From the Desktop Client Policy, and Mobile Client Policy
drop-down list, choose one of the following options:
76
Configure Users for Mobile and Remote Access
• No Service—This policy disables access to all Cisco Jabber services.

• IM & Presence only—This policy enables only instant messaging and presence capabilities.
• IM & Presence, Voice and Video calls—This policy enables instant messaging, presence, voicemail, and
conferencing capabilities for all users with audio or video devices. This is the default option.
Note Jabber desktop client includes Cisco Jabber for Windows users and Cisco Jabber for Mac users.
Jabber mobile client includes Cisco Jabber for iPad and iPhone users and Cisco Jabber for Android
users.
Step 9 If you want the users in this user profile to set the maximum login time for Extension Mobility or Extension
Mobility Cross Cluster through Cisco Unified Communications Self Care Portal, check the Allow End User
to set their Extension Mobility maximum login time check box.
Note By default Allow End User to set their Extension Mobility maximum login time check box
is unchecked.
Step 10 Click Save.
Configure Users for Mobile and Remote Access

For Cisco Jabber users, the Mobile and Remote Access access policy that you configured must be associated
to your Cisco Jabber users during the LDAP sync. For more information on how to provision end users, see
"End User Configuration" section in System Configuration Guide for Cisco Unified Communications Manager.
Configure Endpoints for Mobile and Remote Access

Provision and configure endpoints for Mobile and Remote Access:
• For Cisco Jabber clients, refer to "Cisco Jabber Configuration Task Flow" section in System Configuration
Guide for Cisco Unified Communications Manager.
• For other endpoints, refer to "Endpoint Device Configuration" section in System Configuration Guide
Configure Cisco Expressway for Mobile and Remote Access

For details on how to configure Cisco Expressway for Mobile and Remote Access, refer to the Mobile and
Remote Access Through Cisco Expressway Deployment Guide.
77
Configure Cisco Expressway for Mobile and Remote Access
78
PA R T III
Remote Network Access
• Wireless LAN, on page 81
• VPN Client , on page 85
CHAPTER 8
Wireless LAN
• Wireless LAN Overview, on page 81
• Wireless LAN Configuration Task Flow, on page 81
Wireless LAN Overview

This feature removes the need for users to configure WiFi parameters on their phones. You can configure
WiFi profiles for them. Devices can then automatically download and apply the WiFi configuration from your
system. You can configure a network access profile, which contains further security layers that are related to
VPN connectivity and HTTP proxy settings.
Wireless LAN Configuration Task Flow

Procedure

Step 1 Generate a Phone Feature List, on page 1 Generate a report to identify devices that
wireless LAN profiles.
Step 2 Configure a Network Access Profile, on page Optional: Configure a network access profile
82 if you want to configure VPN and HTTP proxy
settings that you can link to a wireless LAN
profile.
Step 3 Configure a Wireless LAN Profile, on page 82 Configure a wireless LAN profile with common
WiFi settings to apply to devices or device pools
in the enterprise.
Step 4 Configure a Wireless LAN Profile Group, on Group wireless LAN profiles together.
page 82
Step 5 To Link a Wireless LAN Profile Group to a After you complete the device link, TFTP adds
Device or Device Pool, on page 83, perform the wireless LAN profile group to the existing
one of the following subtasks: device configuration file, which the device (or
81
Configure a Network Access Profile

• Link a Wireless LAN Profile Group to a devices that are tied to a device pool) proceeds
Device, on page 83 to download.
• Link a Wireless LAN Profile Group to a
Device Pool, on page 83
Configure a Network Access Profile

Configure a network access profile if you want to configure VPN and HTTP proxy settings that you can link
to a wireless LAN profile.
Procedure
Step 1 From Cisco Unified CM Administration, choose Device > Device Settings > Network Access Profile
Step 3 Configure the fields in the Network Access Profile Configuration window. For more information on the
fields and their configuration options, see Online Help.
Step 4 Click Save.
Configure a Wireless LAN Profile

Configure a wireless LAN profile with common WiFi settings to apply to devices or device pools in enterprise.
Procedure
Step 1 From Cisco Unified CM Administration, choose Device > Device Settings > Wireless LAN Profile
Step 3 Configure the fields in the Wireless LAN Profile Configuration window. For more information on the fields
Step 4 Click Save.
Configure a Wireless LAN Profile Group

Group your wireless LAN profiles.
Procedure
Step 1 From Cisco Unified CM Administration, choose Device > Device Settings > Wireless LAN Profile Group.
82
Link a Wireless LAN Profile Group to a Device or Device Pool
Step 3 Configure the fields in the Wireless LAN Profile Group Configuration window. For more information on
Step 4 Click Save.
Link a Wireless LAN Profile Group to a Device or Device Pool

After you complete the device link, TFTP adds the wireless LAN profile group to the existing device
configuration file, which the device (or devices tied to a device pool) proceeds to download.
Procedure

Step 1 Link a Wireless LAN Profile Group to a Device,
on page 83
Step 2 Link a Wireless LAN Profile Group to a Device
Pool, on page 83
Link a Wireless LAN Profile Group to a Device
Procedure

• Click Find to enter search criteria and choose an existing device from the resulting list.
• Click Add New, and choose the device type from the Phone Type drop-down list.
Step 3 From the Wireless LAN Profile Group drop-down list, choose a wireless LAN profile group that you created.
Step 4 Click Save.
Link a Wireless LAN Profile Group to a Device Pool

If you link a wireless LAN profile group at the device and device pool level, your system uses the device pool
setting.
Procedure
• Click Find to enter search criteria and choose an existing device pool from the resulting list.
• Click Add New.
Step 3 From the Wireless LAN Profile Group drop-down list, choose a wireless LAN profile group that you created.
83
Link a Wireless LAN Profile Group to a Device Pool
Step 4 Click Save.
84
CHAPTER 9
VPN Client
• VPN Client Overview, on page 85
• VPN Client Prerequisites, on page 85
• VPN Client Configuration Task Flow, on page 85
VPN Client Overview

The Cisco VPN Client for Cisco Unified IP Phone creates a secure VPN connection for employees who
telecommute. All settings of the Cisco VPN Client are configured through Cisco Unified Communications
Manager Administration. After the phone is configured within the Enterprise, the users can plug it into their
broadband router for instant connectivity.
Note The VPN menu and its options are not available in the U.S. export unrestricted version of Unified
VPN Client Prerequisites

Pre-provision the phone and establish the initial connection inside the corporate network to retrieve the phone
configuration. You can make subsequent connections using VPN, as the configuration is already retrieved on
the phone.
VPN Client Configuration Task Flow

Pre-provision the phone and establish the initial connection inside the corporate network to retrieve the phone
configuration. You can make subsequent connections using VPN, as the configuration is already retrieved on
the phone.
85
VPN Client Configuration Task Flow
Procedure

Step 1 Complete Cisco IOS Prerequisites, on page Complete Cisco IOS prerequisites. Perform
87 this action if you want to configure Cisco IOS
VPN.
Step 2 Configure Cisco IOS SSL VPN to Support IP Configure Cisco IOS for VPN client on an IP
Phones , on page 87 Phone. Perform this action if you want to
configure Cisco IOS VPN.
Step 3 Complete ASA Prerequisites for AnyConnect, Complete ASA prerequisites for AnyConnect.
on page 89 Perform this action if you want to configure
ASA VPN.
Step 4 Configure ASA for VPN Client on IP Phone, Configure ASA for VPN client on an IP Phone.
on page 89 Perform this action if you want to configure
ASA VPN.
Step 5 Configure the VPN concentrators for each To avoid long delays when the user upgrades
VPN Gateway. the firmware or configuration information on
a remote phone, set up the VPN concentrator
close in the network to the TFTP or Unified
Communications Manager server. If this is not
feasible in your network, you can set up an
alternate TFTP or load server that is next to
the VPN concentrator.
Step 6 Upload VPN Concentrator Certificates, on Upload the VPN concentrator certificates.
page 91
Step 7 Configure VPN Gateway, on page 92 Configure the VPN gateways.
Step 8 Configure VPN Group, on page 93 After you create a VPN group, you can add
one of the VPN gateways that you just
configured to it.
Step 9 Perform one of the following: You must configure a VPN profile only if you
have multiple VPN groups. The VPN Profile
• Configure VPN Profile, on page 94
fields take precedence over the VPN Feature
• Configure VPN Feature Parameters, on Configuration fields.
page 95
Step 10 Add VPN Details to Common Phone Profile, Add the VPN Group and VPN Profile to a
on page 97 Common Phone Profile.
Step 11 Upgrade the firmware for Cisco Unified IP To run the Cisco VPN client, a supported Cisco
Phone to a version that supports VPN. Unified IP Phone must be running firmware
release 9.0(2) or higher. For more information
about upgrading the firmware, see Cisco
Unified IP Phone Administration Guide for
Unified Communications Manager for
yourCisco Unified IP Phone model.
Step 12 Using a supported Cisco Unified IP Phone, Connect your Cisco Unified IP Phone to a
establish the VPN connection. VPN.
86
Complete Cisco IOS Prerequisites
Complete Cisco IOS Prerequisites

Use this procedure to complete Cisco IOS Prerequisites.
Procedure
Step 1 Install Cisco IOS Software version 15.1(2)T or later.

Feature Set/License: Universal (Data & Security & UC) for IOS ISR-G2 and ISR-G3
Feature Set/License: Advanced Security for IOS ISR
Step 2 Activate the SSL VPN License.
Configure Cisco IOS SSL VPN to Support IP Phones

Use this procedure to complete Cisco IOS SSL VPN to Support IP Phones.
Procedure
Step 1 Configure Cisco IOS locally.

a) Configure the Network Interface.
Example:
router(config)# interface GigabitEthernet0/0
router(config-if)# description "outside interface"
router(config-if)# ip address 10.1.1.1 255.255.255.0
router(config-if)# duplex auto
router(config-if)# speed auto
router(config-if)# no shutdown
router#show ip interface brief (shows interfaces summary)
b) Configure static and default routes by using this command:

router(config)# ip route <dest_ip> < mask> < gateway_ip>
Example:
router(config)# ip route 10.10.10.0 255.255.255.0 192.168.1.1
Step 2 Generate and register the CAPF certificate to authenticate the IP phones with an LSC.
Step 3 Import the CAPF certificate from Unified Communications Manager.
a) From the Cisco Unified OS Administration, choose Security > Certificate Management.
Note This location changes based on the Unified Communications Manager version.
b) Find the Cisco_Manufacturing_CA and CAPF certificates. Download the.pem file and save as.txt file.
c) Create trustpoint on the Cisco IOS software.
hostname(config)# crypto pki trustpoint trustpoint_name
hostname(config-ca-trustpoint)# enrollment terminal
hostname(config)# crypto pki authenticate trustpoint
87
Configure Cisco IOS SSL VPN to Support IP Phones
When prompted for the base 64-encoded CA certificate, copy and paste the text in the downloaded .pem
file along with the BEGIN and END lines. Repeat the procedure for the other certificates.
d) Generate the following Cisco IOS self-signed certificates and register them with Unified Communications
Manager, or replace with a certificate that you import from a CA.
• Generate a self-signed certificate.
Router> enable
Router# configure terminal
Router(config)# crypto key generate rsa general-keys label <name>
<exportable -optional>Router(config)# crypto pki trustpoint <name>
Router(ca-trustpoint)# enrollment selfsigned
Router(ca-trustpoint)# rsakeypair <name> 2048 2048
Router(ca-trustpoint)#authorization username subjectname commonname
Router(ca-trustpoint)# crypto pki enroll <name>
Router(ca-trustpoint)# end
• Generate a self-signed certificate with Host-id check enabled on the VPN profile in Unified
Example:
Router> enable
Router# configure terminal
Router(config)# crypto key generate rsa general-keys label <name>
<exportable -optional>Router(config)# crypto pki trustpoint <name>
Router(ca-trustpoint)# enrollment selfsigned
Router(config-ca-trustpoint)# fqdn <full domain
name>Router(config-ca-trustpoint)# subject-name CN=<full domain
name>, CN=<IP>Router(ca-trustpoint)#authorization username
subjectname commonname
Router(ca-trustpoint)# crypto pki enroll <name>
Router(ca-trustpoint)# end
• Register the generated certificate with Unified Communications Manager.

Example:
Router(config)# crypto pki export <name> pem terminal
Copy the text from the terminal and save it as a.pem file and upload it to the Unified Communications
Manager using the Cisco Unified OS Administration.
Step 4 Install AnyConnect on Cisco IOS.

Download the Anyconnect package from cisco.com and install to flash.
Example:
router(config)#webvpn install svc
flash:/webvpn/anyconnect-win-2.3.2016-k9.pkg
Step 5 Configure the VPN feature.

Note To use the phone with both certificate and password authentication, create a user with the phone
MAC address. Username matching is case sensitive. For example:
username CP-7975G-SEP001AE2BC16CB password k1kLGQIoxyCO4ti9 encrypted
88
Complete ASA Prerequisites for AnyConnect
Complete ASA Prerequisites for AnyConnect

Use this procedure to complete ASA Prerequisites for AnyConnect.
Procedure
Step 1 Install ASA software (version 8.0.4 or later) and a compatible ASDM.
Step 2 Install a compatible AnyConnect package.
Step 3 Activate License.
a) Check features of the current license using the following command:
show activation-key detail
b) If necessary, obtain a new license with additional SSL VPN sessions and enable the Linksys phone.
Step 4 Make sure that you configure a tunnel-group with a non-default URL as follows:
tunnel-group phonevpn type remote-access
tunnel-group phonevpn general-attribute
address-pool vpnpool
tunnel-group phonevpn webvpn-attributes
group-url https://172.18.254.172/phonevpn enable
Consider the following when configuring non-default URL:

• If the IP address of the ASA has a public DNS entry, you can replace it with a Fully Qualified Domain
Name (FQDN).
• You can only use a single URL (FQDN or IP address) on the VPN gateway in Unified Communications
Manager.
• It is preferred to have the certificate CN or subject alternate name match the FQDN or IP address in the
group-url.
• If the ASA certificate CN or SAN does not match with the FQDN or IP address, uncheck the host ID
check box in the Unified Communications Manager.
Configure ASA for VPN Client on IP Phone

Use this procedure to configure ASA for VPN Client on IP Phone.
Note Replacing ASA certificates results in non-availability of Unified Communications Manager.
Procedure
Step 1 Local configuration

a) Configure network interface.
89
Configure ASA for VPN Client on IP Phone
Example:
ciscoasa(config)# interface Ethernet0/0
ciscoasa(config-if)# nameif outside
ciscoasa(config-if)# ip address 10.89.79.135 255.255.255.0
ciscoasa(config-if)# duplex auto
ciscoasa(config-if)# speed auto
ciscoasa(config-if)# no shutdown
ciscoasa#show interface ip brief (shows interfaces summary)
b) Configure static routes and default routes.

ciscoasa(config)# route <interface_name> <ip_address> <netmask> <gateway_ip>
Example:
ciscoasa(config)# route outside 0.0.0.0 0.0.0.0 10.89.79.129
c) Configure the DNS.

Example:
ciscoasa(config)# dns domain-lookup inside
ciscoasa(config)# dns server-group DefaultDNS
ciscoasa(config-dns-server-group)# name-server 10.1.1.5 192.168.1.67 209.165.201.6
Step 2 Generate and register the necessary certificates for Unified Communications Manager and ASA.
Import the following certificates from the Unified Communications Manager.
• CallManager - Authenticating the Cisco UCM during TLS handshake (Only required for mixed-mode
clusters).
• Cisco_Manufacturing_CA - Authenticating IP phones with a Manufacturer Installed Certificate (MIC).
• CAPF - Authenticating IP phones with an LSC.
To import these Unified Communications Manager certificates, do the following:

a) From the Cisco Unified OS Administration, choose Security > Certificate Management.
b) Locate the certificates Cisco_Manufacturing_CA and CAPF. Download the.pem file and save asa .txt
file.
c) Create trustpoint on the ASA.
Example:
ciscoasa(config)# crypto ca trustpoint trustpoint_name
ciscoasa(ca-trustpoint)# enrollment terminal
ciscoasa(config)# crypto ca authenticate trustpoint_name
When prompted for base 64 encoded CA Certificate, copy-paste the text in the downloaded .pem file
along with the BEGIN and END lines. Repeat the procedure for the other certificates.
d) Generate the following ASA self-signed certificates and register them with Unified Communications
Manager, or replace with a certificate that you import from a CA.
• Generate a self-signed certificate.
Example:
ciscoasa> enable
ciscoasa# configure terminal
ciscoasa(config)# crypto key generate rsa general-keys label <name>
ciscoasa(config)# crypto ca trustpoint <name>
90
Upload VPN Concentrator Certificates
ciscoasa(ca-trustpoint)# enrollment self

ciscoasa(ca-trustpoint)# keypair <name>
ciscoasa(config)# crypto ca enroll <name>
ciscoasa(config)# end
• Generate a self-signed certificate with Host-id check enabled on the VPN profile in Unified
Example:
ciscoasa> enable
ciscoasa# configure terminal
ciscoasa(config)# crypto key generate rsa general-keys label <name>
ciscoasa(config)# crypto ca trustpoint <name>
ciscoasa(ca-trustpoint)# enrollment self
ciscoasa(ca-trustpoint)# fqdn <full domain name>
ciscoasa(config-ca-trustpoint)# subject-name CN=<full domain name>,CN=<IP>
ciscoasa(config)# crypto ca enroll <name>
ciscoasa(config)# end
• Register the generated certificate with Unified Communications Manager.

Example:
ciscoasa(config)# crypto ca export <name> identity-certificate
Copy the text from the terminal and save it as a.pem file and upload it to Unified Communications
Manager.
Step 3 Configure the VPN feature. You can use the Sample ASA configuration summary below to guide you with
the configuration.
Note To use the phone with both certificate and password authentication, create a user with the phone
MAC address. Username matching is case sensitive. For example:
ciscoasa(config)# username CP-7975G-SEP001AE2BC16CB password k1kLGQIoxyCO4ti9
encrypted
ciscoasa(config)# username CP-7975G-SEP001AE2BC16CB attributes
ciscoasa(config-username)# vpn-group-policy GroupPhoneWebvpn
ciscoasa(config-username)#service-type remote-access
ASA Certificate Configuration

For more information on ASA certificate configuration, see Configure AnyConnect VPN Phone with
Certificate Authentication on an ASA
Upload VPN Concentrator Certificates

Generate a certificate on the ASA when you set it up to support the VPN feature. Download the generated
certificate to your PC or workstation and then upload it to Unified Communications Manager using the
procedure in this section. Unified Communications Manager saves the certificate in the Phone-VPN-trust list.
The ASA sends this certificate during the SSL handshake, and the Cisco Unified IP Phone compares it against
the values stored in the Phone-VPN-trust list.
91
Configure VPN Gateway
If a Locally Significant Certificate (LSC) is installed on the Cisco Unified IP Phone, it will send its LSC by
default.
To use device level certificate authentication, install the root MIC or CAPF certificate in the ASA, so that the
Cisco Unified IP Phone are trusted.
To upload certificates to Unified Communications Manager, use the Cisco Unified OS Administration.
Procedure
Step 1 From Cisco Unified OS Administration, choose Security > Certificate Management.
Step 2 Click Upload Certificate.
Step 3 From the Certificate Purpose drop-down list, choose Phone-VPN-trust.
Step 4 Click Browse to choose the file that you want to upload.
Step 5 Click Upload File.
Step 6 Choose another file to upload or click Close.
For more information, see Certificate Management chapter.
Configure VPN Gateway

Ensure that you have configured VPN concentrators for each VPN gateway. After configuring the VPN
concentrators, upload the VPN concentrator certificates. For more information, see Upload VPN Concentrator
Certificates, on page 91.
Use this procedure to configure the VPN Gateway.
Procedure
Step 1 From Cisco Unified CM Administration, choose Advanced Features > VPN > VPN Gateway.
a) Click Add New to configure new profile.
b) Click the Copy next to the VPN gateway that you want to copy.
c) Locate the appropriate VPN gateway and modify the settings to update an existing profile.
Step 3 Configure the fields in the VPN Gateway Configuration window. For more information, see VPN Gateway
Fields for VPN Client, on page 92.
Step 4 Click Save.
VPN Gateway Fields for VPN Client

The table describes the VPN Gateway fields for VPN Client.
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Configure VPN Group
Table 10: VPN Gateway Fields for VPN Client
Field Description
VPN Gateway Name Enter the name of the VPN gateway.
VPN Gateway Enter a description of the VPN gateway.

Description
VPN Gateway URL Enter the URL for the main VPN concentrator in the gateway.
Note You must configure the VPN concentrator with a group URL and
use this URL as the gateway URL.
For configuration information, refer to the documentation for the VPN

concentrator, such as the following:
• SSL VPN Client (SVC) on ASA with ASDM Configuration Example
VPN Certificates in this Use the up and down arrow keys to assign certificates to the gateway. If you do
Gateway not assign a certificate for the gateway, the VPN client fails to connect to that
concentrator.
Note You can assign up to 10 certificates to a VPN gateway, and you
must assign at least one certificate to each gateway. Only certificates
that are associated with the Phone-VPN-trust role appear in the
available VPN certificates list.
Configure VPN Group

Use this procedure to configure VPN Group.
Procedure
Step 1 From Cisco Unified CM Administration, choose Advanced Features > VPN > VPN Group.
a) Click Add New to configure new profile.
b) Click Copy next to the VPN group that you want to copy an existing VPN group.
c) Locate the appropriate VPN group and modify the settings to update an existing profile.
Step 3 Configure the fields in the VPN Group Configuration window. For more information, see VPN Gateway
Fields for VPN Client, on page 92 for the field description details.
Step 4 Click Save.
VPN Group Fields for VPN Client

The table describes the VPN Group Fields for VPN Client.
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Configure VPN Profile
Table 11: VPN Group Fields for VPN Client
Field Definition
VPN Group Name Enter the name of the VPN group.
VPN Group Description Enter a description of the VPN group.
All Available VPN Scroll to see all available VPN gateways.

Gateways
Selected VPN Gateways Use the up and down arrow buttons to move available VPN gateways into and
in this VPN Group out of this VPN group.
If the VPN client encounters critical error and cannot connect to a particular VPN
gateway, it will attempt to move to the next VPN gateway in the list.
Note You can add up to a maximum of three VPN gateways to a VPN
group. Also, the total number of certificates in the VPN group cannot
exceed 10.
Configure VPN Profile

Use this procedure to configure the VPN Profile.
Procedure
Step 1 From Cisco Unified CM Administration, choose Advanced Features > VPN > VPN Profile.
a) Click Add New to configure new profle.
b) Click Copy next to the VPN profile that you want to copy an existing profile.
c) To update an existing profile, specify the appropriate filters in the Find VPN Profile Where, click Find,
and modify the settings.
Step 3 Configure the fields in the VPN Profile Configuration window. For more information, see VPN Profile
Fields for VPN Client, on page 94 for the field description details.
Step 4 Click Save.
VPN Profile Fields for VPN Client

The table describes the VPN profile field details.
Table 12: VPN Profile Field Details
Field Definition
Name Enter a name for the VPN profile.
94
Configure VPN Feature Parameters
Field Definition
Description Enter a description for the VPN profile.
Enable Auto Network When you check this check box, the VPN client can only run when it detects that
Detect it is out of the corporate network.
Default: Disabled.
MTU Enter the size, in bytes, for the Maximum Transmission Unit (MTU).
Default: 1290 bytes.
Fail to Connect This field specifies the amount of time to wait for login or connect operations to
complete while the system creates the VPN tunnel.
Default: 30 seconds
Enable Host ID Check When you check this check box, the gateway certificate subjectAltName or CN
must match the URL to which the VPN client is connected.
Default: Enabled
Client Authentication From the drop-down list, choose the client authentication method:
Method
• User and password
• Password only
• Certificate (LSC or MIC)
Enable Password When you check this check box, a user password gets saved in the phone until
Persistence either a failed log in attempt occurs, a user manually clears the password, or the
phone resets or loses power.
Configure VPN Feature Parameters

Procedure
Step 1 From Cisco Unified CM Administration, choose Advanced Features > VPN > VPN Feature Configuration.
Step 2 Configure the fields in the VPN Feature Configuration window. For more information, see VPN Feature
Parameters, on page 96.
Step 3 Click Save.
Perform the following tasks:
• Upgrade the firmware for Cisco Unified IP Phones to a version that supports VPN. For more information
about upgrading the firmware, see Cisco Unified IP Phone Administration Guide for your Cisco Unified
IP Phone model.
• Using a supported Cisco Unified IP Phone, establish the VPN connection.
95
VPN Feature Parameters
VPN Feature Parameters

The table describes the VPN feature parameters.
Table 13: VPN Feature Parameters
Field Default
Enable Auto Network When True, the VPN client can only run when it detects that it is out of the
Detect corporate network.
Default: False
MTU This field specifies the maximum transmission unit:

Default: 1290 bytes
Minimum: 256 bytes
Maximum: 1406 bytes
Keep Alive This field specifies the rate at which the system sends the keep alive message.
Note If it is non zero and less than the value specified in Unified
Communications Manager, the keep alive setting in the VPN
concentrator overwrites this setting.
Default: 60 seconds
Minimum: 0
Maximum: 120 seconds
Fail to Connect This field specifies the amount of time to wait for login or connect operations to
complete while the system creates the VPN tunnel.
Default: 30 seconds
Minimum: 0
Maximum: 600 seconds
Client Authentication From the drop-down list, choose the client authentication method:
Method
• User and password
• Password only
• Certificate (LSC or MIC)
Default: User And Password
Enable Password When True, a user password gets saved in the phone, if Reset button or “**#**”
Persistence is used for reset. The password does not get saved and the phone prompts for
credentials if the phone loses power or you initiate a factory reset.
Default: False
96
Add VPN Details to Common Phone Profile
Field Default
Enable Host ID Check When True, the gateway certificate subjectAltName or CN must match the URL
to which the VPN client is connected.
Default: True

Use this procedure to add VPN details to common phone profile.
Procedure
Step 1 From Cisco Unified CM Administration, choose Device > Device Settings > Common Phone Profile.
Step 2 Click Find and choose common phone profile to which you want to add the VPN details.
Step 3 In the VPN Information section, choose the appropriate VPN Group and VPN Profile.
Step 4 Click Save and then Apply Config.
Step 5 Click OK in apply configuration window.
97
98
PA R T IV
Licensing
• Licensing, on page 101
CHAPTER 10
Licensing
• Licensing, on page 101
• Unified Communications Manager Licensing, on page 102
• License Compliance, on page 103
• User Only Licensing, on page 104
• Device Only, on page 104
• User and Device, on page 104
• Maximum Number of Devices Per User, on page 111
• TelePresence Room License, on page 111
• License Substitution, on page 111
• Licensing Scenarios, on page 112
• Adding Users, on page 112
• Adding Unassociated Devices, on page 112
• Adding Users with Associated Devices, on page 113
• Number of Devices Per User, on page 114
• License Usage Report, on page 114
• Cisco Unified Reporting, on page 115
Licensing
Cisco Unified Communications Manager licensing is part of the overall commercial offer of Cisco Unified
Communications Licensing.
User Connect User Connect User Connect Unified

Licensing Licensing Licensing Workspace
(Essential) (Basic) (Enhanced/ Licensing
Enhanced
Plus)
Cisco Unified Mobile Connect Not Available Included Included Included

CM Features (SNR)
101
Licensing
Unified Communications Manager Licensing
User Connect User Connect User Connect Unified

Licensing Licensing Licensing Workspace
(Essential) (Basic) (Enhanced/ Licensing
Enhanced
Plus)
Device Support Number of 1 1 1/2 10

Devices
Device Type Analog/Voice Voice (for Voice (for Voice (for

Support (for details, see details, see the details, see the details, see
the User and User and Device User and Device theUser and
Device table) table) table) Device table)
Number of User 1 1 1 1
Profiles
Clients Jabber Mobile Not Available Not Available Included Included
Jabber Desktop Not Available Not Available Included Included
Jabber Included Included Included Included

IM/Presence
Application Webex Add-on Add-on Add-on Included

Meetings
Webex Social Add-on Add-on Add-on Included
Unity Add-on Add-on Add-on Included

Connection
Cisco Unified Included Included Included Included
CM
Licensing for the Cisco Unified Communications Manager is determined by the total number of users, user
features, and devices configured. Cisco Unified Communications Manager calculates its license usage based
upon the total number of users (with user features and associated devices) and devices configured on the
system. Cisco Unified Communications Manager reports the total license usage (per publisher) to the Cisco
Smart Software Manager and gets back the license compliance or non-compliance status.
Unified Communications Manager Licensing

Cisco Unified Workspace Licensing (UWL) allows organizations to access a wide range of Cisco Collaboration
applications and services in a cost-effective, simple package. It includes soft clients, application server software,
and licensing on a per user basis.
Cisco User Connect Licensing (UCL) is a user-based license for individual Cisco Unified Communications
products. It includes a soft client, application server software licensing, and basic unified communications
applications. Depending on your needs and device of choice, UCL is available in Essential, Basic, Enhanced,
or Enhanced Plus.
The following are the license types for the Unified Communications Manager:
102
Licensing
License Compliance
UC Manager Essential Essential User Connect License - supports one device

providing basic voice or analog device (phone or fax).
(For example: analog phone, ATA 186, ATA 187,
Cisco 3905, Cisco 6901)
UC Manager Basic Basic User Connect License - supports one device,

including all Essential devices, plus basic (voice and
video) call control features. (For example: Cisco 6911,
Cisco 6921)
UC Manager Enhanced Enhanced User Connect License - supports one device,

including all Basic devices, plus advanced (voice and
video) call control features including desktop, mobile
clients. (For example: Cisco 3911, Cisco 3951, Cisco
6941, Cisco 6945, Cisco 6961, Cisco 79xx, Cisco
89xx, Cisco 99xx, Cisco E20, Cisco TelePresence
EX60, Cisco TelePresence EX90, third party SIP)
UC Manager Enhanced Plus Enhanced Plus User Connect License - supports up

to two devices, and including all Enhanced devices.
UC Manager CUWL Supports advanced (voice and video) call control

features including desktop and mobile, professional
collaboration workspace application features with a
maximum of ten devices per user.
UC Manager TelePresence Room TelePresence Room license - supports room based

immersive and multipurpose Cisco TelePresence
System endpoints and Spark Room. (For example:
Cisco TelePresence System Series 3200, 3000, 1300;
Cisco TelePresence MX Series; Cisco TelePresence
TX Series; Cisco TelePresence System Profile Series)
License Compliance
When first installed, the Unified Communications Manager is fully operational in demonstration mode for an
evaluation period of 90 days, until it has successfully registered with the Cisco Smart Software Manager.
After registration, the Unified Communications Manager communicates with Cisco Smart Software Manager
periodically. The Unified Communications Manager reports the total license requirements by license type to
the Cisco Smart Software Manager and then gets back the license status.
Licenses in the non-compliant state for Unified Communications Manager are enforced after a 90-day overage
period. At the conclusion of the grace period, Unified Communications Manager enforces non-compliance
with the following service degradation:
Devices and Users cannot be provisioned. Changing the configuration of a user that affects licensing (For
example: the Enable/Disable IM and Presence and the Enable/Disable Mobility check boxes) is not allowed.
For information about smart licensing operations, see the System Configuration Guide for Cisco Unified
Communications Manager
103
Licensing
User Only Licensing
User Only Licensing

If a user is configured on the system and is not associated with a device, that user does not own any devices
and is a "User Only." A user is associated with a device or owns the device if that user's user ID is entered in
the OwnerUserID field of the device. The licensing for a "User Only" is shown in the User and Device Support
table, for the user not associated with any devices.
Simply adding a user to the system does not consume a license if that user does not own any devices or does
not use a licensed user feature. If, however, the user is configured with a licensed user feature, or that user
does own a device, then the user does consume a license. The only licensed feature currently is Mobile Connect
(also known as Mobility or Single Number Reach or SNR).
Mobile Connect (or Mobility or Single Number Reach) for a user is configured when a Remote Destination
Profile (RDP) has been created with the end-user set as the Device owner (User ID field).
Device Only
If a device is added to Cisco Unified Communications Manager and does not have an entry for OwnerUserID
field in its Device Configuration window, then the device is not assigned or not associated to a user and called
"Device Only". The licensing for "Device Only" devices is listed in the Cisco Unified Communications
Manager Licensing - User and Device Support table. If a device is added to Cisco Unified Communications
Manager and does not have an entry for OwnerUserID, then the device would require the minimum license
type determined by device type, as shown in the Licensing - User and Device Support table.
User and Device

Once a device is assigned or associated with a user, by entering a user ID in the OwnerUserID field of the
device, the licensing requirements for that user and device are determined by the type of device and the number
of devices assigned to the user. For a user that owns one device, if the user ID of that user is added as
OwnerUserID to one Essential device (such as a 3905, 6901, or analog device), then the User and Device
minimum license that is required is an Essential license. This means that the one Essential license supports
both the user and the device. If instead, that user ID of that user is added as OwnerUserID to one Basic device
(such as 6911 or 6921), then the user and device minimum license that is required is one Basic icense. If the
user ID of a user is added as OwnerUserID to one Enhanced device, the user and device minimum license
that is required is an Enhanced license.
For users that own more than one device, the minimum licensing is determined by the number of devices that
are owned by the user. Cisco Unified Communications Manager Licensing table shows the maximum number
of devices per user license that is supported. A user that owns two devices requires an Enhanced Plus license
at a minimum. A user that owns more than two devices requires a CUWL license at a minimum.
Cisco Unified Communications Manager Licensing - User and Device Support table summarizes Cisco Unified
Communications Manager Licensing for User Only, Device Only, and User and Device.
104
Licensing
User and Device
Table 14: Cisco Unified Communications Manager Licensing - User and Device Support
License Type Device Only User and Device User Only
UC Manager Essential • Cisco Unified SIP A user with 1 Essential N/A

Phone 3905 device.
• Cisco Unified IP
Phone 6901
• Analog devices
UC Manager Basic Cisco Unified IP Phone A user with 1 Basic A user with Single
6911 and 6921 models device. Number Reach (Mobile
Connect).
OR OR
OR
Any device from UC A user and associated
Manager Essential device from the UC A user with an UC
license type. Manager Essential Manager Essential
license type. license type.
105
Licensing
User and Device
UC Manager Enhanced A user with 1 Enhanced N/A

device.
OR
A user and associated
device with UC Manager
Essential or UC
Manager Basic license
type.
106
Licensing
User and Device

Phone 3911, 3941,
3951
Phone 6941, 6945,
and 6961 models
Phone 7900 Series
(7900G, 7911G,
7912G, 7931G,
794xG, 796xG, and
7975G models)
Phone 8900 Series
(8941, 8945, and
8961 models)
Phone 9900 Series
(9951 and 9971
models) with or
without a camera
• Cisco Unified
Wireless IP Phones
Series (792xG and
7925G-EX models)
Conference Stations
(7936G and 7937G
stations)
• Cisco Unified
Softphones (Cisco
Unified Personal
Communicator,
Cisco UC Integration
for Lync, Cisco UC
Integration for
Connect, and Cisco
IP Communicator)
• Jabber clients
(Jabber for Mac,
Jabber for Windows,
Jabber for iPhone,
Jabber for Android,
Jabber for iPad, and
107
Licensing
User and Device

Jabber SDK)
• Cisco Virtual
Experience Clients
(VXC) with voice
and video firmware
• Cisco IP Video
Phone E20
• Cisco TelePresence
System EX Series
(EX60 and EX90)
• Third-party SIP
devices
• Cisco Desktop
Collaboration
Experience DX600
Series
• Transnova S3
• Cisco Spark Room
Device
• IMS
OR
Any device from the UC
Manager Essential or UC
Manager Basic license
type.
UC Manager Enhanced N/A A user with 2 devices. N/A

Plus
OR
A user and associated
devices with UC
Manager Essential, UC
Manager Basic, UC
Manager Enhanced, or
UC Manager Enhanced
Plus license type.
108
Licensing
User and Device
UC Manager A user with 1 UC N/A

TelePresence Room Manager TelePresence
License Room device associated.
109
Licensing
User and Device

System 500 Series
System 1100
System 1300 Series
System 3000 Series
System 3200 Series
TX9000 Series
(TX9000, TX9200)
TX1300 Series
System Profile
Series (42-inch 6000
MXP, 52- inch
MXP, 52- inch Dual
MXP, 65-inch, and
65- inch Dual)
System Codecs
C90/C60/C40
System Quick Set
C20
MX Series (MX300
and MX200)
1000
SX series
• Cisco Webex devices
• Generic Desktop
Video Endpoint
• Generic Multiple
Screen Room
110
Licensing
Maximum Number of Devices Per User

System
• Generic Single
Screen Room
System
Device Only means a device configured in Cisco Unified Communications Manager that does not have a user
association, where the OwnerUserID field is blank.
User and Device means a device configured in Cisco Unified Communications Manager that has a user
associated, the OwnerUserID field has a registered userid.
User Only means a user configured in Cisco Unified Communications Manager that does not have any devices
associated with the user - whose user id is not found as OwnerUserID for any Cisco Unified Communications
Manager devices.
Bold text in the above table indicates that a device is supported through license substitution where an available
license of the license type listed may be used to meet lower-level license requirements. This is done in Cisco
Smart Software Manager.
Note MGCP FXS ports do not require any license because they are not considered analog phones.
Maximum Number of Devices Per User

The Essential, Basic, and Enhanced licenses support users with one associated device, where the user's id is
entered in the OwnerUserId field of one device. The Enhanced Plus license supports users with two associated
devices. UWL supports users with three and up to ten associated devices.
TelePresence Room License

Multi-purpose and immersive TelePresence devices are licensed under a separate device license type that is
called the TelePresence Room license. The TelePresence Room license covers both the TelePresence device
and phone that is registered to Cisco Unified Communications Manager, only if the same userid is entered as
the OwnerUserID field for the TelePresence device and the phone. If the same userid is not entered as
OwnerUserID for both the TelePresense device and the phone, then the devices are not associated and two
licenses are required: one TelePresence Room license for the device and one Enhanced for the phone. The
TelePresence touch device does not register to the Cisco Unified Communications Manager, and therefore
does not require a separate license or the OwnerUserID association.
License Substitution
The Cisco Smart Software Manager (CSSM) allows for tiered license substitution of available licenses to
enable compliance. The available higher-level licenses are substituted or loaned to meet lower level license
requirements. For example, if a customer has 100 UC Manager CUWL licenses installed, however Cisco
111
Licensing
Licensing Scenarios
Unified Communications Manager is reporting back license requirements for 10 CUWL licenses and 50 UC
Manager Enhanced Plus licenses, CSSM will calculate that there are 100-10 or 90 UC Manager CUWL license
available to be loaned to lower tiers. Of the 90 UC Manager CUWL available licenses 50 CUWL would then
be used to meet the requirements for the 50 Enhanced Plus licenses. CSSM will show 40 UC Manager CUWL
licenses as available.
Note When Cisco Smart Software Manager On-Prem (Cisco SSM On-Prem) or Smart Software Manager satellite
is used in Unified Communications Manager for licensing, there is a difference in the way the license Hierarchy
Substitution breakdown is displayed in CSSM, when compared to Cisco SSM On-Prem. See the Cisco SSM
On-Prem user interface for details on the insufficient license information if the license authorization status of
Unified CM is Out of Compliance. Refer CSCwf47221 for more details.
Note If the Virtual account is already used by Product Instance using direct communication and license reserved
for Specific License Reservation, the available license quantities are shown incorrectly. Refer CSCwf47223
for more details.
Licensing Scenarios
The following licensing scenarios will walk through the configuration changes on the Cisco Unified
Communications Manager Administration that result in licensing requirements.
Adding Users
When a new user (UserA) is first added to Cisco Unified Communications Manager Administration through
the End User configuration or through the Bulk Administration tool, if the user does not have remote device
profiles under Enable Mobility, then the new user does not require a license.
If a new user (UserB) is first added to Cisco Unified Communications Manager with remote destination
profiles configured under Enable Mobility, then the new user, UserB, requires a Basic license.
UserID Licensed User Feature License Required Note
UserA None None With no assigned devices
UserB Mobility Basic With no assigned devices
Adding Unassociated Devices

If a new device is registered to Cisco Unified Communications Manager and there is no user id entered in the
OwnerUserID field for the device, then the device is unassociated to a user, and requires the license per device
type for unassociated devices, as indicated in Cisco Unified Communications Manager Licensing - User and
Device Support table. For example, Device6901 is added and it requires an Essential license. Device6921 is
added and it requires a Basic license. DeviceEX60 is added and it requires an Enhanced device.
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Licensing
Adding Users with Associated Devices
There are currently no devices that require an Enhanced Plus, CUWL Standard, or CUWL Professional license.
So you will not see a requirement in Cisco Unified Communications Manager for an unassociated device that
requires an Enhanced Plus or above license.
Table 15: Example Device Only License Requirements
Device License Required Note
Device6901 UC Manager Essential With no OwnerUserID
Device6921 UC Manager Basic With no OwnerUserID
DeviceEX60 UC Manager Enhanced With no OwnerUserID
Adding Users with Associated Devices

When a device is added, if the device is associated with a user, then the user and device share a license. For
one device per user, the license that is required is the greater of the user license or device license required.
The following scenarios review the different combinations of device and user associations for one device per
user.
Essential Device Associated to User

If Device6901 (an Essential device) is assigned to UserA, by entering OwnerUserID = UserA, then both the
device and user are supported by one Essential license.
If however, Device6901 (an Essential device) is assigned to UserB (a Basic user), by entering OwnerUserID
= UserB then both device and user are supported by one Basic License.
Basic Device Associated to User

If Device6921 (a Basic device) is assigned to UserA by entering OwnerUserID = UserA, then both the device
and user are supported by one Basic license. Similarly, if Device6921 (a Basic device) is assigned to UserB
(a Basic user) by entering OwnerUserID = UserB, then both the device and user are supported by one Basic
license.
Enhanced Device Associated to User

Most physical phones, soft clients, and desktop video devices such as the EX60 and EX90 are included in the
Enhanced device level. If Device EX60 (an Enhanced device) is assigned to UserA by entering OwnerUserID
= UserA, then both the device and user is supported by one Enhanced license. Similarly, if DeviceEX60 (an
Enhanced device) is assigned to UserB (a Basic user) by entering OwnerUserID = UserB, then both the device
and user are supported by one Enhanced license.
Table 16: Example Users and Device License Requirements
Device OwnerUserID Licensed User Feature License Required
Device6901 UserA None UC Manager Essential
UserB Mobility UC Manager Basic
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Licensing
Number of Devices Per User
Device OwnerUserID Licensed User Feature License Required
Device6921 UserA None UC Manager Basic
UserB Mobility UC Manager Basic
DeviceEX60 UserA None UC Manager Enhanced
UserB Mobility UC Manager Enhanced
Number of Devices Per User

The above examples for Users and Devices apply only when a user is associated with one device - where their
userid is found in only one device configuration OwnerUserID field. When a user is associated with more
than one device, then higher level licenses are required independent of device type.
If UserA is assigned to OwnerUserID for one device then the scenarios above apply. If, however, UserA is
assigned OwnerUserID for two devices, then one Enhanced Plus license is required for both the user and the
two associated devices. If UserA is assigned OwnerUserID for more than two devices, then one UWL Standard
license is required. UserA can be assigned up to ten devices with one UWL Standard license. If more than
ten devices are assigned to one user, then the user requires one UWL Standard license and also requires an
additional license for the additional device.
License Usage Report

Usage details are available by license type, users, and unassigned devices. Usage information is updated once
every six hours and maybe updated manually by clicking on Update Usage Details. Clicking Update Usage
Details is a resource-intensive process and may take a few minutes depending on the size of your system.
There is a link provided to review the Unified Communications licensing information in View all license
type descriptions and device classifications.
The Status message displays if there is an alarm or licensing alert (license non-compliance). See Alarms alerts
and license status notifications for further information on status messages. See License Compliance for further
information on license compliance and non-compliance.
The License Requirements by Type table shows the current system license requirements. It shows current
license usage (number of licenses required) by license type and summarizes the number of users and unassigned
devices that are requiring licenses by license type. The Report links by license type are provided by (number
of) Users or (number of) Unassigned devices and allow drill-down links. For the User report, the user id link
provides details on user configuration per the user id. The view details link provides license requirements per
user id. For the Unassigned Devices report, the Device Type and License Type that is required is displayed
for each unassigned device.
License Usage Reports are also available summarized by Users and Unassigned devices. The Users row lists
the total number of users configured on the system. View Usage Report for the users provides a report for all
users configured on the system and their corresponding license requirements. View Usage Report for the
Unassigned Devices shows the total number of unassigned devices (devices with no associated user).
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Licensing
Cisco Unified Reporting
Note Assigning a user ID to a device using Cisco Unified Communications Administration moves the device from
"Unassigned Devices" to "Users" in the License Usage Report. However, adding a device to the list of controlled
devices for an end-user does not modify the "License Usage Report" results for the device.

The following reports are available from the Cisco Unified Reporting console for Cisco Unified
Communications Solutions.
1. From Cisco Unified Communications Manager Administration login page Navigation bar, click Cisco
Unified Reporting.
2. Choose System Reports.
3. Choose Unified CM Device Counts Summary.
The generated report will summarize, per cluster, the device counts by model.
Unified Reporting.
3. Choose Unified CM User Device Count.
The generated report will summarize, per cluster, the phone to user relationship the number of phones with
no users, users with one phone, and users with more than one phone.
Unified Reporting.
3. Choose Unified CM User Device Count.
The generated report will summarize, per cluster, the phone to user relationship the number of phones with
no users, users with one phone, and users with more than one phone.
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Licensing
116
PA R T V
Monitoring and Recording
• Silent Monitoring , on page 119
• Recording , on page 127
CHAPTER 11
Silent Monitoring
• Silent Monitoring Overview, on page 119
• Silent Monitoring Prerequisites, on page 120
• Configure Silent Monitoring Task Flow, on page 120
• Silent Monitoring Interactions, on page 125
• Silent Monitoring Restrictions, on page 126
Silent Monitoring Overview

Silent call monitoring allows a supervisor to eavesdrop on a phone conversation. The most common scenario
is in a call center where a call agent is speaking with a customer. Call centers need to be able to guarantee the
quality of customer service that an agent in a call center provides. With silent monitoring, the supervisor can
hear both call participants, but neither of the call participants can hear the supervisor.
Silent monitoring can only be invoked by a CTI application through the JTAPI or TAPI interfaces. Many
Cisco applications, such as Cisco Unified Contact Center Enterprise and Cisco Unified Contact Center Express
have the ability to use silent monitoring. Any CTI application that monitors calls must have the corresponding
monitoring privileges that are enabled for the application-user or end-user account.
Silent monitoring is call based. When a supervisor invokes a silent monitoring session, the following occurs:
• The supervisor selects a specific call to be monitored.
• The start-monitoring request from the application triggers the supervisor phone to go off hook and
automatically triggers a monitoring call to the agent.
• The agent phone automatically answers the monitoring call. The monitoring call does not get presented
to the agent.
Secure Silent Monitoring

You can also configure secure silent monitoring. Secure silent monitoring allows encrypted media (sRTP)
calls to be monitored. Monitoring calls are always established using the highest level of security that is
determined by the capabilities of the agent phone regardless of the security status of the call being observed.
The highest level of security is maintained by exchanging the secure media key in any call between the
customer, agent, and supervisor. Monitoring calls using secured media carries approximately 4000 bits per
second of additional bandwidth overhead, same as standard secure media (sRTP) calls.
119
Silent Monitoring Prerequisites
If the agent phone has encryption that is enabled, the supervisor phone must also have encryption enabled in
order to allow secure silent monitoring. If the agent phone has encryption that is enabled, but the supervisor
phone does not, the monitoring request fails.
Whisper Coaching
Unified Communications Manager also supports whisper coaching, a CTI enhancement on silent monitoring
whereby a supervisor can speak to the agent while the monitoring session is underway without the customer
hearing. Whisper coaching can only be initiated by a CTI application. If silent monitoring is already configured,
then no additional configuration of Unified Communications Manager is required for whisper coaching.
Silent Monitoring Prerequisites

Silent monitoring can only be invoked by an external CTI application. Cisco applications such as Cisco Unified
Contact Center Enterprise or Cisco Unified Contact Center Express can initiate silent monitoring sessions.
For details, see the following:
• Cisco Unified Contact Center Enterprise—For details on how to set up silent monitoring in Cisco Unified
Contact Center Enterprise, see Cisco Remote Silent Monitoring Installation and Administration Guide.
• Cisco Unified Contact Center Express—This chapter contains a sample configuration to set up Silent
Monitoring for Unified Contact Center Express via Cisco Finesse. For additional documentation that is
related to your Cisco Unified Contact Center Express deployment, go to https://www.cisco.com/c/en/us/
support/customer-collaboration/unified-contact-center-express/tsd-products-support-series-home.html.
Configure Silent Monitoring Task Flow

This task flow describes the tasks that you must perform within Unified Communications Manager to allow
CTI applications to use the monitoring feature.
Before you begin

• Determine which phones support silent monitoring by running a phone feature list report. For more
information,Generate a Phone Feature List, on page 1
Procedure

Step 1 Perform one of the following procedures: Turn on the Built in Bridge on agent phones.
You can use a service parameter to configure
• Enable Built in Bridge for Phones
the clusterwide default setting or you can enable
Clusterwide, on page 121
the Built in Bridge for individual phones.
• Enable Built in Bridge for a Phone, on
page 121 Note The Built in Bridge setting on
individual phones overrides the
clusterwide default setting.
120
Enable Built in Bridge for Phones Clusterwide

Step 2 Enable Monitoring Privileges for Supervisor, Add the supervisor to a group that allows silent
on page 122 monitoring.
Step 3 Assign a Monitoring Calling Search Space, on Set up the monitoring calling search space for
page 122 the supervisor phone.
Step 4 Configure Silent Monitoring Notification Tones, Configure whether you want to play notification
on page 123 tones to the call participants.
Step 5 Configure Secure Silent Monitoring, on page Optional. If your calls are encrypted, configure
123 secure silent monitoring.
Step 6 Configure Silent Monitoring for Unified For Unified Contact Center Express
Contact Center Express, on page 124 deployments, configure Silent Monitoring via
Cisco Finesse.
Enable Built in Bridge for Phones Clusterwide

When you set the Built-in-Bridge clusterwide service parameter to enable, the Built-in-Bridge default setting
for all phones in the cluster is changed to enabled. However, the Built-in-Bridge setting in the Phone
Configuration window for individual phones overrides the clusterwide service parameter.
Procedure
Step 2 From the Server drop-down list, choose the server on which the CallManager service is running.
Step 4 Set the Builtin Bridge Enable service parameter to On.
Step 5 Click Save.
Enable Built in Bridge for a Phone

Use this procedure to enable the Built in Bridge on an individual phone. The Built in Bridge setting on an
individual phone overrides the clusterwide service parameter.
Before you begin

Use a service parameter to set the Built in Bridge defaults for all phones in the cluster. For details, see Enable
Built in Bridge for Phones Clusterwide, on page 121.
Procedure

Step 2 Click Find to select the agent phone.
121
Enable Monitoring Privileges for Supervisor
Step 3 From the Built in Bridge drop-down list, choose one of the following options:
• On—The Built in Bridge is enabled.
• Off—The Built in Bridge is disabled.
• Default—The setting of the clusterwide Builtin Bridge Enable service parameter is used.
Step 4 Click Save.
Enable Monitoring Privileges for Supervisor

In order for a supervisor to be able to monitor agent conversations, the supervisor must be part of a group that
allows monitoring.
Before you begin

Perform one of the following procedures to enable the Built in Bridge on agent phones:
• Enable Built in Bridge for Phones Clusterwide, on page 121
• Enable Built in Bridge for a Phone, on page 121
Procedure
Step 2 Select the supervisor from the list of users.
Step 3 In the Permissions Information section, click Add to Access Control Group.
Step 4 Add the Standard CTI Allow Call Monitoring and Standard CTI Enabled user groups.
Step 5 Click Save.
Assign a Monitoring Calling Search Space

For monitoring to work, you must assign a Monitoring Calling Search Space to the supervisor phone line.
The Monitoring Calling Search Space must include both the supervisor phone line and the agent phone line.
Procedure

Step 2 Click Find and select the supervisor phone.
The left navigation pane displays the available phone lines for the supervisor's phone.
Step 3 Perform the following steps for each of the supervisor's phone lines that are used for monitoring:
a) Click the phone line. The Directory Number Configuration window displays configuration information
for that phone line.
b) From the Monitoring Calling Search Space drop-down list, choose a calling search space that includes
both the supervisor phone line and the agent phone line.
122
Configure Silent Monitoring Notification Tones
c) Click Save.
Configure Silent Monitoring Notification Tones

In certain jurisdictions, a notification tone must be played to either the agent, the customer, or both, that
indicates that the call is being monitored. By default, Unified Communications Manager does not play
notification tones. You must configure a service parameter to allow notification tones.
Procedure
Step 2 From the Server drop-down list, choose the server one which the CallManager service is running.
Step 4 Configure values for the following service parameters:
• If you want to play a notification tone to the agent, change the value of the Play Monitoring Notification
Tone To Observed Target service parameter to True.
• If you want to play a notification tone to the customer, change the value of the Play Monitoring
Notification Tone To Observed Connected Parties service parameter to True.
Step 5 Click Save.

Step 6 Reset the agent phone, if you changed the service parameter configuration.
Configure Secure Silent Monitoring

To configure secure silent monitoring using sRTP, you must configure phone security profiles that include
encryption and apply them to the supervisor phone and to any agent phones that are being monitored.
Procedure

Step 1 Configure an Encrypted Phone Security Profile Configure phone security profiles that include
, on page 123 encryption for the agent phone and supervisor
phone.
Step 2 Assign Security Profile to Phone, on page 124 Apply the encrypted phone security profile to
the agent phone and the supervisor phone.
Configure an Encrypted Phone Security Profile

To configure secure silent monitoring, you must configure the phone security profile for your supervisor
phone and any agent phones to specify Encrypted as the Device Security Mode.
123
Assign Security Profile to Phone
Procedure
Step 2 Perform either of the following steps:
• Click Add New to create a new phone security profile.
• Click Find and select an existing phone security profile.
Step 3 If you have created a new phone security profile, select your phone model from the Phone Security Profile
Type drop-down list.
Step 4 Enter a Name for the Phone Security Profile.
Step 5 From the Device Security Mode drop-down list, choose Encrypted.
Step 6 Click Save.
Step 7 Repeat the above steps to configure phone security profiles for your supervisor phone and any agent phones.
Assign Security Profile to Phone

Perform the following steps to assign a phone security profile to a phone. For secure silent monitoring to
work, you must assign the phone security profile to both the agent phone and the supervisor phone.
Procedure
Step 1 From Cisco Unified CM Administration, chooseI Device > Phone.

Step 2 Click Find and select the agent phone on which you want to configure a phone security profile.
Step 3 From the Device Security Profile drop-down list, choose the phone security profile that you have set up.
Step 4 Click Save.
Step 5 Repeat the previous steps for the supervisor phone.
Configure Silent Monitoring for Unified Contact Center Express

The following steps contain a sample Silent Monitoring for Cisco Unified Contact Center Express configuration
via Cisco Finesse.
Before you begin

Make sure that both the agent and supervisor phone are compatibile for Cisco Finesse. Refer to the Unified
CCX Software Compatibility Matrix at https://www.cisco.com/c/en/us/support/customer-collaboration/
unified-contact-center-express/products-device-support-tables-list.html.
Procedure
Step 1 Configure a test agent and supervisor on Unified Contact Center Express.
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Silent Monitoring Interactions
Note The IP Contact Center (IPCC) Extension for the Agents and Supervisors must be unique. This
can be verified from Cisco Unified Communications Manager under Call Routing > Route Plan
Report.
Step 2 Ensure that the agent phone has the Built in Bridge (BIB) On. This can be done on the phone or at the Cluster
level (Set the Default Service Parameter to On).
Step 3 Log in to Finesse as an Agent.
Step 4 Log in to Finesse as a Supervisor and ensure that the supervisor is in NOT READY State.
Step 5 Ensure that the Resource Manager Contact Manager (RMCM) user has the required roles for Call Monitoring
and Call Recording -- Standard Computer Telephony Integration (CTI) Allow Call Monitoring and Recording.
Note This is automatically done by Unified Contact Center Express at the initial setup of the RMCM
user. Ensure the roles exist on the Application User window of Cisco Unified Communications
Manager.
Step 6 Assign the Monitoring CSS (Calling Search Space) on the Supervisor Phone to contain the Partition of the
agent line.
Step 7 Place a call to Unified Contact Center Express so that the call is routed to the agent logged in. Once the agent
is in the TALKING state, from the supervisor, start the Silent Monitoring. The supervisor will then be able
to hear the conversation between the agent and the caller
Silent Monitoring Interactions

Feature Interaction
Call preservation If the agent call that is being monitored goes to call preservation, Unified
Communications Manager also puts the monitoring call into call preservation
mode.
Transfer of secure Unified Communications Manager supports transferring a secure monitoring

monitoring call session so long as the destination supervisor device exceeds the security capabilities
of the agent that is being monitored.
Recording Tones Recording Tones take precedence over Monitoring Tones for calls that are both
recorded and monitored. If a call is recorded and monitored, only the recording
tone plays.
Secure Tones If Secure Tones are configured and the call is secured, the secure tone plays to
both call participants at the outset of the call irrespective of whether Monitoring
Tones are configured.
If Secure Tones and Monitoring Tones are both configured, the secure tone plays
once, followed by the monitoring tones.
If Secure Tones, Monitoring Tones, and Recording Tones are all configured, and
the call is recorded and monitored, the secure tone plays once followed by the
recording tone. The monitoring tone does not play.
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Silent Monitoring Restrictions
Silent Monitoring Restrictions

Feature Restriction
Barge Unified Communications Manager does not support barge with silent
monitoring. If an agent call is being monitored, the barge-in call from a
shared line fails. If the agent call has already been barged, the monitoring
call fails.
Transfer of Secure Silent Unified Communications Manager does not support transferring Secure
Monitoring over an intercluster Silent Monitoring calls over an intercluster trunk.
trunk
Silent Monitoring Restriction The monitoring fails if the supervisor logs in through the non-secure mode
and the Agent logs in to the MRA mode.
For more information, see the Secure Silent Monitoringsection.
126
CHAPTER 12
Recording
• Recording Overview, on page 127
• Recording Prerequisites, on page 130
• Recording Configuration Task Flow, on page 131
• Recording Call Flow Examples, on page 140
• Recording Interactions and Restrictions, on page 140
Recording Overview
Call recording is a Unified Communications Manager feature that enables a recording server to archive agent
conversations. Call recording is one of the essential features in call centers, financial institutions and other
enterprises. The call recording feature sends copies of the agent and the end-user media streams to the recording
server over a SIP trunk. Each media stream is sent separately in an effort to best support a wide range of voice
analytic applications.
Unified Communications Manager offers IP phone-based or network-based recording.
• In IP phone based recording, recording media is sourced from the phone. The phone forks two media
streams to the recording server.
• In network-based recording, recording media can be sourced from either the phone or the gateway. When
you implement network-based recording, the gateway in your network must connect to Unified
Communications Manager over a SIP trunk.
Unified Communications Manager supports call recording in both single cluster and multi-cluster environments
and offers three different recording modes:
• Automatic Silent Recording—Automatic silent recording records all calls on a line appearance
automatically. Unified Communications Manager invokes the recording session automatically with no
visual indication on the phone that an active recording session is established.
• Selective Silent Recording—A supervisor can start or stop the recording session via CTI-enabled
desktop. Alternatively, a recording server can invoke the session based on predefined business rules and
events. There is no visual indication on the phone that an active recording session is established.
• Selective User Call Recording—An agent can choose which calls to record. The agent invokes the
recording session through CTI-enabled desktop, or by a softkey or programmable line key. When selective
user call recording is used, the Cisco IP phone displays recording session status messages.
127
Multi-Fork Recording
Unified Communications Manager supports recording to a single recording server and can be deployed with
CUBE as media proxy to record to more than one recording server.
• In multi-fork recording, Unified Communications Manager is connected to CUBE Media Proxy server
through SIP trunk. The CUBE Media Proxy server receives two media streams from phone or gateway
and it forks the media streams to one or more recording servers simultaneously.
• In recording to single recording servers, Unified Communications Manager is directly connected to
recording server through SIP trunk. The phone or the gateway forks two media streams to the recording
server.
Multi-Fork Recording
Unified Communications Manager supports simultaneous, multiple stream recordings through Cisco Unified
Border Element (CUBE) as Media Proxy. In Multi-fork recording, the recording stream is sent to a CUBE
Media Proxy server, which relays the media stream to up to five recording servers simultaneously. This is
supported for both phone-based and network-based recording, and for both automatic and selective recording.
The multi-forking feature provides the following benefits:
• Adds redundancy and failover to your recording deployment.
• Provides additional media streams for speech analysis and monitoring.
• Helps organization, such as financial industry, to be compliant to MiFID requirements, that mandate
recording of customer calls to multiple recording servers for redundancy.
When you implement multi-fork recording, you must configure the CUBE Media Proxy server in your network
which connects to Unified Communications Manager over a SIP trunk.
For more information on CUBE Media Proxy, see Cisco Unified Border Element Configuration Guide.
Note Connection from Unified Communications Manager to CUBE Media Proxy server over a SIP trunk must be
configured with Early Offer.
The following example illustrates the phone-based recording of multi-fork recording through CUBE Media
Proxy.
Figure 4: Phone-based recording
128
Recording Media Source Selection
The following example illustrates the network-based recording of multi-fork recording through CUBE Media
Proxy.
Figure 5: Network-based recording
For more information on method summary, see "Cisco Device-Specific Extensions" section of Cisco Unified
JTAPI Developers Guide for Cisco Unified Communications Manager Release 12.5(1).
Supported Platforms
Multi-fork recording through CUBE Media Proxy server is supported on the following Cisco Router Platforms
running on Cisco IOS XE Gibraltar Release 16.10.1:
• Cisco 4000 Series-Integrated Services Routers (ISR G3 - ISR4331, ISR4351, ISR4431, ISR4451).
• Cisco Aggregated Services Routers (ASR - ASR1001-X, ASR1002-X, ASR1004 with RP2, ASR1006
with RP2).
• Cisco Cloud Services Routers (CSR1000V series).
Restrictions for Multi-fork recording through CUBE Media Proxy

Multi-fork recording through CUBE Media Proxy server does not support the following:
• Video recording.
• Secure media (SRTP) forking of non secure calls.
• SRTP fall back.
• Midcall block.
Recording Media Source Selection

When you configure network-based recording, you must configure either the phone or the gateway as your
preferred source of recording media for the agent phone line. However, depending on your deployment,
Unified Communications Manager may not select your preferred choice as the recording media source. The
following table displays the logic Unified Communications Manager uses to select the recording media source.
129
Recording Prerequisites
Table 17: Recording Media Source Selection
Preferred Media Source Media Type Gateway in call path? Selected Media Source
Gateway Unsecure (RTP) Yes Gateway
No Phone
Secure (sRTP) Yes Phone
No Phone
Phone Unsecure (RTP) Yes Phone
No Phone
Secure (sRTP) Yes Phone
No Phone
Alternate Recording Media Source if the First Choice is Unavailable

If the recording media source that Unified Communications Manager selects is unavailable, Unified
Communications Manager attempts to use an alternate source. The following table shows the logic Unified
Communications Manager uses to select an alternate source for recording media.
Table 18: Alternate Recording Media Source if First Choice is Unavailable
Selected Media Source Gateway Preferred Phone Preferred
First attempt First gateway in call path Phone
Second attempt Last gateway in call path First gateway in call path
Third attempt Phone Last gateway in call path
Recording Prerequisites
• Cisco Unified IP Phone support—To view a list of the Cisco Unified IP Phone that support recording,
log in to Cisco Unified Reporting and run the Unified CM Phone Feature List report, selecting Record
as the feature. For a detailed procedure, see Generate a Phone Feature List, on page 1.
• Gateway support—For details on which gateways support recording, see https://developer.cisco.com/
web/sip/wiki/-/wiki/Main/Unified+CM+Recording+Gateway+Requirements.
• If you are configuring multiple-stream recording, deploy and configure a CUBE Media Proxy. For details,
see the section CUBE Media Proxy in the Cisco Unified Border Element Configuration Guide.
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Recording Configuration Task Flow
Recording Configuration Task Flow

Before you begin
Procedure

Step 1 Create a Recording Profile, on page 131 Create a recording profile.
Step 2 Configure SIP Profile for Recording, on page Optional. Configure the SIP Profile if you want
132 to deliver the Conference Bridge Identifier to
the recorder.
Step 3 Configure SIP Trunks for Recording, on page Configure the recorder server or CUBE Media
132 Proxy as a SIP trunk device.
Step 4 Configure Route Pattern for Recording, on page Create a route pattern that routes to the recorder
133 server or CUBE Media Proxy.
Step 5 Configure Agent Phone Line for Recording, on Configure the agent phone line for recording.
page 133
Step 6 Enable the built in bridge for your agent phones. To use the agent phone as the recording media
Perform one of the following tasks to enable source you must enable the phone's built in
the built-in-bridge for recording: bridge for recording. You can use a service
parameter to set the built in bridge defaults
• Enable Built in Bridge for Cluster , on
across the cluster, or enable the built in bridge
page 134
on an individual phone.
• Enable Built in Bridge for a Phone, on
page 134 Note The Built in Bridge setting on
individual phones overrides the
clusterwide defaults.
Step 7 Enable Gateway for Recording, on page 135 Configure Unified Communications services
on the gateway.
Step 8 Configure Recording Notification Tones, on Configure whether you want a notification tone
page 135 to play when calls are recorded.
Step 9 Perform one of the following procedures, Configure a Record feature button or softkey
depending on whether your phone uses feature for your phone.
buttons or softkeys:
• Configure a Record Feature Button, on
page 136
• Configure a Record Softkey, on page 137
Create a Recording Profile

Use this procedure to create a recording profile.
131
Configure SIP Profile for Recording
Procedure
Step 1 From Cisco Unified CM Administration, choose Device > Device Settings > Recording Profile.
Step 3 In the Name field, enter a name for your recording profile.
Step 4 In the Recording Calling Search Space field, select the calling search space that contains the partition with
the route pattern that is configured for the recording server.
Step 5 In the Recording Destination Address field, enter the directory number or the URL of the recording server
or the URL of the CUBE Media Proxy server.
Step 6 Click Save.
Configure SIP Profile for Recording

Use this procedure to deliver the conference bridge identifier to the recorder and configure the SIP Profile.
Procedure
Step 1 From Cisco Unified CM Administration, choose Device > Device Settings > SIP Profile.
Step 2 Select the SIP profile that you want to use for your network.
Step 3 Set a value for the Early Offer Support for Voice and Video calls field. SIP trunk from Unified
Communications Manager to CUBE Media Proxy server must be enabled for an Early Offer support and
configuration options are Best Effort (no MTP inserted) and Mandatory (insert MTP if needed).
Note We recommend that you enable SIP trunk for Mandatory (insert MTP if needed).
Step 4 Check the Deliver Conference Bridge Identifier check box.

Step 5 Click Save.
Configure SIP Trunks for Recording

Use this procedure to assign the recording server information in the SIP Trunk Configuration window.
Procedure
Step 1 From Cisco Unified CM Administration, choose Device > Trunk.

Step 3 From the Trunk Type drop-down list, choose SIP Trunk.
Device Protocol is auto-populated to SIP, which is the only available option.
Step 4 From the Trunk Service Type drop-down list, choose the service type that you want to use in your network.
The default value is None.
132
Configure Route Pattern for Recording
Step 5 Click Next.

Step 6 In the Destination Address field of the SIP Information pane, enter an IP address, fully qualified domain
name, or DNS SRV of the recording server or CUBE Media proxy.
Step 7 From the SIP Profile drop-down list in the SIP Information pane, choose the SIP profile that you want to
use in your network.
Step 8 From the Recording Information pane, select one of the following options:
• None—This trunk is not used for recording.
• This trunk connects to a recording-enabled gateway.
• This trunk connects to other clusters with recording-enabled gateways.
Step 9 Click Save.

Note SIP trunk from Unified Communications Manager to Media Proxy must be enabled for Early
Offer support in the SIP Profile that is used for this trunk. The configuration options are
Mandatory (insert MTP if needed) and Best Effort (no MTP inserted).
Configure Route Pattern for Recording

Use this procedure to describe the route pattern configurations that are specific to recorders. You must configure
a route pattern that routes to the recording server or CUBE Media Proxy server.
Procedure
Step 1 From Cisco Unified CM Administration, choose Call Routing > Route/Hunt > Route Pattern.
Step 2 Click Add New to create a new route pattern.
Step 3 Complete the fields in the Route Pattern Configuration window.For more information on the fields and
Step 4 For call recording, complete the following fields:
• Pattern—Enter a pattern that matches the recording destination address from the recording profile.
• Gateway/Route List—Choose the SIP trunk or route list that points to the recording server.
Step 5 Click Save.
Configure Agent Phone Line for Recording

Use this procedure to configure the agent phone line for recording.
Procedure

Step 2 Click Find.
133
Enable Built in Bridge for Cluster
Step 3 Select the agent's phone.

Step 4 In the left Association pane, click the phone line for the agent to view the settings.
Step 5 From the Recording Option drop-down list, choose one of the following options:
• Call Recording Disabled—Calls on this phone line are not recorded.
• Automatic Call Recording Enabled—All calls on this phone line are recorded.
• Selective Call Recording Enabled—Only selected calls on this phone line are recorded.
Step 6 From the Recording Profile drop-down list, choose the recording profile that is configured for the agent.
Step 7 From the Recording Media Source drop-down list, choose whether you want to use the gateway or the phone
as the preferred source of recording media.
Step 8 Set the Busy Trigger field to a minimum of 3 if you also have Multilevel Precedence and Preemption (MLPP)
configured.
Step 9 Click Save.
Enable Built in Bridge for Cluster

Use this procedure to enable the phone's built in bridge for recording to use the agent phone as the recording
media source.
When you set the Built-in-Bridge clusterwide service parameter to enable, the Built-in-Bridge default setting
for all phones in the cluster is changed to enabled. However, the Built-in-Bridge setting in the Phone
Configuration window for an individual phone overrides the clusterwide service parameter setting if the
default option is not selected for that phone.
Procedure
Step 4 Set the Builtin Bridge Enable service parameter to On.
Step 5 Click Save.
Enable Built in Bridge for a Phone

Use this procedure to enable the Built in Bridge for an individual phone. If the default option is not selected,
the Built in Bridge setting in the Phone Configuration window overrides the clusterwide service parameter.
Optionally, use a service parameter to set the Built in Bridge defaults across the cluster. For more information,
see Enable Built in Bridge for Cluster , on page 134.
Procedure
134
Enable Gateway for Recording

Note The recording can fail if the Built-in-Bridge is ON and if you check the Media Termination
Point Required check box.
Step 4 Click Save.
Enable Gateway for Recording

Use this procedure to configure the gateway for recording. You must enable Unified Communications Gateway
Services. The following task flow contains the high-level process to enable Unified Communications Gateway
Services.
Procedure
Step 1 Configure Unified Communications Manager IOS Services on the Device.

Step 2 Configure the XMF Provider.
Step 3 Verify Unified Communications Gateway Services.
For detailed configuration steps, including examples, refer to the Cisco Unified Communications Gateway
Services chapter for either of the following documents:
• For more information, see ASR routers Cisco Unified Border Element (Enterprise) Protocol-Independent
Features and Setup Configuration Guide. Cisco IOS XE Release 35.
• For more information, see ISR routers Cisco Unified Border Element Protocol-Independent Features and
Setup Configuration Guide, Cisco IOS Release 15M&T.
Configure Recording Notification Tones

Use this procedure to configure notification tone to play when calls are recorded. For legal compliance, an
explicit notification in the form of a periodic tone can be made audible to the agent, the caller, or both, to
indicate that a recording session is in progress. This tone can also be disabled.
Note Recording tone settings override monitoring tone settings when both are enabled for the same call.
135
Configure a Record Feature Button
Procedure
Step 2 From the Server drop-down list, choose the server on which the Cisco CallManager service is running.
Step 4 If you want a notification tone to be played to the agent, set the Play Recording Notification Tone to Observed
Target (agent) service parameter to True.
Step 5 If you want a notification tone to be played to the customer, set the value of the Play Recording Notification
Tone To Observed Connected Parties (customer) service parameter to True.
Step 6 Click Save.
Configure a Record Feature Button

Use this procedure to assign the Record feature button to your phone if your phone uses feature buttons.
Procedure

Step 1 Configure a Phone Button Template for Configure a phone button template that includes
Recording, on page 136 the Record button.
Step 2 Associate a Phone Button Template with a Associate the phone button template that you
Phone, on page 137 set up for recording to the phone.
Configure a Phone Button Template for Recording

Use this procedure to create a phone button template that includes the Record feature button.
Procedure
Step 1 From Cisco Unified CM Administration, choose Device > Device Settings > Phone Button Template.
Step 2 Click Find to display list of supported phone templates.
Step 3 Perform the following steps if you want to create a new phone button template; otherwise, proceed to the next
step.
a) Select a default template for the model of phone and click Copy.
b) In the Phone Button Template Information field, enter a new name for the template.
c) Click Save.
Step 4 Perform the following steps if you want to add phone buttons to an existing template.
a) Click Find and enter the search criteria.
Step 5 From the Line drop-down list, choose feature that you want to add to the template.
Step 6 Click Save.
136
Associate a Phone Button Template with a Phone

• Click Apply Config if you modified a template that is already associated with devices to restart the
devices.
• If you created a new softkey template, associate the template with the devices and then restart them.
Associate a Phone Button Template with a Phone

Use this procedure to associate the phone button template that you created for the Record button of the phone.
Procedure

Step 2 Click Find to display the list of configured phones.
Step 3 Choose the phone to which you want to add the phone button template.
Step 4 In the Phone Button Template drop-down list, choose the phone button template that contains the new feature
button.
Step 5 Click Save.
A dialog box is displayed with a message to press Reset to update the phone settings.
Configure a Record Softkey

Use this procedure to add a Record softkey to the phone, if your phone uses softkeys. The Record softkey is
only available in the Connected call state for the Cisco Chaperone Phone with Feature Hardkeys template.
Procedure

Step 1 Configure a Softkey Template for Recording, Configure a softkey template that includes the
on page 138 Record softkey.
Step 2 Perform one of the following procedure: Associate the softkey template to a phone
directly, or to a Common Device Configuration.
• Associate a Softkey Template with a
You can then associate the Common Device
Phone, on page 138
Configuration to a group of phones.
Common Device Configuration, on page
139
137
Configure a Softkey Template for Recording
Configure a Softkey Template for Recording
Procedure
Step 2 Perform the following steps to create a new softkey template; otherwise, proceed to the next step.
a) Click Add New.
c) Enter a new name for the template in the Softkey Template Name field.
d) Click Save.
Step 3 Perform the following steps to add softkeys to an existing template.
b) Select the required existing template.
Step 4 Check the Default Softkey Template check box to designate this softkey template as the default softkey
template.
Step 5 Choose Configure Softkey Layout from the Related Links drop-down list in the upper right corner and
click Go.
Step 6 From the Select a Call State to Configure drop-down list, choose the call state for which you want the softkey
to display.
Step 7 From the Unselected Softkeys list, choose the softkey to add and click the right arrow to move the softkey
to the Selected Softkeys list. Use the up and down arrows to change the position of the new softkey.
Step 8 Repeat the previous step to display the softkey in additional call states.
Step 9 Click Save.
devices.
• If you created a new softkey template, associate the template with the devices and then restart them. For
more information, see Add a Softkey Template to a Common Device Configuration and Associate a
Softkey Template with a Phone sections.
Associate a Softkey Template with a Phone

Use this procedure to assign the Record softkey to the phone by associating the softkey template that includes
the Record softkey directly to a phone.
Procedure
138
Associate a Softkey Template with a Common Device Configuration
Step 2 Click Find to select the phone to add the softkey template.
Step 3 From the Softkey Template drop-down list, choose the template that contains the new softkey.
Step 4 Click Save.
Step 5 Press Reset to update the phone settings.

Use this procedure to add a Record softkey to the phone by associating the softkey template to a Common
Device Configuration.
Procedure

Step 1 Add a Softkey Template to the Common Device
Configuration, on page 139
Step 2 Add Common Device Configuration to Phone,
on page 140
Add a Softkey Template to the Common Device Configuration
Procedure
Step 1 From Cisco Unified CM Administration, choose Device > Device Settings > Common Device Configuration.
Step 2 Perform the following steps to create a new Common Device Configuration and associate the softkey template
with it; otherwise, proceed to the next step.
a) Click Add New.
b) Enter a name for the Common Device Configuration in the Name field.
c) Click Save.
Step 3 Perform the following steps to add the softkey template to an existing Common Device Configuration.
b) Click an existing Common Device Configuration.
Step 4 In the Softkey Template drop-down list, choose the softkey template that contains the softkey that you want
to make available.
Step 5 Click Save.
• If you modified a Common Device Configuration that is already associated with devices, click Apply
Config to restart the devices.
• If you created a new Common Device Configuration, associate the configuration with devices and then
restart them.
139
Add Common Device Configuration to Phone
Add Common Device Configuration to Phone
Procedure

Step 2 Click Find and select the phone device to add the softkey template.
Step 3 From the Common Device Configuration drop-down list, choose the common device configuration that
contains the new softkey template.
Step 4 Click Save.
Step 5 Click Reset to update the phone settings.
Recording Call Flow Examples

For call flow examples for both network-based call recording and IP phone-based call recording use cases,
refer to Call Recording Examples for Network-Based and Phone-Based Recording at the following URL:
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/configExamples/cucm_b_recording-use-cases.html
Recording Interactions and Restrictions

Feature Interactions and Restrictions
Monitoring Tones Recording Tones take precedence over Monitoring Tones for calls that are both
recorded and monitored. If both are configured, and a call is both recorded and
monitored, only the recording tone plays.
Multilevel Precedence and If you also have Multilevel Precedence and Preemption (MLPP) configured, the
Preemption Busy Trigger setting on the agent phone line that you are recording must be set
to a minimum of 3.
Secure Tones If Secure Tones are configured, the secure tone plays to both call participants at
the outset of a secure call, irrespective of whether Recording Tones are configured.
If Secure Tones and Recording Tones are both configured and the call is secure,
the secure tone plays once at the outset of the call followed by the recording tone.
If Secure Tones, Recording Tones, and Monitoring Tones are all configured, and
the call is secured, recorded, and monitored, the secure tone plays once followed
by the recording tone. The monitoring tone does not play.
Customer Voice Portal Agent - customer calls that are routed through the Customer Voice Portal may
be recorded using the agent phone as the recording source.
SIP Proxy Servers If you are using the gateway as your recording source, you cannot place SIP
proxy servers between Unified Communications Manager and the gateway.
140
Busy Hour Call Each recording session adds two calls to the Busy Hour Call Completion (BHCC)
Completion Rate rate with a minimal impact on CTI resources.
Selective Recording with When Selective Recording is configured, the Media Sense server does not record
Media Sense the consult call during a transfer. For example, if a call between an agent and a
customer is being recorded and the agent initiates a transfer to a second agent,
the consult call that takes place between the two agents, prior to the call being
transferred, is not recorded.
To ensure that the consult call is recorded, the agent must press the ‘Record’
softkey when the consult call starts.
Recording on To record a call for authenticated phones, On the Cisco Unified CM Service
authenticated phones Parameter page, set the Authenticated Phone Recording field to Allow
Recording. The default value is Do Not Allow Recording. Unified
Communications Manager allows call recording for authenticated phones while
using non secure recorder. In case of secure recorder, recording is allowed only
if the recorder supports Secure Real-Time Transport protocol (SRTP) fallback.
Codec locking for auto Skinny Client Control Protocol (SCCP) phone adversities one single codec when
recording calls in select recording is enabled and there is a select and join conference performed in Unified
and join conference Communications Manager.
Recording Calls Do Not Survive Agent Hold

Recording calls get torn down when the agent puts the call on hold, and they get reestablished when the agent
resumes the call.
Figure 6: Recording Calls Do Not Survive Agent Hold
141
142
PA R T VI
Call Center Features
• Agent Greeting , on page 145
• Auto-Attendant , on page 149
• Manager Assistant , on page 157
CHAPTER 13
Agent Greeting
• Agent Greeting Overview, on page 145
• Agent Greeting Prerequisites, on page 145
• Agent Greeting Configuration Task Flow, on page 145
• Agent Greeting Troubleshooting, on page 147
Agent Greeting Overview

Agent Greeting enables Unified Communications Manager to automatically play a prerecorded announcement
following a successful media connection to the agent device. Agent Greeting is audible for the agent and the
customer.
The process of recording a greeting is similar to recording a message for voicemail. Depending on how your
contact center is set up, you can record different greetings that play for different types of callers (for example,
an English greeting for English speakers or an Italian greeting for Italian speakers).
By default, agent greeting is enabled when you log in to your agent desktop but you can turn it off and on as
necessary.
Agent Greeting Prerequisites

• Install Cisco Unified Contact Center Enterprise. See Cisco Unified Contact Center Enterprise Installation
and Upgrade Guide.
• Install Cisco Unified Customer Voice Portal. See Installation and Upgrade Guide for Cisco Unified
Customer Voice Portal .
• Ensure that you enable Built In Bridge. To view the details, see Configure Built In Bridge, on page 147.
Agent Greeting Configuration Task Flow

Agent Greeting configuration tasks are completed in Cisco Unified Contact Center Enterprise (Unified CCE)
and Cisco Unified Customer Voice Portal (Unified CVP). To view detailed steps for the following tasks, see
the Agent Greeting section in the Cisco Unified Contact Center Enterprise Features Guide.
145
Agent Greeting Configuration Task Flow
Before you begin

• Review Agent Greeting Prerequisites, on page 145
Procedure

Step 1 Configure a media server for Agent Greeting. Agent Greeting uses the Unified CVP media
server to store and serve prompt and greeting
• Configure a server to act as a media
files.
server.
• Add the media server in Unified CVP.
• Configure the media server to write files.
Step 2 Republish .tcl scripts to Voice Extensible The .tcl script files that ship with Unified CVP
Markup Language (VXML) Gateway. Release 9.0(1) include updates to support
Agent Greeting. You must republish these
updated files to your VXML Gateway.
Republishing scripts to the VXML Gateways
is a standard task in Unified CVP upgrades. If
you did not upgrade Unified CVP and
republish the scripts, you must republish the
scripts before you can use Agent Greeting.
Step 3 Set the cache size on the VXML Gateway. To ensure adequate performance, set the size
of the cache on the VXML Gateway to the
maximum allowed. The maximum size is 100
megabytes; the default is 15 kilobytes. Failure
to set the VXML Gateway cache to its
maximum can result in slowed performance
to increased traffic to the media server.
Step 4 Create voice prompts to record greetings. Create audio files for each of the voice prompts
that agents hear as they record a greeting.
Step 5 Configure call types. Complete to record and play agent greetings.
Step 6 Configure a dialed number. Complete to record and play agent greetings.
Step 7 Schedule the script.

Step 8 Define network VRU scripts. For Agent Greeting record and play scripts to
interact with Unified CVP, Network VRU
scripts are required.
Step 9 (Optional) Import sample Agent Greeting
scripts.
Step 10 Modify the Unified CCE call routing scripts. Modify the Unified CCE call routing scripts
to use the Play Agent Greeting script.
146
Configure Built In Bridge
Configure Built In Bridge

The Built in Bridge field setting in the Phone Configuration window for an individual phone overrides the
setting for the Builtin Bridge Enable clusterwide service parameter.
Procedure

Step 4 Click Save.
Agent Greeting Troubleshooting

For information about how to troubleshoot Agent Greeting issues, see “Troubleshooting Agent Greeting”
chapter in the Agent Greeting and Whisper Announcement Feature Guide for Cisco Unified Contact Center
Enterprise guide.
147
Agent Greeting Troubleshooting
148
CHAPTER 14
Auto-Attendant
• Auto-Attendant Overview, on page 149
• Cisco Unity Connection Configuration, on page 150
• Cisco Unified CCX Configuration, on page 154
• Cisco Unity Express Configuration, on page 156
Auto-Attendant Overview
Auto-Attendant allows callers to locate people in your organization without talking to a receptionist. You can
customize the prompts that are played for the caller.
Auto-Attendant works with Unified Communications Manager to receive calls on specific telephone extensions.
The software interacts with the caller and allows the caller to search for and select the extension of the party
(in your organization) that the caller is trying to reach.
Auto-Attendant provides the following functions:
• Answers a call
• Plays a user-configurable welcome prompt
• Plays a main menu prompt that asks the caller to perform one of three actions:
• Press 0 for the operator
• Press 1 to enter an extension number
• Press 2 to spell by name
If the caller chooses to spell by name (by pressing 2), the system compares the letters that are entered
with the names that are configured to the available extensions. One of the following results can
occur:
• If a match exists, the system announces a transfer to the matched user and waits for up to 2
seconds for the caller to press any Dual Tone Multifrequency (DTMF) key to stop the transfer.
If the caller does not stop the transfer, the system performs an explicit confirmation: it prompts
the user for confirmation of the name and transfers the call to the primary extension of that
user.
• If more than one match occurs, the system prompts the caller to choose the correct extension.
• If too many matches occur, the system prompts the caller to enter more characters.
149
Cisco Unity Connection Configuration
• If no match occurs, that is, if the user presses wrong options, the system prompts that the user
pressed the wrong options and prompts the user to press the correct options.
• When the caller specifies the destination, the system transfers the call.
• If the line is busy or not in service, the system informs the caller accordingly and replays the main menu
prompt.
Auto-Attendant solution can be deployed in three different ways as follows using different Cisco products
that can provide interactive voice response functionality.
• Auto-Attendant using Cisco Unity Connection (CUC); the most widely used Auto-Attendant solution
configuration by customers
• Auto-Attendant using Cisco Unified Contact Center Express (Unified CCX)
• Auto-Attendant using Cisco Unity Express (CUE)
Cisco Unity Connection Configuration

The Cisco Unity Connection server provides Automated-Attendant functionality for both external and internal
callers. An Auto-Attendant allows callers to be automatically transferred to an extension without the intervention
of an operator or receptionist.
Auto-Attendants offer a menu system; it may also allow a caller to reach a live operator by dialing a number,
usually “0”. Multiple Auto-Attendants may be implemented to support individual site locations. Within Cisco
Unity Connection, an Auto-Attendant is a customized application tree structure that is built by creating and
linking multiple Call Handlers together. The Auto-Attendant is defined by entry and exit points, and intermediate
routing decisions based on the callers DTMF input choices.
For more information about Auto-Attendant default behavior and examples, see System Administration Guide
for Cisco Unity Connection.
Cisco Unity Connection Configuration Task Flow

You can use this task flow to configure auto-attendant using Cisco Unity Connection:
Procedure

Step 1 Configure CTI Route Point, on page 151 Perform this task on the Cisco Unified CM
Administration. Create a CTI Route Point which
maps to the Direct-Inward Dial (DID) number
of the company (board number).
Step 2 Configure Auto-Attendant System Call Handler, Call handlers answer calls, greet callers with
on page 152 recorded prompts, provide callers with
information and options, route calls, and take
messages.
150
Configure CTI Route Point

Note You can customize the greeting
for the AutoAttendant Call
Handler by choosing Edit >
Greetings. For more information
about customizing greetings, see
System Administration Guide for
Cisco Unity Connection.
Step 3 Configure Caller Input Option, on page 152 Caller input option enables you to designate a
single digit to represent a user extension,
alternate contact number, call handler, interview
handler, or directory handler. The caller presses
a single key during a call handler greeting
instead of entering the full extension, and Cisco
Unity Connection responds accordingly. Several
different keys configured as caller input options
offers the callers a menu of choices in the call
handler greeting.
Step 4 Configure Extension for Operator Call Handler, Configure an extension for the operator to allow
on page 153 callers to speak to an operator during a call
handler greeting.
Step 5 Modify Standard Call Transfer Rule for Modify the Standard Call Transfer Rule to
Operator, on page 153 enable the call to be transferred to the operator
when the caller presses 0 to speak to an
operator.
Step 6 Update Default System Transfer Restriction Update the Default System Transfer restriction
Table, on page 153 table. The Default System Transfer restriction
table restricts numbers that can be used for
Caller system transfers, which allow
unidentified callers to transfer to a number that
they specify.
Procedure
Step 1 From Cisco Unified CM Administration, choose Device > CTI Route Point.
Step 3 In the Device Name field, enter a device name for the route point.
Step 4 From the Device Pool drop-down list, choose Default.
Step 5 Click Save.
The Add successful message is displayed.
Step 6 From the Association area, click Line [1] - Add a new DN.
The Directory Number Configuration window is displayed.
Step 7 In the Directory Number field, enter the directory number that matches with the DID of the company.
151
Configure Auto-Attendant System Call Handler
Step 8 From the Route Partition drop-down list, choose the required route partition.
Step 9 From the Call Forward and Call Pickup Settings area, for Forward All, choose the appropriate calling
search space and check the Voice Mail check box.
Step 10 Click Save.
Configure Auto-Attendant System Call Handler
Procedure
Step 1 From Cisco Unity Connection Administration, from the Cisco Unity Connection tree on the left, navigate to
Call Management and choose System Call Handlers.
The New Call Handler window is displayed.
Step 3 In the Display Name field, enter AutoAttendant.
Step 4 In the Extension field, enter the same extension that you provided for the CTI Route Point.
Step 5 Click Save.
The Edit Call Handler Basics (AutoAttendant) window is displayed.
Step 6 Edit the required fields and click Save.
Configure Caller Input Option
Procedure
Step 2 Click AutoAttendant.
The Edit Call Handler Basics (AutoAttendant) window is displayed.
Step 3 Choose Edit > Caller Inputs.
The Caller Input window is displayed.
Step 4 In the Key column, click 0.
The Edit Caller Input (0) window is displayed.
Step 5 Click the Call Handler radio button, choose Operator from the drop-down list, and click the Attempt
Transfer radio button.
Step 6 Click Save.
The Updated Caller Input status message is displayed.
Step 7 Choose Edit > Caller Inputs.
The Caller Input window is displayed.
Step 8 In the Key column, click 1.
The Edit Caller Input (0) window is displayed.
Step 9 In the Conversation radio button, choose Caller System Transfer from the drop-down list.
Step 10 Click Save.
152
Configure Extension for Operator Call Handler
Configure Extension for Operator Call Handler
Procedure
Step 2 Click Operator.
The Edit Call Handler Basics (Operator) window is displayed.
Step 3 Enter the extension of the operator in the Extension field and click Save.
Modify Standard Call Transfer Rule for Operator
Procedure
Step 2 Click Operator.
The Edit Call Handler Basics (Operator) window is displayed.
Step 3 From the Edit menu, choose Transfer Rules.
The Transfer Rules window is displayed.
Step 4 Click Standard.
The Edit Transfer Rule (Standard) window is displayed.
Step 5 In the Transfer Calls to option, click the Extension radio button and enter the configured operator extension
number.
Step 6 Click Save.
Update Default System Transfer Restriction Table
Procedure
System Settings and choose Restriction Tables.
Step 2 Click Default System Transfer.
The Edit Restriction Table Basics (Default System Transfer) window is displayed.
Step 3 Uncheck the check box in the Blocked column for 6 in the Order column.
153
Cisco Unity Connection Auto-Attendant Troubleshooting
Step 4 Click Save.
Cisco Unity Connection Auto-Attendant Troubleshooting

For information about troubleshooting Auto-Attendant using Cisco Unity Connection, see the following:
• http://www.cisco.com/c/en/us/support/docs/voice-unified-communications/
unified-communications-manager-callmanager/107517-calltrf.html
• http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/connection/8x/troubleshooting/guide/8xcuctsgx/
8xcuctsg110.html
8xcuctsg040.html
8xcuctsg180.html
Cisco Unified CCX Configuration

Auto-Attendant comes standard with the five-seat bundle of Cisco Unified Contact Center Express (Unified
CCX).
Note For information about the supported versions of Cisco Unified CCX with Unified Communications Manager,
see Cisco Collaboration Systems Release Summary Matrix for IP Telephony.
For information about getting started with scripts, see the Cisco Unified Contact Center Express Getting
Started with Scripts.
Cisco Unified CCX Prerequisites

• Install and configure Cisco Unified CCX before you can use Auto-Attendant. Cisco Unified CCX controls
the software and its connection to the telephony system.
• Configure users on Unified Communications Manager.
Cisco Unified CCX Auto-Attendant Task Flow

Auto-Attendant configuration tasks are completed in Cisco Unified Contact Center Express (Unified CCX).
To view detailed steps for the following tasks, see Cisco Unified CCX Administration Guide and the Cisco
Unified Contact Center Express Getting Started with Scripts respectively.
Before you begin

• Learn more about the Auto-Attendant feature by reviewing Auto-Attendant Overview, on page 149.
154
Cisco Unified CCX Auto-Attendant Task Flow
• Learn more about Cisco UCCX with Auto-Attendant functionality by reviewing Cisco Unified CCX
• Review Cisco Unified CCX Prerequisites, on page 154.
Procedure

Step 1 Configure Unified CM Telephony call control The Unified CCX system uses Unified CM
groups. Telephony call control groups to pool together
a series of CTI ports, which the system uses to
serve calls as they arrive or depart from the
Unified CCX server.
Step 2 Add a Cisco Media Termination (CMT) dialog The Cisco Media subsystem is a subsystem of
control group. the Unified CCX Engine. The Cisco Media
subsystem manages the CMT media resource.
CMT channels are required for Unified CCX
to be able to play or record media.
The Cisco Media subsystem uses dialog groups
to organize and share resources among
applications. A dialog group is a pool of dialog
channels in which each channel is used to
perform dialog interactions with a caller, during
which the caller responds to automated prompts
by pressing buttons on a touch-tone phone.
Caution All media termination strings
begin with “auto” and contain the
same ID as the call control
group—not the CMT dialog
group. Perform this procedure if
the default media termination is
configured and the ID differs.
Step 3 Configure a Cisco script application. The Unified CCX script applications are
applications that are based on scripts created in
the Unified CCX Editor. These applications
come with every Unified CCX system and
executes scripts that are created in the Unified
CCX Editor.
Step 4 Provision a Unified CM Telephony trigger. A Unified CM Telephony trigger responds to
calls that arrive on a specific route point by
selecting telephony and media resources to
serve the call and invoking an application script
to handle the call.
Step 5 Customize Auto-Attendant. The Cisco Unified CCX Administration page

allows you to modify any existing
• Modify an existing Auto-Attendant
Auto-Attendant instance as necessary.
instance
155
Cisco Unity Express Configuration

• Configure the Auto-Attendant prompts Cisco Unified CCX allows you to customize
the Auto-Attendant prompts from the Cisco
Unified CCX Administration Media
Configuration window. It allows you to record
the welcome prompt, configure the welcome
prompt, and upload a spoken name.
Cisco Unity Express Configuration

For information about Auto-Attendant Configuration using Cisco Unity Express, see the “Configuring Auto
Attendants” chapter in Cisco Unity Express VoiceMail and Auto Attendant CLI Administrator Guide for 3.0
and Later Versions.
For information about deploying a sample Auto-Attendant script, see “Deployment of sample script aa.aef”
chapter in the Getting Started with Cisco Unified IP IVR.
For information about an Auto-Attendant example, see “Auto Attendant Script Example” chapter in the Cisco
Unity Express Guide to Writing and Editing Scripts for 7.0 and Later Versions.
For information about Auto-Attendant design considerations, see “Auto Attendant Design Considerations”
chapter in the Cisco Unity Express Design Guide.
Cisco Unity Express Auto-Attendant Troubleshooting

For information about Auto-Attendant troubleshooting using Cisco Unity Connection, see the “Troubleshooting
Cisco Unity Express Automated Attendant” in Excerpts from Cisco IP Communications Express: CallManager
Express with Cisco Unity Express.
156
CHAPTER 15
Manager Assistant
• Cisco Unified Communications Manager Assistant Overview, on page 157
• Manager Assistant Prerequisites, on page 159
• Manager Assistant Task Flow for Proxy Lines, on page 160
• Manager Assistant Task Flow for Shared Lines, on page 168
• Manager Assistant Interactions, on page 186
• Manager Assistant Restrictions, on page 188
• Cisco Unified Communications Manager Assistant Troubleshooting , on page 189
Cisco Unified Communications Manager Assistant Overview

The Unified Communications Manager Assistant feature is a plug-in that an assistant can use to handle calls
on behalf of a manager, intercept manager calls, and route them appropriately.
Manager Assistant supports up to 3500 managers and 3500 assistants. To accommodate this number of users,
you can configure up to three Manager Assistant applications in one Unified Communications Manager cluster
and assign managers and assistants to each instance of the application.
Manager Assistant supports shared line support and proxy line support.
Manager Assistant Architecture

The Manager Assistant architecture comprises the following:
• Cisco IP Manager Assistant service—After you install Unified Communications Manager, activate
this service from the Cisco Unified Serviceability interface.
• Assistant Console interface—Allows assistants to access the Manager Assistant features on their
computer to handle calls for managers. The Manager Assistant handles calls for an assistant and for as
many as 33 managers.
• Cisco Unified IP Phone interface: Managers and assistants use softkeys and the Cisco Unified IP Phone
Services button to access the Manager Assistant features.
For more information, see chapter Manager Assistant, in Feature Configuration Guide for Cisco Unified
157
Manager Assistant Shared Line Overview
Manager Assistant Database Access Architecture

The database stores all Manager Assistant configuration information. When the manager or assistant logs in,
the Cisco IP Manager Assistant service retrieves all data that is related to the manager or assistant from the
database and stores it in memory. The database includes two interfaces:
• Manager interface—The manager phone makes available the manager features except Manager
Configuration. Manager Assistant automatically logs in a manager in to the Cisco IP Manager Assistant
service when the Cisco IP Manager Assistant service starts.
Note Managers also have access to Unified Communications Manager features such
as Do Not Disturb and Immediate Divert.
• Assistant interface—The assistant accesses the Manager Assistant features by using the Assistant
Console application and the Cisco Unified IP Phone. The Assistant Console, an application, provides
call-control functions such as answer, divert, transfer, and hold. The assistant uses the Assistant Console
to log in and log out, to set up assistant preferences, and to display the Manager Configuration window
that is used to configure manager preferences.
For more information, see chapter Manager Assistant, in Feature Configuration Guide for Cisco Unified
Softkeys
Manager Assistant supports the following softkeys:
• Redirect
• Transfer to VoiceMail
• Do Not Disturb
Manager Assistant supports the following softkey templates:

• Standard Manager—Supports manager for proxy mode
• Standard Shared Mode Manager—Supports manager for shared mode
• Standard Assistant—Supports assistant in proxy or shared mode
• Standard User—The system makes call-processing (such as Hold and Dial) softkeys available with the
Standard User template.
Manager Assistant Shared Line Overview

When you configure Manager Assistant in shared line mode, the manager and assistant share a directory
number, for example, 8001. The assistant handles calls for a manager on the shared directory number. When
a manager receives a call on 8001, both the manager phone and the assistant phone ring.
The Manager Assistant features that do not apply to shared line mode include Default Assistant Selection,
Assistant Watch, Call Filtering, and Divert All Calls. An assistant cannot see or access these features on the
Assistant Console application.
158
Manager Assistant Proxy Line Overview
Manager Assistant Proxy Line Overview

When you configure Manager Assistant in proxy line mode, the assistant handles calls for a manager using a
proxy number. The proxy number is not the directory number for the manager, but is an alternate number
chosen by the system that an assistant uses to handle manager calls. In proxy line mode, a manager and an
assistant have access to all features that are available in Manager Assistant, which include Default Assistant
Selection, Assistant Watch, Call Filtering, and Divert All Calls.
Manager Assistant Prerequisites

• The user should install JRE on 32 or 64-bit Windows platform before the user upgrades the Manager
Assistant client to a newer version for Releases 11.5(1)SU9, 12.0(1)SU4, and 14 onwards.
Important Before you perform the upgrade, ensure that you uninstall the Cisco Unified
Communications Manager Assistant client that is currently installed on your
machine. This is applicable from Releases 12.0(1)SU4 and 14 onwards.
• Manager Assistant supports the following browsers and platform:

• Unified Communications Manager Assistant Administration and the Assistant Console are supported
on Internet Explorer 11 with Windows 10 (64 bit), Firefox with Windows 10 (64 bit) or later, and
Safari with MacOS (10.x) or later.
Note To run IPMA plug-in on Windows 11, you should install the IPMA Release 15
version plug-in to any of these supported OS platforms: Windows 10, Windows
2019, and Windows 2022. You must then copy the installed version of the IPMA
plug-in to Windows 11 and then launch IMPA.
• On a computer running Windows 10 or Apple MAC OS X, you can open one of the browsers
specified above.
• To display Manager Assistant features in other languages, install the locale installer before you configure
the Manager Assistant.
• The Assistant Console application is supported on computers that run Windows 10, Windows 2019, and
Windows 2022.
• You must configure the phones and users, and associated the devices to the users. In addition, for shared
line appearances between managers and assistants, you must configure the same directory number on
the manager primary line and assistant secondary line.
• To add managers and assistants in bulk, install the Cisco Unified Communications Manager Bulk
Administration Tool. For more information, see the Bulk Administration Guide.
159
Manager Assistant Task Flow for Proxy Lines
Manager Assistant Task Flow for Proxy Lines

Before you begin
• Review Manager Assistant Prerequisites, on page 159.
Procedure

Step 1 Run the Cisco Unified CM Assistant
Configuration Wizard, on page 160
Step 2 Configure Manager And Assign Assistant For
Proxy Line, on page 166
Step 3 Configure Assistant Line Appearances for
Proxy Line, on page 167
Step 4 Install Assistant Console Plugin, on page 185 The assistant accesses the Unified
Communications Manager Assistant features
by using the Assistant Console application and
the Cisco Unified IP Phone. The Assistant
Console provides call-control functions such as
answer, divert, transfer, and hold.
Step 5 Configure the manager and Assistant Console See Cisco Unified Communications Manager
applications. Assistant User Guide for Cisco Unified
Run the Cisco Unified CM Assistant Configuration Wizard

You can run the Cisco Unified CM Assistant Configuration Wizard to automatically create partitions, calling
search spaces, and route points. The wizard also creates Bulk Administration Tool (BAT) templates for the
manager phones, the assistant phones, and all other user phones. You can use the BAT templates to configure
the managers, assistants, and all other users. For more information about BAT, see Bulk Administration Guide
for Cisco Unified Communications Manager .
Before you begin

Ensure that the configuration wizard runs on the same server (the Unified Communications Manager server)
as the Bulk Administration Tool.
Procedure
Step 1 From Cisco Unified CM Administration, choose Application > Cisco Unified CM Assistant Configuration
Wizard.
Step 2 Click Next to begin the Cisco Unified CM Assistant Configuration wizard process.
160
Run the Cisco Unified CM Assistant Configuration Wizard
Step 3 In the Partition for Managers window, enter a name, provide a description, and then click Next. Alternatively,
you can accept the default partition name and description.
Step 4 In the Partition for CTI Route Point window, enter a name, provide a description, and then click Next.
Alternatively, you can accept the default CTI route point name.
Step 5 In the Partition for All Users window, enter a name, provide a description and then click Next. Alternatively,
you can accept the default partition name and description for all users.
Step 6 In the Intercom Partition window, enter a name, provide a description, and then click Next. Alternatively,
you can accept the default intercom partition name.
Step 7 In the Assistant Calling Search Space window, enter a name, and provide a description. Alternatively, you
can use the default calling search space name and description.
The Available Partitions and Selected Partitions boxes under the Route Partitions for this Calling Search Space
automatically list Partitions for the Assistant Calling Search Space. You can accept the default values or you
can choose the applicable partition from the Available Partitions box. Use the up and down arrows to move
partitions from one box to the other.
Step 8 Click Next.
Step 9 In the Everyone Calling Search Space window, enter a name, and provide a description. Alternatively, you
can accept the default calling search space name and description for everyone.
The Available Partitions and Selected Partitions boxes under the Route Partitions for this Calling Search Space
automatically list Partitions for the Assistant Calling Search Space. You can accept the default values or you
can choose the applicable partition from the Available Partitions box. Use the up and down arrows to move
partitions from one box to the other.
Step 10 Click Next.
If you have existing calling search spaces that are configured on the system, the Existing Calling Search
Spaces window is displayed; otherwise, proceed to the next step.
Manager Assistant requires that the existing calling search spaces add the prefix Generated_Route Point
and Generated_Everyone partitions. The Available Calling Search Spaces and Selected Calling Search
Spaces boxes automatically list these partitions. Use the up and down arrows to move partitions from one box
to the other.
Note The prefix that is added to the existing calling search spaces may change if the administrator has
changed the names of the partitions.
Step 11 Click Next.

Step 12 In the CTI Route Point window, enter a name in the CTI route point name field; otherwise, use the default
CTI route point name.
Step 13 From the drop-down list, choose the appropriate device pool.
Step 14 Enter a route point directory number; otherwise, use the default route point directory number.
Step 15 From the drop-down list, choose the appropriate numbering plan and then click Next.
Step 16 In the Phone Services window, enter the primary phone service name; otherwise, use the default Phone
Service name.
Step 17 From the drop-down list, choose the primary Cisco Unified Communications Manager Assistant server or
enter a server name or IP address.
Step 18 Enter the secondary phone service name; otherwise, use the default phone service name.
Step 19 From the drop-down list, choose the secondary Cisco Unified Communications Manager Assistant server or
enter a server name or IP address and then click Next.
The Confirmation window is displayed. It provides all the information that you chose. If the information is
not correct, you can cancel the configuration process or return to the previous configuration windows.
161
Manager Assistant Service Parameters for Proxy Line
Step 20 Click Finish.

Upon completion, a final status window is displayed.
Any errors that the configuration wizard generates is sent to a trace file. Access this file by using the following
CLI command: file get activelog tomcat/logs/ccmadmin/log4j
What to do next
The Cisco Unified CM Assistant Configuration Wizard only creates the Cisco IP Manager Assistant service
parameters. You must enter the remaining service parameters manually. For service parameter information,
see Manager Assistant Service Parameters for Proxy Line, on page 162.

From Cisco Unified CM Administration, choose System > Service Parameters. Choose the server on which
the Cisco IP Manager Assistant service is active and click ? for detailed descriptions.
Setting Description
Cisco IP Manager Assistant (Active) Parameters
CTIManager (Primary) IP This parameter specifies the IP address of the primary CTIManager that this Cisco IPMA
Address process calls.
No default value.
CTIManager (Backup) IP This parameter specifies the IP address of the backup CTIManager that this Cisco IPMA
Address process calls when primary CTIManager is down.
No default value.
Route Point Device Name for This parameter specifies the device name of the CTI route point that this Cisco IPMA server
Proxy Mode all calls to managers' primary lines for intelligent call routing.
Cisco recommends that you use same CTI route point device for all servers running the IPM
must configure the CTI route point device name if any manager or assistant will be configu
mode.
CAPF Profile Instance Id for This service parameter specifies the Instance ID of the Application CAPF Profile for the
Secure Connection to IPMASecureSysUser that this Manager Assistant will use to open a secure connection to
CTIManager
Configure this parameter if CTIManager Connection Security Flag is enabled.
Clusterwide Parameters (Parameters that apply to all servers)

Important Click Advanced to view the hidden parameters.
Cisco IPMA Server (Primary) This parameter specifies the IP address of the primary Cisco IPMA server.
IP Address
No default value.
Cisco IPMA Server (Backup) This parameter specifies the IP address of the backup Cisco IPMA server. The backup ser
IP Address IPMA service when the primary IPMA server fails.
No default value.
162
Setting Description
Cisco IPMA Server Port This parameter specifies the TCP/IP port on the Cisco IPMA servers to which the IPMA
will open socket connections. You may change the parameter if a port conflict exists.
Default value: 2912
Cisco IPMA Assistant This parameter specifies the interval, in seconds, at which the Cisco IPMA server sends
Console Heartbeat Interval (commonly referred to as heartbeat) to the IPMA Assistant Consoles. The IPMA Assist
failover when they fail to receive heartbeat from the server before the time that is specif
expires.
Default value: 30 seconds
Cisco IPMA Assistant This parameter specifies the time, in seconds, that the IPMA Assistant Consoles wait t
Console Request Timeout from the Cisco IPMA server.
Cisco IPMA RNA Forward This parameter determines whether Cisco IPMA Ring No Answer (RNA) forwarding
Calls values are True (Cisco IPMA forwards unanswered calls to next available assistant) or
does not forward calls).
This parameter works in conjunction with the Cisco IPMA RNA Timeout parameter
after the time that is specified in the Cisco IPMA RNA Timeout parameter. If a voic
specified for the line, unanswered calls that cannot be forwarded to an assistant are sen
this timer expires.
Default value: False
Alpha Numeric UserID This parameter determines whether Cisco IPMA Assistant Phone uses an alphanumeric
user ID.
Default value: True
Cisco IPMA RNA Timeout This parameter specifies the time, in seconds, that the Cisco IPMA server waits befor
unanswered call to the next available assistant. This parameter works in conjunction w
RNA Forward Calls parameter; forwarding occurs only if the Cisco IPMA RNA Forw
is set to True.
CTIManager Connection This parameter determines whether security for the Cisco IP Manager Assistant servic
Security Flag connection is enabled. If it is enabled, Cisco IPMA opens a secure connection to CTIM
CAPF profile that is configured for the instance ID (as specified in the CAPF Profile
Secure Connection to CTIManager service parameter) for the application user IPM
Default value: Non Secure
To enable security, you must select an instance ID in the CAPF Profile Instance ID for
to CTIManager service parameter.
Redirect call to Manager upon This parameter determines whether the Cisco Unified IP Manager Assistant applicatio
failure to reach Assistant back to the intended manager if the call fails to reach the selected proxy assistant.
163
Setting Description
Advanced Clusterwide parameters

Important Configure unique IP addresses for each pool so that the same Cisco IPMA server IP address does not appea
one pool.
Enable Multiple Active Mode This parameter determines whether multiple instances of the Cisco IP Manager Assistant
run for scalability. If it is enabled, Cisco IPMA can run on the other nodes as configured i
Pool 3 parameters.
To enable multiple active mode, you must enter the IP addresses of the nodes on which y
the additional instances of Cisco IPMA. Configure the Cisco IP Manager Assistant servic
those nodes.
Pool 2: Cisco IPMA Server If multiple active mode is enabled, this parameter specifies the IP address of the primary Ci
(Primary) IP Address of the second instance of Cisco IPMA.
Configure the Cisco IP Manager Assistant service parameters on this node.
Pool 2: Cisco IPMA Server If multiple active mode is enabled, this parameter specifies the IP address of the backup Ci
(Backup) IP Address of the second instance of Cisco IPMA. The backup server provides IPMA service when th
server fails.
(Primary) IP Address of the third instance of Cisco IPMA.
(Backup) IP Address of the third instance of Cisco IPMA. The backup server provides IPMA service when the
server fails.
Clusterwide Parameters (Softkey Templates)

Important Configure these parameters if you want to use the Manager Assistant automatic configuration for managers
Assistant Softkey Template This parameter specifies the assistant softkey template that is assigned to assistant devices d
Configuration. The value that is specified in this parameter is used when the Automatic C
check box is checked on the Cisco IPMA Assistant Configuration page.
Manager Softkey Template for This parameter specifies the manager softkey template for proxy mode that is assigned to
Proxy Mode during Automatic Configuration. This parameter applies only for managers that use proxy
Clusterwide Parameters (IPMA Device Configuration Defaults for Proxy Mode)
Manager Partition This parameter defines the partition that is assigned to manager lines that IPMA handles on
during Automatic Configuration. Make sure the partition you want to use has already bee
Unified CM Administration. If the Cisco IPMA Configuration Wizard is run, it will pop
This parameter applies only for managers that use proxy mode.
164
Setting Description
All User Partition This parameter specifies the partition that is configured on all proxy lines and the interc
devices, as well as the intercom line on manager devices, during Automatic Configura
partition you want to use has already been added to Cisco Unified CM Administration
Configuration Wizard is run, it will populate this value. This parameter applies only
assistants that use proxy mode.
IPMA Calling Search Space This parameter specifies the calling search space that is configured for manager lines
that IPMA handles and the intercom line, as well as the assistant intercom line on assi
Automatic Configuration. Make sure the calling search space you want to use has alre
Cisco Unified CM Administration. If Cisco IPMA Configuration Wizard is run, it w
value. This parameter applies only for managers or assistants that use proxy mode.
Manager Calling Search Space This parameter defines the manager calling search space that is configured on proxy l
devices during Automatic Configuration. This calling search space must be a calling s
already exists in the system. If Cisco IPMA Configuration Wizard is run, it will popu
parameter applies only for assistants that use proxy mode.
Cisco IPMA Primary Phone This parameter defines the IP phone service to which manager/assistant devices will b
Service Automatic Configuration. If Cisco IPMA Configuration Wizard is run, it will popu
parameter applies only for managers or assistants that use proxy mode.
Cisco IPMA Secondary Phone This parameter defines the secondary IP phone service to which manager or assistant
Service subscribed during Automatic Configuration. If Cisco IPMA Configuration Wizard is
this value. This parameter applies only for managers or assistants that use proxy mode
Clusterwide Parameters (Proxy Directory Number Range for Proxy Mode)
Starting Directory Number This parameter specifies the starting directory number that is used as the starting num
generation of proxy directory numbers during IPMA assistant configuration. After an a
line number is used for an assistant, the next number will be generated for the next as
This parameter applies only for assistants that use proxy mode.
Ending Directory Number This parameter specifies the ending directory number for automatic generation of prox
during IPMA assistant configuration. Configuration will stop at this number. This par
for assistants that use proxy mode.
Clusterwide Parameters (Proxy Directory Number Range for Proxy Mode)
Number of Characters to be This parameter specifies the number of characters to be stripped from the manager dir
Stripped from Manager DN in the process of generating the proxy DN. Generating a proxy DN may involve stripp
digits and adding a prefix. Digits are stripped starting from the left. This parameter appli
that use proxy mode.
Prefix for Manager DN This parameter specifies the prefix to be added to a manager DN in the process of gen
DN. Generating a proxy DN may involve some stripping of digits and adding a prefix
applies only for assistants that use proxy mode.
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Configure Manager And Assign Assistant For Proxy Line
Configure Manager And Assign Assistant For Proxy Line

For information about configuring a new user and associating a device to the user, see Administration Guide
Note Make sure you configure manager information before you configure assistant information for an assistant.
Procedure
Step 2 Click Find.
The search result displays all the end users that are configured in Unified Communications Manager.
Step 3 From the Related Links drop-down list, choose Manager Configuration and click Go.
Tip To view existing assistant configuration information, click the assistant name in the Associated
Assistants list and click View Details. The Cisco Unified CM Assistant - Assistant
Configuration window is displayed. To return to the manager configuration information, click
the manager name in the Associated Managers list and click View Details.
The Cisco Unified CM Assistant - Manager Configuration window is displayed.
Step 4 From the Device Name/Profile drop-down list, choose the device name or device profile to associate a device
name or device profile with a manager. For more information about Extension Mobility with Manager Assistant,
see Manager Assistant Interactions, on page 186.
Note If the manager telecommutes, click the Mobile Manager check box and optionally choose a
device profile from the Device Name/Profile drop-down list. After you choose a device profile,
the manager must log in to the phone by using extension mobility before accessing Manager
Assistant.
Step 5 From the Intercom Line drop-down list, choose the intercom line appearance for the manager, if applicable.
Note The chosen intercom line applies to the Manager Assistant and Unified Communications Manager
intercom features.
Step 6 From the Assistant Pool drop-down list, choose the appropriate pool number (1 to 3).
Step 7 From the Available Lines selection box, choose a line that you want Manager Assistant to control and click
the down arrow to make the line display in the Selected Lines selection box. Configure up to five Manager
Assistant—controlled lines.
Tip To remove a line from the Selected Lines selection box and from Manager Assistant control,
click the up arrow.
Step 8 Check the Automatic Configuration check box to automatically configure the softkey template, subscribe
to the Manager Assistant phone service, calling search space, and partition for Manager Assistant—Controlled
selected lines and intercom line; and Auto Answer with Speakerphone for intercom line for the manager phone
based on the Cisco IP Manager Assistant service parameters.
Note Automatic Configuration for intercom applies only when using the Manager Assistant intercom
feature for the Cisco Unified IP Phones 7940 and 7960.
166
Configure Assistant Line Appearances for Proxy Line
Step 9 Click Save.

If you checked the Automatic Configuration check box and the service parameters are invalid, a message
displays. Ensure that the service parameters are valid. Upon successful completion of the automatic
configuration, the manager device resets. If you configured a device profile, the manager must log out and
log in to the device for settings to take effect.
Configure Assistant Line Appearances for Proxy Line

A proxy line specifies a phone line that appears on the assistant Cisco Unified IP Phone. Manager Assistant
uses proxy lines to manage calls that are intended for a manager. The administrators can manually configure
a line on the assistant phone to serve as the proxy line, or you can enable the Automatic Configuration check
box to generate a DN and to add the line to the assistant phone.
Note 1. Make sure that you configure manager information and assign an assistant to the manager before you
configure assistant information for an assistant.
2. If you want to automatically configure proxy line on the assistant phone, configure the service parameters
in Proxy Directory Number Range and Proxy Directory Number Prefix sections.
Procedure
Step 2 Click Find.
Step 3 Click on the user name to display user information for the chosen assistant
The End User Configuration window is displayed.
Step 4 From the Related Links drop-down list, choose Assistant Configuration and click Go.
Note The system automatically sets the softkey template and intercom line on the basis of the Cisco
IP Manager Assistant service parameter settings when the Automatic Configuration check box
is checked. In addition, the system also sets Auto Answer with Speakerphone for intercom line.
The Assistant Configuration window is displayed.
Step 5 From the Device Name drop-down list, choose the device name to associate with the assistant.
Step 6 From the Intercom Line drop-down list, choose the incoming intercom line appearance for the assistant.
Step 7 From the Primary Line drop-down list, choose the primary line for the assistant.
Step 8 To associate the manager line to the assistant line, perform the following steps from the Manager Association
to Assistant Line selection box:
a) From the Available Lines drop-down list, choose the assistant line that will be associated with the manager
line.
b) From the Manager Names drop-down list, choose the preconfigured manager name for whom this proxy
line will apply.
c) From the Manager Lines drop-down list, choose the manager line for which this proxy line will apply.
Step 9 Click Save.
167
Manager Assistant Task Flow for Shared Lines
The update takes effect immediately. If you chose Automatic Configuration, the assistant device automatically
resets.
Manager Assistant Task Flow for Shared Lines

Before you begin
• Review Manager Assistant Prerequisites, on page 159.
Procedure

Step 1 Configure Partitions for Manager Assistant Configure a partition for lines that is used by
Shared Line Support, on page 169 Manager Assistant.
Step 2 Configure Calling Search Spaces for Manager Configure calling search spaces for manager
Assistant Shared Line Support, on page 170 and assistant lines.
Step 3 Configure Cisco IP Manager Assistant Service Configure these parameters to use automatic
Parameters, on page 171 configuration for managers and assistants.
Step 4 Configure Intercom Settings
• Configure an Intercom Partition, on page
172
• Configure an Intercom Calling Search
Space, on page 311
• Configure an Intercom Directory Number,
on page 312
• Configure an Intercom Translation
Pattern, on page 312
Step 5 Configure Multiple Manager Assistant Pool, Configure multiple pools if you need to support
on page 174 a large number of managers and assistants.
You can configure up to three active Cisco IP
Manager Assistant servers, with each
managing up to 2500 pairs of managers and
assistants.
Step 6 Configure Secure TLS Connection to CTI for Follow these procedures if your system is
Manager Assistant running in mixed mode.
• Configure IPMASecureSysUser
Application User, on page 175
• Configure CAPF Profile, on page 176
• Configure Cisco WebDialer Web Service
, on page 177
168
Configure Partitions for Manager Assistant Shared Line Support

Step 7 Configure CTI Route Point, on page 178 Cisco Unified Communications Manager
Assistant requires creation of CTI route point
to intercept and route calls from managers.
Step 8 Configure IP Phone Services for Manager and
Assistant, on page 178
Step 9 Configure Phone Button Templates for
Manager, Assistant, and Everyone, on page
182
Step 10 Configure Manager and Assign Assistant for
Shared Line Mode, on page 183
Step 11 Configure Assistant Line Appearances for
Shared Line, on page 184
Step 12 Install Assistant Console Plugin, on page 185 The assistant accesses the Cisco Unified
Communications Manager Assistant features
by using the Assistant Console application and
the Cisco Unified IP Phone. The Assistant
Console provides call-control functions such
as answer, divert, transfer, and hold.
Step 13 Configure the manager and assistant console See Cisco Unified Communications Manager
applications. Assistant User Guide for Cisco Unified
Configure Partitions for Manager Assistant Shared Line Support

You must create three partitions: Generated_Everyone, Generated_Managers, and Generated_Route_Point.
Procedure
Step 1 From Cisco Unified CM Administration, choose Call Routing > Class of Control > Partition.
Step 2 Click Add New to create a new partition.
Step 3 In the Partition Name, Description field, enter a name for the partition that is unique to the route plan.
Partition names can contain alphanumeric characters, as well as spaces, hyphens (-), and underscore characters
(_). See the online help for guidelines about partition names.
Step 4 Enter a comma (,) after the partition name and enter a description of the partition on the same line.
The description can contain up to 50 characters in any language, but it cannot include double quotes ("),
percentage sign (%), ampersand (&), backslash (\), angle brackets (<>), or square brackets ([ ]).
If you do not enter a description, Cisco Unified Communications Manager automatically enters the partition
name in this field.
Step 5 To create multiple partitions, use one line for each partition entry.
Step 6 From the Time Schedule drop-down list, choose a time schedule to associate with this partition.
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Partition Name Guidelines for Manager Assistant Shared Line Support
The time schedule specifies when the partition is available to receive incoming calls. If you choose None, the
partition remains active at all times.
Step 7 Select one of the following radio buttons to configure the Time Zone:
• Originating Device—When you select this radio button, the system compares the time zone of the calling
device to the Time Schedule to determine whether the partition is available is available to receive an
incoming call.
• Specific Time Zone—After you select this radio button, choose a time zone from the drop-down list.
The system compares the chosen time zone to the Time Schedule to determine whether the partition is
available is available to receive an incoming call.
Step 8 Click Save.
Partition Name Guidelines for Manager Assistant Shared Line Support

The list of partitions in a calling search space is limited to a maximum of 1024 characters. This means that
the maximum number of partitions in a CSS varies depending on the length of the partition names. Use the
following table to determine the maximum number of partitions that you can add to a calling search space if
partition names are of fixed length.
Table 19: Partition Name Guidelines
Partition Name Length Maximum Number of Partitions
2 characters 340
3 characters 256
4 characters 204
5 characters 172
... ...
10 characters 92
15 characters 64
Configure Calling Search Spaces for Manager Assistant Shared Line Support
A calling search space is an ordered list of route partitions that are typically assigned to devices. Calling search
spaces determine the partitions that calling devices can search when they are attempting to complete a call.
You must create two calling search spaces: Generated_CSS_I_E and Generated_CSS_M_E.
Procedure
Step 1 From Cisco Unified CM Administration, choose Call Routing > Class of Control > Calling Search Space.
170
Configure Cisco IP Manager Assistant Service Parameters
Step 3 In the Name field, enter a name.

Ensure that each calling search space name is unique to the system. The name can include up to 50 alphanumeric
Step 4 In the Description field, enter a description.

The description can include up to 50 characters in any language, but it cannot include double-quotes ("),
percentage sign (%), ampersand (&), back-slash (\), or angle brackets (<>).
Step 5 From the Available Partitions drop-down list, perform one of the following steps:
• For a single partition, select that partition.
• For multiple partitions, hold down the Control (CTRL) key, then select the appropriate partitions.
Step 6 Select the down arrow between the boxes to move the partitions to the Selected Partitions field.
Step 7 (Optional) Change the priority of selected partitions by using the arrow keys to the right of the Selected
Partitions box.
Step 8 Click Save.
Configure Cisco IP Manager Assistant Service Parameters

Configure Cisco IP Manager Assistant service parameters if you want to use the Manager Assistant automatic
configuration for managers and assistants. You must specify the cluster-wide parameters once for all Cisco
IP Manager Assistant services and general parameters for each Cisco IP Manager Assistant service that is
installed.
Procedure
Step 2 From the Server drop-down list, choose the server on which the Cisco IP Manager Assistant service is active.
Step 3 From the Service drop-down list, choose Cisco IP Manager Assistant service.
The Service Parameter Configuration window, which lists the parameters, is displayed.
Step 4 Configure the Cisco IP Manager Assistant Parameters, Clusterwide Parameters (Parameters that apply
to all servers), and Clusterwide Parameters (Softkey Templates).
click ? for detailed descriptions.
Step 5 Click Save.
Configure Intercom Settings

Procedure

Step 1 Configure an Intercom Partition, on page 172
171
Configure an Intercom Partition

Step 2 Configure an Intercom Calling Search Space,
on page 173
Step 3 Configure an Intercom Directory Number, on
page 173
Step 4 Configure an Intercom Translation Pattern, on
page 173
Configure an Intercom Partition
Procedure
Step 1 From Cisco Unified CM Administration, choose Call Routing > Intercom > Intercom Route Partition.
The Find and List Intercom Partitions window appears.

An Add New Intercom Partition window appears.
Step 3 Under the Intercom Partition Information section, in the Name box, enter the name and description of the
intercom partition that you want to add.
Note To enter multiple partitions, use one line for each partition entry. You can enter up to 75 partitions;
the names and descriptions can have up to a total of 1475 characters. The partition name cannot
exceed 50 characters. Use a comma (,) to separate the partition name and description on each
line. If a description is not entered, Unified Communications Manager uses the partition name
as the description.
Step 4 Click Save.

Step 5 Locate the partition that you want to configure.
Intercom Partition Configuration window is displayed
Step 6 Configure the fields in the Intercom Partition Configuration field area. See the online help for more information
about the fields and their configuration options.
Step 7 Click Save.
The Intercom Partition Configuration window appears.
Step 8 Enter the appropriate settings. For detailed information about the Intercom Partition Configuration parameters,
see online help.
Step 9 Click Save.
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Configure an Intercom Calling Search Space
Procedure
Step 1 In the menu bar, choose Call Routing > Intercom > Intercom Calling Search Space.
Step 2 Click the Add New.
Step 3 Configure the fields in the Intercom Calling Search Space field area. For more information on the fields and
Step 4 Click Save.
Configure an Intercom Directory Number
Procedure
Step 1 Choose Call Routing > Intercom > Intercom Directory Number.
The Find and List Intercom Directory Numbers window is displayed.
Step 2 To locate a specific intercom directory number, enter search criteria and click Find.
A list of intercom directory numbers that match the search criteria displayed.
Step 3 Perform one of the followings tasks:
a) To add an intercom directory number, click Add New.
b) To update an intercom directory number, click the intercom directory number to update.
The Intercom Directory Number Configuration window displayed.
Step 4 Configure the fields in the Intercom Directory Number Configuration field area. For more information on the
Step 5 Click Save.
Step 7 Click Reset Phone.
Step 8 Restart devices.
During the restart, the system may drop calls on gateways.
Configure an Intercom Translation Pattern
Procedure
Step 1 Choose Call Routing > Intercom > Intercom Translation Pattern.
The Find and List Intercom Translation Patterns window appears.
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Configure Multiple Manager Assistant Pool

a) To copy an existing intercom translation pattern, locate the partition to configure, click Copy eside the
intercom translation pattern to copy.
b) To add a new intercom translation pattern, click the Add New.
Step 3 Configure the fields in the Intercom Translation Pattern Configuration field area. For more information on
Step 4 Click Save.
Ensure that the intercom translation pattern that uses the selected partition, route filter, and numbering plan
combination is unique. if you receive an error that indicates duplicate entries, check the route pattern or hunt
pilot, translation pattern, directory number, call park number, call pickup number, or meet-me number
configuration windows.
The Intercom Translation Pattern Configuration window displays the newly configured intercom translation
pattern.
What to do next
Refer to the Manager Assistant Task Flow for Shared Lines, on page 168 to determine the next task to complete.
Configure Multiple Manager Assistant Pool

Procedure
Step 2 From the Server drop-down list, choose the server on which the Cisco IP Manager Assistant service is active.
Step 3 From the Service drop-down list, choose the Cisco IP Manager Assistant service.
The Service Parameter Configuration window, which lists the parameters, is displayed.
Step 4 Click Advanced.
The advanced parameters for Clusterwide Parameters (Parameters that apply to all servers) are displayed.
Step 5 Configure the following parameters to add multiple manager assistant pools in Clusterwide Parameters
(Parameters that apply to all servers):
a) Enable Multiple Active Mode—The default is False. When this parameter is set to True, the administrator
can configure up to 7000 managers and assistants by using multiple pools.
b) Pool 2: Cisco IPMA Server (Primary) IP Address—No default. The administrator must manually enter
this IP address. Administrator can assign up to 2500 managers and assistants to this address.
c) Pool 2: Cisco IPMA Server (Backup) IP Address—No default. The administrator must manually enter
this IP address.
d) Pool 3: Cisco IPMA Server (Primary) IP Address—No default. The administrator must manually enter
this IP address and can assign up to 2500 managers and assistants to this address.
e) Pool 3: Cisco IPMA Server (Backup) IP Address—No default. The administrator must manually enter
this IP address.
click ? for detailed descriptions.
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Configure Secure TLS Connection to CTI for Manager Assistant
Step 6 Click Save.
What to do next
Configure Secure TLS Connection to CTI for Manager Assistant

Manager Assistant uses WDSecureSysUser application user credentials to establish a secure TLS connection
to CTI to make calls.
To configure the WDSecureSysUser application user to establish a secure TLS connection, complete the
following tasks.
Before you begin

• Install and configure the Cisco CTL Client.
For more information about CTL Client, see Security Guide for Cisco Unified Communications Manager.
• Verify that the Cluster Security Mode in the Enterprise Parameters Configuration window is 1
(mixed mode). Operating the system in mixed mode impacts other security functions in your system. If
your system is not currently running in mixed mode, do not switch to mixed mode until you understand
these interactions. For more information, see Security Guide for Cisco Unified Communications Manager.
• Verify that the Cluster SIPOAuth Mode field in the Enterprise Parameters Configuration window
is set to Enabled.
• Activate the Cisco Certificate Authority Proxy Function (CAPF) service on the first node.
Procedure

Step 1 Configure IPMASecureSysUser Application Configure IPMASecureSysUser Application
User, on page 175 User.
Step 2 Configure CAPF Profile, on page 176 Configure Certificate Authority Proxy Function
(CAPF) Profile for the IPMASecureSysUser
Application User.
Step 3 Configure Cisco WebDialer Web Service , on Configure service parameters for the Cisco IP
page 177 Manager Assistant service.
Configure IPMASecureSysUser Application User

Use this procedure to configure IPMASecureSysUser application user.
Procedure
Step 1 From Cisco Unified CM Administration, choose User Management > Application User.
175
Configure CAPF Profile
Step 2 Click Find.

Step 3 From the Find and List Application Users Application window, choose WDSecureSysUser.
Step 4 Configure the fields in the Application User Configuration window and click Save.

Certificate Authority Proxy Function (CAPF) is a component that performs tasks to issue and authenticate
security certificates. When you create an application user CAPF profile, the profile uses the configuration
details to open secure connections for the application.
Procedure
Step 1 From Cisco Unified CM Administration, choose User Management > Application User CAPF Profile.
• Click Add New in the Find window, to add a new CAPF profile.
• Click Copy for that record in the Copy column, to copy an existing profile, and locate the appropriate
profile.
To update an existing entry, locate and display the appropriate profile.
Step 3 Configure or update the relevant CAPF profile fields. See the Related Topics section information about the
fields and their configuration options.
Step 4 Click Save.
Step 5 Repeat the procedure for each application and end user that you want to use security.
CAPF Profile Settings
Setting Description
Application User From the drop-down list, choose the application user for the CAPF operation.
This setting displays configured application users.
This setting does not appear in the End User CAPF Profile window.
End User ID From the drop-down list, choose the end user for the CAPF operation. This setting
displays configured end users.
This setting does not appear in the Application User CAPF Profile window.
Instance ID Enter 1 to 128 alphanumeric characters (a-z, A-Z, 0-9). The Instance ID identifies
the user for the certificate operation.
You can configure multiple connections (instances) of an application. To secure
the connection between the application and CTIManager, ensure that each instance
that runs on the application PC (for end users) or server (for application users)
has a unique certificate.
This field relates to the CAPF Profile Instance ID for Secure Connection to
CTIManager service parameter that supports web services and applications.
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Configure Cisco WebDialer Web Service
Setting Description
Certificate Operation From the drop-down list, choose one of the following options:
• No Pending Operation—This message is displayed when no certificate
operation is occurring. (default setting)
• Install/Upgrade—This option installs a new certificate or upgrades an
existing locally significant certificate for the application.
Authentication Mode The authentication mode for the Install/Upgrade certificate operation specifies
By Authentication String, which means CAPF installs, upgrades, or troubleshoots
a locally significant certificate only when the user or administrator enters the
CAPF authentication string in the JTAPI/TSP Preferences window.
Authentication String To create your own authentication string, enter a unique string.
Each string must contain 4 to 10 digits.
To install or upgrade a locally significant certificate, the administrator must enter
the authentication string in the JTAPI/TSP preferences GUI on the applicationPC.
This string supports one-time use only; after you use the string for the instance,
you cannot use it again.
Generate String To automatically generate an authentication string, click this button. The 4-
to10-digit authentication string appears in the Authentication String field.
Key Size (bits) From the drop-down list, choose the key size for the certificate. The default setting
is 1024. The other option for key size is 512.
Key generation, which is set at low priority, allows the application to function
while the action occurs. Key generation may take up to 30 or more minutes.
Operation Completes by This field, which supports all certificate operations, specifies the date and time
by which you must complete the operation.
The values that are displayed apply for the first node.
Use this setting with the CAPF Operation Expires in (days) enterprise parameter,
which specifies the default number of days in which the certificate operation
must be completed. You can update this parameter at any time.
Certificate Operation This field displays the progress of the certificate operation, such as pending,
Status failed, or successful.
You cannot change the information that is displayed in this field.
Configure Cisco WebDialer Web Service
Procedure
Step 2 From the Server drop-down list, choose the server on which the Cisco WebDialer Web service is active.
177
Step 3 From the Service drop-down list, choose the Cisco WebDialer Web service.
A list of parameters appears.
Step 4 Navigate to and update the CTIManager Connection Security Flag and CAPF Profile Instance ID for Secure
Connection to CTIManager parameters.
To view parameter descriptions, click the parameter name link.
Note CTIManager supports IPv4 and IPv6 addresses.
Step 5 Click Save.

Step 6 Repeat the procedure on each server on which the service is active.
What to do next

Procedure
Step 1 From Cisco Unified CM Administration, choose Device > CTI Route Point.
The CTI Route Point Configuration window is displayed.
Step 3 In the Device Name field, enter the device name.
Step 4 From the Device Pool drop-down list, choose Default.
Step 5 From the Calling Search Space drop-down list, choose Generated_CSS_M_E.
Step 6 Check the Use Device Pool Calling Party Transformation CSS check box.
Step 7 Click Save.
Add successful status message is displayed.
Step 8 From the Association area, click Line [1] - Add a new DN.
Step 9 Enter a directory number in the Directory Number field.
Step 10 From the Route Partition drop-down list, choose Generated_Route_Point.
Step 11 Click Save.
Configure IP Phone Services for Manager and Assistant

Procedure
Step 1 From Cisco Unified CM Administration, choose Device > Device Settings > Phone Services.
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Cisco IP Phone Services Configuration Fields
The IP Phone Services Configuration window is displayed.

Step 3 For each supported phone for managers and assistants, enter the required fields and click Save. See Cisco IP
Phone Services Configuration Fields, on page 179 for more information about the fields and their configuration
options.
The Update successful message is displayed.

Field Description
Service Information
Service Name Enter the name of the service. If the service is not marked as an enterprise
subscription, the service name will display in areas where you can subscribe
to a service, for example, under Cisco Unified Communications Self Care
Portal.
Enter up to 128 characters for the service name.
For Java MIDlet services, the service name must exactly match the name that
is defined in the Java Application Descriptor (JAD) file.
Note Unified Communications Manager allows you to create two or
more IP phone services with identical names. Cisco recommends
that you do not do so unless most or all phone users are advanced,
or unless an administrator always configures the IP phone
services. Be aware that if AXL or any third-party tool accesses
the list of IP phone services for configuration, you must use
unique names for IP phone services.
Note When the service URL points to an external customized URL,
you cannot localize the service name according to the device
locale of the phone. The service name gets displayed in English
alphabets only.
ASCII Service Name Enter the name of the service to display if the phone cannot display Unicode.
Service Description Enter a description of the content that the service provides. The description
can include up to 50 characters in any language, but it cannot include double
quotation marks (") or single quotation marks (').
179
Field Description
Service URL Enter the URL of the server where the IP phone services application is located.
Make sure that this server remains independent of the servers in your Unified
Communications Manager cluster. Do not specify a Unified Communications
Manager server or any server that is associated with Unified Communications
Manager (such as a TFTP server or directory database publisher server).
For the services to be available, the phones in the Unified Communications
Manager cluster must have network connectivity to the server.
For Cisco-signed Java MIDlets, enter the location where the JAD file can be
downloaded; for example, a web server or the back-end application server to
which the Java MIDlet communicates.
For Cisco-provided default services, the service URL is displayed as
Application:Cisco/<name of service> by default; for example,
Application:Cisco/CorporateDirectory. If you modify the
service URL for Cisco-provided default services, verify that you configured
both for the Service Provisioning setting, which displays in the Phone,
Enterprise Parameter, and Common Phone Profile Configuration windows.
For example, you use a custom corporate directory, so you change
Application:Cisco/CorporateDirectory to the external service
URL for your custom directory; in this case, change the Service Provisioning
value to Both.
Secure-Service URL Enter the secure URL of the server where the Cisco Unified IP Phone services
application is located. Make sure that this server remains independent of the
servers in your Unified Communications Manager cluster. Do not specify a
Unified Communications Manager server or any server that is associated with
Unified Communications Manager (such as a TFTP server or publisher database
server).
For the services to be available, the phones in the Unified Communications
Manager cluster must have network connectivity to the server.
Note If you do not provide a Secure-Service URL, the device uses the
nonsecure URL. If you provide both a secure URL and a
nonsecure URL, the device chooses the appropriate URL, based
on its capabilities.
Service Category Choose a service application type (XML or Java MIDlet).

If you choose Java MIDlet, when the phone receives the updated configuration
file, the phone retrieves the Cisco-signed MIDlet application (JAD and JAR)
from the specified Service URL and installs the application.
Service Type Choose whether the service is provisioned to the Services, Directories, or
Messages button or option on the phone; that is, if the phone has these buttons
or options. To determine whether your phone supports these buttons or options,
see the Cisco Unified IP Phone Administration Guide that supports your phone
model.
180
Field Description
Service Vendor Allows you to specify the vendor or manufacturer for the service. This field
is optional for XML applications, but it is required for Cisco-signed Java
MIDlets.
For Cisco-signed Java MIDlets, the value that you enter in this field must
exactly match the vendor that is defined in the MIDlet JAD file.
This field displays as blank for Cisco-provided default services.
You can enter up to 64 characters.
Service Version Enter the version number for the application.

For XML applications, this field is optional and is informational only. For
Cisco-signed Java MIDlets, consider the following information:
• If you enter a version, the service version must exactly match the version
that is defined in the JAD file. If you enter a version, the phone attempts
to upgrade or downgrade the MIDlet if the version is different than what
is installed on the phone.
• If the field is blank, the version gets retrieved from the Service URL.
Leaving the field blank ensures that the phone attempts to download the
JAD file every time that the phone reregisters to Unified Communications
Manager as well as every time that the Cisco-signed Java MIDlet is
launched; this ensures that the phone always runs the latest version of the
Cisco-signed Java MIDlet without you having to manually update the
Service Version field.
This field displays as blank for Cisco-provided default services.

You can enter numbers and periods in this field (up to 16 ASCII characters).
Enable Allows you to enable or disable the service without removing the configuration
from Cisco Unified CM Administration (and without removing the service
from the database).
Uncheck the check box to remove the service from the phone configuration
file and the phone.
Service Parameter Information
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Configure Phone Button Templates for Manager, Assistant, and Everyone
Field Description
Parameters Lists the service parameters that apply to this IP phone service. Use the
following buttons to configure service parameters for this pane:
• New Parameter—Click this button to display the Configure Cisco
Unified IP Phone Service Parameter window, where you configure a
new service parameter for this IP phone service.
• Edit Parameter—Highlight a service parameter that is displayed in the
Parameters pane, then click this button to display the Configure Cisco
Unified IP Phone Service Parameter window, where you can edit the
selected service parameter for this IP phone service.
• Delete Parameter—Highlight a service parameter that is displayed in
the Parameters pane, then click this button to delete a service parameter
for this IP phone service. A popup window asks you to confirm deletion.
Configure Phone Button Templates for Manager, Assistant, and Everyone

The procedures in this section describe how to configure phone button for manager and assistant.
Procedure

Step 1 Configure a Phone Button Template for Perform this step to assign manage and assistant
Manager Assistant, on page 182 button features to line or speed dial keys.
Step 2 Associate a Manager Assistant Button Template Perform this step to configure the manager and
with a Phone, on page 183 assistant button for a phone.
Configure a Phone Button Template for Manager Assistant

Use this procedure to configure a phone button template for the Manager Assistant feature.
Procedure
step.
c) Click Save.
182
Associate a Manager Assistant Button Template with a Phone
Step 6 Click Save.
devices.
Associate a Manager Assistant Button Template with a Phone
Before you begin

Configure a Phone Button Template for Manager Assistant, on page 182
Procedure

button.
Step 5 Click Save.
Configure Manager and Assign Assistant for Shared Line Mode

Procedure
Step 2 Click Find.
Step 3 From the Related Links drop-down list, choose Manager Configuration and click Go.
Step 4 Check the Automatic Configuration check box to automatically configure the softkey template and Auto
Answer with Speakerphone for intercom line for the manager phone based on the Cisco IP Manager Assistant
service parameters.
Note Automatic Configuration for intercom applies only when the Unified Communications Manager
Assistant intercom feature is used for the Cisco Unified IP Phones 7940 and 7960.
Step 5 Check Uses Shared Lines check box.

Step 6 From the Device Name/Profile drop-down list, choose the device name or device profile to associate a device
name or device profile with a manager.
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Configure Assistant Line Appearances for Shared Line
Note If the manager telecommutes, check the Mobile Manager check box and optionally choose a
device profile from the Device Name/Profile drop-down list. When device profile is chosen, the
manager must log in to the phone by using Cisco Extension Mobility before accessing Manager
Assistant.
See the related topics for more information about Extension Mobility with Manager Assistant.
Step 7 From the Intercom Line drop-down list, choose the intercom line appearance for the manager, if applicable.
The chosen intercom line applies to the Manager Assistant and Unified Communications Manager intercom
features.
Step 8 From the Assistant Pool drop-down list, choose the appropriate pool number (1 to 3).
Step 9 Choose the name of the assistant from the Available Assistants selection box and move it to the Associated
Assistants selection box by clicking the down arrow to assign an assistant to the manager.
You can go to the Assistant Configuration window by highlighting the assistant name and clicking the View
Details link.
Step 10 Choose the appropriate line from the Available Lines list box and move it to the Selected Lines list box by
clicking the down arrow to configure the Manager Assistant controlled lines.
Make sure that the controlled line is always the shared line DN.
Step 11 Click Save.
If you checked the Automatic Configuration check box and the service parameters are invalid, a message
is displayed. Ensure that the service parameters are valid. After successful completion of the automatic
configuration, the manager device resets. If you configured a device profile, the manager must log out and
log in to the device for the changes to take effect.
Configure Assistant Line Appearances for Shared Line

Administrators can set up one or more lines with a shared line appearance. The Unified Communications
Manager system considers a directory number to be a shared line if it appears on more than one device in the
same partition.
Procedure
Step 2 Click Find.
Step 3 Click on the username to display user information for the chosen assistant.
Step 4 From the Related Links drop-down list, choose Assistant Configuration and click Go.
The Assistant Configuration window is displayed. The system automatically sets the softkey template and
intercom line on the basis of the Cisco IP Manager Assistant service parameter settings when you check the
Automatic Configuration check box. In addition, the system also sets Auto Answer with Speakerphone for
intercom line.
Step 5 From the Device Name drop-down list, choose the device name to associate with the assistant.
Step 6 From the Intercom Line drop-down list, choose the incoming intercom line appearance for the assistant.
Step 7 From the Primary Line drop-down list, choose the primary line for the assistant.
184
Install Assistant Console Plugin
a) To view existing manager configuration information, highlight the manager name in the Associated
Managers list and click View Details.
The Manager Configuration window is displayed.
b) To return to the Assistant Configuration window, highlight the assistant name and click View Details
link in the Manager Configuration window.
In the Associated Manager selection list box, the name of the previously configured manager is displayed.
Step 8 To associate the manager line to the assistant line, perform the following steps from the Manager Association
to Assistant Line selection box:
a) From the Available Lines drop-down list, choose the assistant line that will be associated with the manager
line.
b) From the Manager Names drop-down list, choose the preconfigured manager name for whom this proxy
line will apply.
c) From the Manager Lines drop-down list, choose the manager line for which this proxy line will apply.
Step 9 Click Save.
The update takes effect immediately. If you chose Automatic Configuration, the assistant device automatically
resets.
Install Assistant Console Plugin

Procedure
Step 1 From Cisco Unified CM Administration, choose Application > Plugins.

The Find and List Plugins window is displayed.
Step 2 Click Find.
A list of installable application plug-ins is displayed.
Step 3 Click on the Download link for Cisco Unified CM Assistant Console and save the executable to a location.
Step 4 Run the executable file.
Note If you install the application on a Windows Vista PC, a security window may be displayed. Allow
the installation to continue.
The Cisco Unified CallManager Assistant Console installation wizard is displayed.
Step 5 In the Introduction window, click Next.
Step 6 In the License Agreement window, click Next.
Step 7 Choose a location where you want the application to install and click Next.
Note By default, the application installs in C:\Program Files\Cisco\ Unified
CallManager Assistant Console.
Step 8 In the Pre-installation Summary window, review the summary and click Install.
The installation begins.
Step 9 After the installation is complete, click Finish.
Step 10 Provide the assistant the username and password that is required to log in to the console.
Step 11 To launch the Assistant Console, click the desktop icon or choose Cisco Unified Communications Manager
Assistant > Assistant Console from the Start...Programs menu.
185
Manager Assistant Interactions
Step 12 The Advanced tab in the Cisco Unified Communications Manager Assistant Settings window allows you
to enable trace for the Assistant Console.
Step 13 Provide the assistant with the port number and the IP address or hostname of the Unified Communications
Manager server on which the Cisco IP Manager Assistant service is active. The first time that the assistant
logs in to the console, the assistant must enter the information in the Cisco Unified Communications Manager
Assistant Server Port and the Cisco Unified Communications Manager Assistant Server Hostname or
IP Address fields.

Feature Interaction
Bulk Administration Tool You can use the Bulk Administration Tool to add many users (managers
and assistants) at once instead of adding users individually.
The Bulk Administration Tool templates that the Cisco Unified CM
Assistant Configuration Wizard creates for Cisco Unified IP Phones
support only the Unified Communications Manager intercom lines.
For more information, see the Bulk Administration Guide for Cisco
Calling Party Normalization Manager Assistant automatically supports localized and globalized calls
if you configure the Calling Party Normalization feature. Manager
Assistant can display localized calling party numbers on the user
interfaces. In addition, for an incoming call to the manager, Manager
Assistant can display localized and globalized calling party numbers
when filter pattern matching occurs.
Extension Mobility You can simultaneously use Manager Assistant with the Cisco Extension
Mobility feature. When you log in to the Cisco Unified IP Phone using
Extension Mobility, the Cisco IP Manager Assistant service is
automatically enabled on that phone. You can then access the Manager
Assistant features.
For more information about Cisco Extension Mobility, see Extension
Mobility Overview, on page 397.
Internet Protocol Version 6 (IPv6) Manager Assistant does not support IPv6, so you cannot use phones with
an IP Addressing Mode of IPv6 Only with Manager Assistant. To use
Manager Assistant with the phone, ensure that you configure the phone
with an IP Addressing Mode of IPv4 Only or IPv4 and IPv6.
186
Feature Interaction
Reporting tools Manager Assistant provides statistical information in the CDR Analysis
and Reporting (CAR) tool and provides a summary of changes to
configurations in a change log.
The administrator can view a summary of changes that are made to the
Manager or Assistant Configurations in Unified CM
AssistantChangeLog*.txt. A manager can change defaults by accessing
the Manager Configuration from a URL. An assistant can change the
manager defaults from the Assistant Console. For information about the
URL and Manager Configuration, see the Cisco Unified Communications
Manager Assistant User Guide.
When the manager or assistant makes changes, the changes are sent to a
log file called ipma_changeLogxxx.log. The log file resides on the server
that runs the Cisco IP Manager Assistant service. Use the following
command to obtain the log file: file get activelog tomcat/logs/ipma/log4j/
For more information about downloading the log file, see the Cisco
Unified Real -Time Monitoring Tool Administration Guide.
CDR Analysis and Reporting Manager Assistant supports call-completion statistics and inventory
reporting for managers and assistants. The CAR tool supports
call-completion statistics. Cisco Unified Serviceability supports inventory
reporting.
For more information, see the following guides:
• Cisco Unified Serviceability Administration Guide
• Call Reporting and Billing Administration Guide for Cisco Unified
Multilevel Precedence and The following points describe the interactions between Manager Assistant
Preemption (MLPP) with shared line support and MLPP:
• The system preserves call precedence in the handling of calls by
Manager Assistant. For example, when an assistant diverts a call,
the system preserves the precedence of the call.
• Filtering of precedence calls occurs in the same manner as all other
calls. The precedence of a call will not affect whether a call is
filtered.
• Because Manager Assistant does not have information about the
precedence of a call, it does not provide any additional indication
of the precedence of a call on the Assistant Console.
187
Manager Assistant Restrictions
Feature Interaction
Intercom Manager Assistant supports the following two types of intercom:

• Manager Assistant intercom (used with Cisco Unified IP Phones
7940 and 7960). You can configure this intercom feature using the
DN configuration and end user (manager and assistant) configuration
windows.
• Unified Communications Manager intercom (used with Cisco Unified
IP Phones 7940 and 7960). You can configure this intercom feature
using the intercom partition, intercom calling search space, intercom
directory number, intercom translation pattern, DN, and end user
(manager and assistant) configuration windows.
Message Waiting Indicator The Message Waiting Indicator feature interacts with proxy line support
only.
The Message Waiting Indicator (MWI) on and off numbers should have
the partition of the manager line in their calling search space. The partition
can exist in any order of priority within each calling search space.
Time-of-Day Routing The Time-of-Day feature interacts with proxy line support only.
Time-of-Day routing routes calls to different locations based on the time
that the call gets made; for example, during business hours, calls get
routed to a manager office, and after hours, the calls go directly to
voicemail service.
For more information about Time-of-Day Routing, see the System
Configuration Guide for Cisco Unified Communications Manager.
Manager Assistant Restrictions

Feature Restriction
Assistant Console Application To install the Assistant Console application on a computer with Microsoft
Internet Explorer 7 (or later), install the Microsoft Java Virtual Machine
(JVM) before the Assistant Console installation.
Call Management features The Assistant Console does not support hunt groups or queues, recording
and monitoring, one-touch Call Pickup, and On-Hook transfer (the ability
to transfer a call by pressing the Transfer softkey and going on hook to
complete the transfer).
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Cisco Unified Communications Manager Assistant Troubleshooting
Feature Restriction
Cisco IP Phones Manager Assistant supports SIP on Cisco Unified IP Phones 7900 Series,
except for Cisco Unified IP Phones 7940 and 7960.
Manager Assistant supports up to 3500 managers and 3500 assistants by
configuring multiple Cisco IP Manager Assistant servers (pools). When you
enable multiple pools, the manager and all configured assistants for that
manager should belong to the same pool.
Cisco Unified IP Phones 7960 and 7940 support only the Unified
Communications Manager Assistant Intercom lines feature. Cisco Unified
IP Phones 7900 (except 7940 and 7960) support only the Unified
Communications Manager Intercom feature.
One manager can have up to ten assigned assistants and one assistant can
support up to 33 managers (if each manager has one Unified
Communications Manager–controlled line).
Only one assistant at a time can assist a manager.
Manager Assistant supports up to 3500 managers and 3500 assistants per
Unified Communications Manager cluster.
Intercom After an upgrade, Manager Assistant users that use the incoming intercom
line do not get upgraded automatically to the Unified Communications
Manager Intercom feature.
The system does not support calls between the Unified Communications
Manager Intercom feature and regular lines (which may be configured as
Manager Assistant Intercom lines).
Single Sign-On Manager Assistant is not supported in the Single Sign-On environment.
Speed Dial Cisco Unified IP Phones 7940, 7942, and 7945 support only two lines or
speed-dial buttons.
Cisco Unified Communications Manager Assistant

Troubleshooting
This section describes the troubleshooting tools for Manager Assistant and the client desktop, and
troubleshooting information for Manager Assistant.
189
Calling Party Gets Reorder Tone
Tool Description Location
Cisco Unified CM Assistant server The trace files reside on the server that runs the Cisco IP Manager
trace files Assistant service.
You can download these files from the server using one of the following
methods:
• Use the CLI command file get activelog tomcat/logs/ipma/log4j.
• Use the trace collection features in the Cisco Unified Real-Time
Monitoring Tool (RTMT). For more information, see the Cisco
Unified Real-Time Monitoring Tool Administration Guide.
You can enable debug tracing by choosing Cisco Unified

Serviceability > Trace > Configuration.
Cisco IPMA client trace files $INSTALL_DIR\logs\ACLog*.txt on the client desktop, in the same
location where the Unified CM Assistant assistant console resides.
To enable debug tracing, go to the Settings dialog box in the Assistant
Console. In the Advanced panel, check the Enable Trace check box.
Note This check box enables only debug tracing. Error tracing
always remains On.
Cisco IPMA client install trace $INSTALL_DIR\InstallLog.txt on the client desktop, in the same
files location where the Assistant Console resides.
Cisco IPMA Client AutoUpdater $INSTALL_DIR\UpdatedLog.txt on the client desktop, in the same
trace files location where the Unified CM Assistant Console resides.
Install directory By default—C:\Program Files\Cisco\Unified

Communications Manager Assistant Console\
Calling Party Gets Reorder Tone

Problem
Calling party gets a reorder tone or a message:
This call cannot be completed as dialed.
Possible Cause
The calling search space of the calling line may not be configured correctly.
Solution
Check the calling search space of the line. For more information about configuration, see the System
You can also use the Cisco Dialed Number Analyzer service to check for flaws in the calling search space.
For more information, see the Cisco Unified Communications Manager Dialed Number Analyzer Guide.
190
Calls Do Not Get Routed When Filtering Is On or Off
Calls Do Not Get Routed When Filtering Is On or Off

Problem
Calls are not routed properly.
Possible Cause 1
Cisco CTI Manager service may have stopped.
Solution 1
Restart the Cisco CTI Manager and Cisco IP Manager Assistant services from Cisco Unified Serviceability >
Tools > Control Center—Feature Services.
Possible Cause 2
The Unified Communications Manager Assistant route point was not configured properly.
Solution 2
Use wildcards to match the directory number of the Unified Communications Manager Assistant CTI route
point and the primary directory numbers of all managers that are configured for Unified Communications
Manager Assistant.
Possible Cause 3
The status window on the manager phone displays the message Filtering Down. This message can
indicate that Unified Communications Manager Assistant CTI route point may be deleted or may not be in
service.
Solution 3
Use the following procedure to configure the CTI route point and restart the Cisco IP Manager Assistant
service:
1. From Cisco Unified CM Administration, choose Device > CTI Route Point.
2. Find the route point, or add a new route point. For more information about configuration, see the System
3. Restart the Cisco CTI Manager and Cisco IP Manager Assistant services from Cisco Unified
Serviceability > Tools > Control Center—Feature Services.
Cisco IP Manager Assistant Service Unreachable

Problem
After you open the Assistant Console, the following message is displayed:
Cisco IPMA Service Unreachable
191
Cisco IP Manager Assistant Service Unreachable
Possible Cause 1
Cisco IP Manager Assistant service may have stopped.
Solution 1
Restart the Unified Communications Manager Assistant from Cisco Unified Serviceability > Tools > Control
Center—Feature Services.
Possible Cause 2
The server address for the primary and secondary Unified Communications Manager Assistant servers may
be configured as DNS names, but the DNS names are not configured in the DNS server.
Solution 2
Use the following procedure to replace the DNS name.
1. From Cisco Unified CM Administration, choose System > Server.
2. Replace the DNS name of the server with the corresponding IP address.
3. Restart the Unified Communications Manager Assistant from Cisco Unified Serviceability > Tools >
Control Center—Feature Services.
Possible Cause 3
The Cisco CTI Manager service may have stopped.
Solution 3
Possible Cause 4
The Unified Communications Manager Assistant service might be configured to open a CTI connection in
secure mode, but the security configuration may not be complete.
If this scenario occurs, the following message is displayed in the alarm viewer or in the Unified Communications
Manager Assistant service logs:
IPMA Service cannot initialize - Could not get Provider.
Solution 4
Check the security configuration in the service parameters of Cisco IP Manager Assistant service.
192
Cannot Initialize Cisco IP Manager Assistant Service
Cannot Initialize Cisco IP Manager Assistant Service

Problem
The Cisco IP Manager Assistant service cannot open a connection to CTI Manager, and the following message
is displayed:
IPMA Service cannot initialize - Could not get Provider
Possible Cause
The Cisco IP Manager Assistant service cannot open a connection to CTIManager. You can see the message
in the alarm viewer or in the Unified CM Assistant service logs.
Solution
Restart the Cisco CTI Manager and Cisco IP Manager Assistant services from Cisco Unified Serviceability >
Tools > Control Center—Feature Services.
Assistant Console Installation from Web Fails

Problem
Assistant Console installation from the web fails. The following message is displayed:
Exception: java.lang.ClassNotFoundException: InstallerApplet.class
Possible Cause
Using the Sun Java plug-in virtual machine instead of the Microsoft JVM with the standard Unified
Communications Manager Assistant Console install causes failures.
Solution
The administrator directs the user to the following URL, which is a JSP page that supports the Sun Java
plug-in:
https://<servername>:8443/ma/Install/IPMAConsoleInstallJar.jsp
HTTP Status 503—This Application Is Not Currently Available

Problem
http://<server-name>:8443/ma/Install/IPMAConsoleInstall.jsp displays the following error message:
HTTP Status 503—This application is not currently available
Possible Cause
Cisco IP Manager Assistant service has not been activated or is not running.
193
Manager Is Logged Out While the Service Is Still Running
Solution
Ensure that the Cisco IP Manager Assistant service has been activated by checking the activation status of
the service from Cisco Unified Serviceability > Tools > Service Activation.
If the Cisco IP Manager Assistant service has already been activated, restart the Unified Communications
Manager Assistant from Cisco Unified Serviceability > Tools > Control Center—Feature Services.
Manager Is Logged Out While the Service Is Still Running

Problem
Although the manager is logged out of Unified Communications Manager Assistant, the service still runs.
The display on the manager IP phone disappears. Calls do not get routed, although filtering is On. To verify
that the manager is logged out, view the application log using the Cisco Unified Real-Time Monitoring Tool.
Look for a warning from the Cisco Java Applications that indicates that the Cisco IP Manager Assistant service
logged out.
Possible Cause
The manager pressed the softkeys more than four times per second (maximum limit allowed).
Solution
The Unified Communications Manager administrator must update the manager configuration. Perform the
following procedure to correct the problem:
1. From Cisco Unified CM Administration, choose User Management > End User.
The Find and List Users window is displayed.
2. Enter the manager name in the search field and click Find.
3. From the search results list, choose the manager that you want to update.
4. From the Related Links drop-down list, choose Cisco IPMA Manager and click Go.
5. Make the necessary changes to the manager configuration and click Update.
Manager Cannot Intercept Calls That Are Ringing on the Assistant Proxy Line
Problem
The manager cannot intercept the calls that are ringing on the assistant proxy line.
Possible Cause
The calling search space of the proxy line is not configured properly.
194
No Page Found Error
Solution
Check the calling search space of the proxy line for the assistant phone. Perform the following procedure to
correct the problem:
1. From Cisco Unified CM Administration, choose Device > Phone.
The Find and List Phones search window is displayed.
2. Click the assistant phone.
3. Verify the calling search space configuration for the phone and for the directory number (line) and update
as appropriate.
No Page Found Error

Problem
http://<server-name>:8443/ma/Install/IPMAConsoleInstall.jsp displays the following
error message:
No Page Found Error
Possible Cause 1
Network problems.
Solution 1
Ensure that the client has connectivity to the server. Ping the server name that is specified in the URL and
verify that it is reachable.
Possible Cause 2
Misspelled URL.
Solution 2
Because URLs are case sensitive, ensure that the URL matches exactly with the URL in the instructions.
System Error - Contact System Administrator

Problem
After you open the Assistant Console, the following message is displayed:
System Error - Contact System Administrator
Possible Cause 1
You may have upgraded the Unified Communications Manager. The system does not upgrade the Assistant
Console automatically when you upgrade the Unified Communications Manager.
195
Unable to Call Manager When Cisco IP Manager Assistant Service is Down
Solution 1
Uninstall the console by choosing Start > Programs > Cisco Unified Communications
Manager Assistant > Uninstall Assistant Console and reinstall the console from URL
https://<server-name>:8443/ma/Install/IPMAConsoleInstall.jsp.
Possible Cause 2
The user is not configured correctly in the database.
Solution 2
Ensure that the user ID and the password are run as a Unified Communications Manager user through Cisco
Unified CM Administration.
Possible Cause 3
When you deleted a manager from an assistant, Cisco Unified CM Administration left a blank line for the
assistant.
Solution 3
From the Assistant Configuration window, reassign the proxy lines.
Unable to Call Manager When Cisco IP Manager Assistant Service is Down

Problem
Calls do not get routed properly to managers when Cisco IP Manager Assistant service goes down.
Possible Cause
The Unified Communications Manager Assistant CTI route point does not have Call Forward No Answer
enabled.
Solution
Perform the following procedure to properly configure the Unified Communications Manager Assistant route
point.
1. From Cisco Unified CM Administration, choose Device > CTI Route Point.
The Find and List CTI Route Point search window is displayed.
2. Click Find.
A list of configured CTI route points is displayed.
3. Choose the Unified Communications Manager Assistant CTI route point that you want to update.
4. In the CTI Route Point Configuration window, click the line to update from the Association area.
5. In the Call Forward and Pickup Settings section, check the Forward No Answer Internal and the
Forward No Answer External check box and enter the CTI route point DN in the Coverage/Destination
field (for example, CFNA as 1xxx for the route point DN 1xxx).
196
User Authentication Fails
6. In the Calling Search Space drop-down list, choose CSS-M-E (or appropriate calling search space).
7. Click Update.

Problem
User authentication fails when you sign in using the login window from the Assistant Console.
Possible Cause
The following probable causes can apply:
• Incorrect management of the user in the database
• Incorrect management of the user as an assistant or a manager
Solution
Ensure that the user ID and the password are ran as a Unified Communications Manager user through Cisco
You must run the user as an assistant or a manager by associating the Unified Communications Manager
Assistant user information, which you access through Cisco Unified CM Administration under User
Management > End User.
197
198
PA R T VII
Voice Messaging Features
• Audible Message Waiting Indicator , on page 201
• Immediate Divert , on page 205
CHAPTER 16
Audible Message Waiting Indicator
• Audible Message Waiting Indicator Overview, on page 201
• Audible Message Waiting Indicator Prerequisites, on page 201
• Audible Message Waiting Indicator Configuration Task Flow, on page 201
• Audible Message Waiting Indicator Troubleshooting, on page 203
Audible Message Waiting Indicator Overview

You can configure Audible Message Waiting Indicator (AMWI) to play a stutter dial tone on the Cisco Unified
IP Phone to notify users of new voice messages. Users hear a stutter dial tone whenever the phone goes off
hook on a line on which a voice message was left.
You can configure AMWI for all the phones in a cluster or for only certain directory numbers. The
directory-number-level configuration takes precedence over the cluster-wide configuration.
Audible Message Waiting Indicator Prerequisites

You can configure AMWI only on Cisco Unified IP Phone that are running phone firmware Release 8.3(1)
or later.
Audible Message Waiting Indicator Configuration Task Flow

Before you begin
• Review Audible Message Waiting Indicator Prerequisites, on page 201.
Procedure

support the Audible Message Waiting Indicator
feature.
201
Configure Audible Message Waiting Indicator Service Parameters

Step 2 Configure Audible Message Waiting Indicator Configure AMWI default setting for all phones
Service Parameters, on page 202 in a cluster.
Step 3 Configure Audible Message Waiting Indicator Configure AMWI for a directory number that
for a Directory Number, on page 202 is associated to a device.
Step 4 Configure Audible Message Waiting Indicator Configure AMWI for SIP profiles. Perform this
for a SIP Profile, on page 203 procedure to configure AMWI for SIP phones.
Configure Audible Message Waiting Indicator Service Parameters

This procedure describes how to configure AMWI default setting for all the phones in a cluster.
Before you begin

Generate a Phone Feature List, on page 1
Procedure
Step 2 From the Server drop-down list, choose the server that is running the Cisco CallManager service.
Step 4 In the Clusterwide Parameters ( Feature - General) section, choose the Audible Message Waiting Indication
Policy service parameter. This parameter determines whether the Audible Message Waiting Indicator is turned
on of off for all the devices in the cluster.
Step 5 Click Save.
Configure Audible Message Waiting Indicator for a Directory Number

Follow these steps to configure AMWI for a directory number that is associated with a device.
Note The AMWI setting on an individual directory number overrides the clusterwide setting.
Procedure

Step 2 In the Association section, click Add a new DN.
The Directory Number Configuration window appears.
Step 3 Select the Audible Message Waiting Indicator Policy. Choose one of the following options:
• Off
202
Configure Audible Message Waiting Indicator for a SIP Profile
• On—When you select this option, the users will receive a stutter dial tone when the handset is off hook
.
• Default—When you select this option, the phone uses the default that was set at the system level.
Step 4 Configure the remaining fields in the Directory Number Configuration window. See the online help for
more information about the fields and their configuration options.
Step 5 Click Save.
Configure Audible Message Waiting Indicator for a SIP Profile

Follow these steps to configure Audible Message Waiting Indicator (AMWI) for a SIP profile.
Note The AMWI setting on an individual SIP profile overrides the clusterwide setting.
Procedure
The Find and List SIP Profiles window appears.
Step 2 Enter the search criteria to use and click Find.
The window displays a list of SIP profiles that match the search criteria.
Step 3 Click the SIP profile that you want to update.
The SIP Profile Configuration window appears.
Step 4 Check the Stutter Message Waiting check box to activate stutter dial tone when the phone is off hook and
a message is waiting.
Step 5 Click Save.
Audible Message Waiting Indicator Troubleshooting

Audible Message Waiting Indicator Is Not Heard on the Phone
Problem Phone does not play stutter dial tone to notify the user of new voice messages.
If the user uses an SCCP phone, check the following:

• Ensure that the phone firmware release is 8.3(1) or later.
• Check the AMWI setting for the phone and the line on which the user went off hook.
• Verify that the Cisco CallManager service is running on the server.
203
Localized AMWI Tone Is Not Played in a Specific Locale
• Check the sniffer trace between the phone and Unified Communications Manager. Make sure that the
phone receives the StartTone message with tone type equal to 42.
If the user uses a SIP phone, check the following:

• Ensure that the phone firmware release is 8.3(1) or later.
• Check the line (directory number) configuration. The phone must display the settings such as
line1_msgWaitingAMWI : 1, line2_msgWaitingAMWI : 0.
• Ensure that the Stutter Message Waiting check box is checked in the SIP Profile Configuration window
in Cisco Unified CM Administration.
Localized AMWI Tone Is Not Played in a Specific Locale

Problem The phone that is configured in a non-English locale does not play the localized tone.
Solution Check the following:
• From Cisco Unified CM Administration, verify the User Locale in the Device Profile Configuration
window (Device > Device Settings > Device Profile).
• Make sure that the user resets the phone after changing the locale.
• Check user/local/cm/tftp /<locale name> directory and verify that the AMWI tone
is defined in the localized g3-tones.xml file.
204
CHAPTER 17
Immediate Divert
• Immediate Divert Overview, on page 205
• Immediate Divert Prerequisites, on page 206
• Immediate Divert Configuration Task Flow, on page 206
• Immediate Divert Interactions, on page 211
• Immediate Divert Restrictions, on page 212
• Immediate Divert Troubleshooting, on page 214
Immediate Divert Overview

The Immediate Divert feature is a Unified Communications Manager supplementary service that allows you
to immediately divert a call to a voicemail system. When Immediate Divert diverts a call, the line becomes
available to make or receive new calls. Access the Immediate Divert feature by using the iDivert or Divert
softkey on the IP phone.
Immediate Divert provides the following functions:
• Diverts a call to a voicemail system in the following manner:
• Legacy iDivert diverts the call to the voice mailbox of the party that invokes the iDivert feature.
• Enhanced iDivert diverts the call to either the voice mailbox of the party that invokes the iDivert
feature or to the voice mailbox of the original called party.
• Diverts inbound calls that are in the Call Offering, Call on Hold, or Call Active states.
• Diverts outbound calls in the Call Active or Call on Hold states.
Note Although the Immediate Divert feature is not available to CTI applications, a CTI redirect operation exists
that performs the same function as Immediate Divert. Application developers can use the CTI redirect operation
to accomplish Immediate Divert.
205
Immediate Divert Prerequisites
Immediate Divert Prerequisites

• You must configure the voicemail profiles and hunt pilots.
For information on how to configure voicemail profiles and hunt pilots, see System Configuration Guide
for Cisco Unified Communications Manager
• The following devices support Immediate Divert:
• Voice-messaging systems such as Cisco Unity Connection that use the Skinny Client Control
Protocol (SCCP).
• QSIG devices (QSIG-enabled H.323 devices, MGCP PRI QSIG T1 gateways, and MGCP PRI QSIG
E1 gateways), depending on the setting of the Use Legacy Immediate Divert and Allow QSIG
During iDivert clusterwide service parameters.
• The following table lists the phones that use the Divert or iDivert softkey.
Table 20: Cisco Unified IP Phones That Use Immediate Divert Softkeys
Cisco Unified IP Phone Divert Softkey iDivert Softkey What to configure in

Model softkey template
Cisco Unified IP Phone X iDivert

6900 Series (except
6901 and 6911)
Cisco Unified IP Phone X iDivert

7900 Series
Cisco Unified IP Phone X Configured by default

8900 Series
Cisco Unified IP Phone X Configured by default

9900 Series
Note Cisco Unified IP Phones 8900 and 9900 series have the Divert softkey assigned by default.
Immediate Divert Configuration Task Flow

Before you begin
• Review Immediate Divert Prerequisites, on page 206.
206
Configure Immediate Divert Service Parameters
Procedure

Step 1 Configure Immediate Divert Service Configure the service parameters to enable
Parameters, on page 207 Immediate Divert across various devices and
applications.
Step 2 Configure a Softkey Template for Immediate Create and configure a softkey template and
Divert, on page 208 add the iDivert softkey to that template.
Step 3 To Associate a Softkey Template with a Optional. To make the softkey template
Common Device Configuration, on page 209, available to phones, you must complete either
complete the following subtasks: this step or the following step. Follow this step
if your system uses a Common Device
• Add a Softkey Template to the Common
Configuration to apply configuration options
Device Configuration, on page 210
to phones.
• Associate a Common Device
Configuration with a Phone, on page 210 This is the most commonly used method for
making a softkey template available to phones.
Step 4 Associate a Softkey Template with a Phone, on Optional. Use this procedure either as an
page 211 alternative to associating the softkey template
with the Common Device Configuration, or in
conjunction with the Common Device
Configuration. Use this procedure in
Configuration if you need assign a softkey
template that overrides the assignment in the
Common Device Configuration or any other
default softkey assignment.
Configure Immediate Divert Service Parameters

Procedure
Step 4 Configure the relevant service parameters and click Save.
Table 21: Service Parameter Fields for Immediate Divert
Field Description
Call Park Display Timer Enter a number from 0 to 100 (inclusive) to control the timer for the Immediate
Divert text display on the IP phones. Set this timer for the server or for each
server in a cluster that has the Cisco CallManager service and Immediate Divert
configured. The default value for this service parameter is 10 seconds.
207
Configure a Softkey Template for Immediate Divert
Field Description
Use Legacy Immediate Select one of the following options from the drop-down list:
Divert
• True—The user that invokes the iDivert feature can divert an incoming call
only to his own voice mailbox. This is the default setting.
• False—Immediate Divert allows diversion of an incoming call to either the
voice mailbox of the original called party or to the voice mailbox of the user
that invokes the iDivert feature.
Allow QSIG During Select one of the following options from the drop-down list:
iDivert
• True—Immediate Divert diverts calls to voicemail systems that can be
reached over QSIG, SIP, and QSIG-enabled H.323 devices.
• False—Immediate Divert does not support access to voicemail systems over
QSIG or SIP trunks. This is the default setting.
Immediate Divert User Enter a number from 5 to 30 (inclusive) to determine the time given to the iDivert
Response Timer softkey user to choose the party to whom to divert a call. If the user does not
choose a party, the call remains connected. The default value for this service
parameter is 5 seconds.
Configure a Softkey Template for Immediate Divert

To divert incoming calls or outgoing calls, configure a softkey template and assign the iDivert softkey to that
template. You can configure the iDivert softkey in the following call states:
• Connected
• On hold
• Ring in
Immediate Divert supports the following call states:

• For incoming calls:
• Call offering (shown as Ring In on the softkey template).
• Call on hold
• Call active
• For outgoing calls:

• Call on hold
• Call active
208
Procedure
a) Click Add New.
d) Click Save.
template.
click Go.
to display.
Step 9 Click Save.
devices.

Optional. There are two ways to associate a softkey template with a phone:
• Add the softkey template to the Phone Configuration.
• Add the softkey template to the Common Device Configuration.
The procedures in this section describe how to associate the softkey template with a Common Device
Configuration. Follow these procedures if your system uses a Common Device Configuration to apply
configuration options to phones. This is the most commonly used method for making a softkey template
available to phones.
209
To use the alternative method, see Associate a Softkey Template with a Phone, on page 211
Procedure

Step 1 Add a Softkey Template to the Common Device
Step 2 Associate a Common Device Configuration
with a Phone, on page 210
Procedure
a) Click Add New.
c) Click Save.
to make available.
Step 5 Click Save.
restart them.
Associate a Common Device Configuration with a Phone
Procedure

Step 4 Click Save.
210

Optional. Use this procedure as an alternative to associating the softkey template with the Common Device
Configuration. This procedure also works in conjunction with the Common Device Configuration. You can
use it when you need to assign a softkey template that overrides the assignment in the Common Device
Configuration or any other default softkey assignment.
Before you begin

Configure a Softkey Template for Immediate Divert, on page 208
Procedure

Step 4 Click Save.
Immediate Divert Interactions

Feature Interaction
Multilevel Precedence and Immediate Divert diverts calls to voice-messaging mailboxes regardless of the
Preemption (MLPP) type of call (for example, a precedence call).
When Alternate Party Diversion (call precedence) is activated, Call Forward No
Answer (CFNA) gets deactivated.
Call Forward When the Forward No Answer setting on the Directory Number Configuration
window is not configured, Call Forward uses the clusterwide CFNA timer service
parameter, Forward No Answer Timer.
If a user presses the iDivert softkey at the same time as the call is being forwarded,
the call gets diverted to an assigned call forward directory number (because the
timer was too short), not the voice-messaging mailbox. To resolve this situation,
set the CFNA timer service parameter to enough time (for example, 60 seconds).
Call Detail Records Immediate Divert uses the immediate divert code number in the Onbehalf of
(CDR) fields (for example, joinOnbehalfOf and lastRedirectRediectOnBehalfOf) in
CDR.
211
Immediate Divert Restrictions
Feature Interaction
Call Park and Directed When user A calls user B, and user B parks the call; user B retrieves the call and
Call Park then decides to send the call to a voice-messaging mailbox by pressing the iDivert
or Divert softkey. User A receives the voice-messaging mailbox greeting of user
B.
Conference When a conference participant presses the iDivert softkey, the remaining
conference participants receive the voice-messaging mailbox greeting of the
immediate divert initiator. Conference types include Ad Hoc, Meet-Me, Barge,
cBarge, and Join.
Hunt List For calls that reach the phone directly through a hunt list pilot (as part of the
hunting algorithms), the iDivert softkey appears dimmed if the Use Legacy
Immediate Divert clusterwide service parameter is set to True; otherwise, it does
not appear dimmed.
For calls that do not reach the phone directly through a hunt list pilot (as part of
the hunting algorithms), the iDivert softkey does not appear dimmed when the
Use Legacy Immediate Divert clusterwide service parameter is set to True or
False.
Note For Jabber in desk phone mode, iDivert feature redirection to VM
is done through CTI application where ‘Use Legacy Immediate
Divert’ parameter will not take effect and HP number will be sent
as diversion info to Voice mail servers.
Auto Call Pickup If the Use Legacy Immediate Divert clusterwide service parameter is set to False,
and the Auto Call Pickup Enabled clusterwide service parameter is set to True,
and a user of call pickup group uses call pickup to answer a call, the IP phone
display will not present any choices to the user when the iDivert softkey is pressed.

Voice Mail Profile When you use QSIG integration with your voicemail system, a voicemail profile
that includes either a voicemail pilot or a voicemail mask or both should leave
the Make this the default Voice Mail Profile for the System check box
unchecked. Ensure the default Voice Mail Profile setting is always set to No
Voice Mail.
Call Forward All (CFA) When Call Forward All (CFA) and Call Forward Busy (CFB) are activated, the
and Call Forward Busy system does not support Immediate Divert (CFA and CFB have precedence over
(CFB) Immediate Divert).
212
Busy Voicemail System The iDivert detects a busy condition on the voicemail ports, when iDivert reaches
a voicemail system over a local or SCCP connection.
Note Immediate Divert cannot divert a call to a busy voicemail port;
voicemail ports can exist as members of a route or hunt list.
The call cannot divert to a busy voicemail system, but the original
call gets maintained. The phone displays “Busy” message on which
iDivert was invoked to indicate that the call was not diverted.
When a voicemail system is reached over a QSIG or SIP trunk,
iDivert can be detected, but the call does not get maintained. When
the Allow QSIG During iDivert clusterwide service parameter is
set to True, or the Use Legacy Immediate Divert clusterwide
service parameter is set to False, Immediate Divert supports access
to voicemail systems that can be reached over QSIG or SIP trunks.
When the Allow QSIG During iDivert clusterwide service
parameter is set to False, and the Use Legacy Immediate Divert
clusterwide service parameter is set to True, Immediate Divert
does not support access to voicemail systems over QSIG or SIP
trunks.
Malicious Caller ID System does not support using Malicious Caller ID and Immediate Divert features
together.
Forward No Answer A race condition in connection with the Forward No Answer Timeout exists when
Timeout you press the iDivert softkey. For example, if a manager presses the iDivert
softkey immediately after the Forward No Answer timeout, call forward forwards
the call to a preconfigured directory number. However, if the manager presses
the iDivert softkey before the Forward No Answer timeout, immediate divert
diverts the call to the voice-messaging mailbox of the manager.
Calling Parties and Called The calling parties and called parties can divert the call to their voice mailboxes
Parties if both simultaneously press the iDivert softkey.
Conference Types When one participant in a conference presses the iDivert softkey, all remaining
participants receive an outgoing greeting of the participant who pressed iDivert.
Conference types include Meet-Me, Ad Hoc, cBarge, and Join.
Split or Join Operation If the last action on a call was Auto Pickup, Call Transfer, Call Park, Call Park
Reversion, Conference, Meet-Me Conference, or any application that performs
a split or join operation, enhanced iDivert does not present a screen to a called
party to choose the voice mailbox. Instead, enhanced iDivert immediately diverts
the call to the voice mailbox that is associated with the called party.
213
Immediate Divert Troubleshooting
Immediate Divert Troubleshooting

Key is not active
The phone displays this message when the user presses iDivert:
Key is not active
The voice-messaging profile of the user who pressed iDivert does not have a voice-messaging pilot.
Configure a voice-messaging pilot in the user voice-messaging profile.
Temporary Failure
Temporary Failure
The voice-messaging system does not work, or a network problem exists.

Troubleshoot your voice-messaging system. See troubleshooting or voice-messaging documentation.
Busy
Busy
This message means that the voice-messaging system is busy.

Configure more voice-messaging ports or try again.
214
PA R T VIII
Conferencing Features
• Ad Hoc Conferencing , on page 217
• Meet-Me Conferencing , on page 229
• Conference Now , on page 235
CHAPTER 18
Ad Hoc Conferencing
• Ad Hoc Conferencing Overview, on page 217
• Ad Hoc Conferencing Task Flow, on page 217
• Conference Interactions, on page 225
• Conference Restrictions, on page 225
Ad Hoc Conferencing Overview

Ad Hoc conferences allow the conference controller (or in some cases, another participant) to add participants
to the conference.
Ad Hoc conferences comprise two types: basic and advanced. In basic ad hoc conferencing, the originator of
the conference acts as the controller of the conference and is the only participant who can add or remove other
participants. In advanced Ad Hoc conferencing, any participant can add or remove other participants. Advanced
Ad Hoc conferencing also allows you to link multiple ad hoc conferences together.
Advanced Ad Hoc conferencing allows you to link multiple Ad Hoc conferences together by adding an Ad
Hoc conference to another Ad Hoc conference as if it were an individual participant. If you attempt to link
multiple conferences together when the Advanced Ad Hoc Conference Enabled service parameter is set to
False, the IP phone displays a message. You can also use the methods that are available for adding individual
participants to an Ad Hoc conference to add another conference to an Ad Hoc conference.
Ad Hoc Conferencing Task Flow

Procedure

Step 1 Configure Softkey Template for Conferencing, Add the Conference List, Join, and Remove
on page 218 Last Conference Party softkeys to a softkey
template.
Step 2 To Associate Softkey Template Common Optional. To make the softkey template
Device, on page 219, complete the following available to phones, you must complete either
subtasks: this step or the following step. Follow this step
217
Configure Softkey Template for Conferencing

• Add a Softkey Template to a Common if your system uses a Common Device
Device Configuration, on page 220 Configuration to apply configuration options
• Associate a Common Device to phones. This is the most commonly used
Configuration with a Phone, on page 221 method for making a softkey template available
to phones.
Step 4 Configure Ad Hoc Conferencing, on page 221 Enable advanced conferencing, specify the
maximum number of participants, and specify
when to drop a conference connection.
Step 5 Configure Join Across Lines, on page 224 Enable Join Across Lines to create a conference.
Configure Softkey Template for Conferencing

Use this procedure to make the following conferencing softkeys available:
Softkey Description Call States
Conference List (ConfList) View a list of participant directory On Hook

numbers that are in an Ad Hoc
Connected
conference. The name of the
participant is displayed if it is
configured in Cisco Unified
Administration.
Join Join up to 15 established calls (for On Hold

a total of 16) to create a conference.
Remove Last Conference Party The conference controller can On Hook

(Remove) invoke the conference list and
Connected
remove any participant in the
conference by using the Remove
softkey.
218
Associate Softkey Template Common Device
Procedure
a) Click Add New.
d) Click Save.
template.
click Go.
to display.
Step 9 Click Save.
devices.
What to do next
Complete one of the following procedures:
• Associate Softkey Template Common Device, on page 219
• Associate a Softkey Template with a Phone, on page 221
Associate Softkey Template Common Device

219
Add a Softkey Template to a Common Device Configuration
To use the alternative method, go to Associate a Softkey Template with a Phone, on page 221
Before you begin

Configure Softkey Template for Conferencing, on page 218
Procedure

Step 1 Add a Softkey Template to a Common Device Perform this step to add a conferencing softkey
Configuration, on page 220 template to the Common Device Configuration.
Step 2 Associate a Common Device Configuration Perform this step to link the conferencing
with a Phone, on page 221 softkey Common Device Configuration to a
phone.
Procedure
a) Click Add New.
c) Click Save.
to make available.
Step 5 Click Save.
restart them.
220
Procedure

Step 4 Click Save.

Procedure

Step 4 Click Save.
Configure Ad Hoc Conferencing

Configure advanced Ad Hoc conferencing to allow non-controller participants to add and remove other
participants and the ability of all participants to link ad hoc conferences together.
Procedure
Step 2 From the Server drop-down list, choose the server.
Step 4 Configure the fields in the Clusterwide Parameters (Features - Conference) area. For parameter descriptions,
see Ad Hoc Conferencing Service Parameters, on page 222.
221
Ad Hoc Conferencing Service Parameters
Step 5 Click Save.
What to do next
Configure Join Across Lines, on page 224

The following table lists the main service parameters for Ad Hoc conferencing. For additional conferencing
service parameters, refer to the Service Parameter Configuration window's Advanced option. Conferencing
service parameters appear under Clusterwide Parameters (Feature - Conference).
Table 22: Ad Hoc Conference Service Parameters
Service Parameters Description
Drop Ad Hoc Conference Drop Ad Hoc Conference, prevents toll fraud (where an internal conference
controller disconnects from the conference while outside callers remain
connected). The service parameter settings specify conditions under which an ad
hoc conference gets dropped.
• Never—The conference does not get dropped. (We recommend that you
use the default option to avoid unintentional termination of a conference).
• When No OnNet Parties Remain in the Conference—The system drops
the active conference when the last on-network party in the conference hangs
up or drops out of the conference. Unified Communications Manager releases
all resources that are assigned to the conference.
Note Drop Ad Hoc Conference feature in an ILS deployment will
not drop the parties when it set at When No OnNet Parties
Remain in the Conference because the route patterns learned
are classified as On Net.
• When Conference Controller Leaves—The active conference terminates

when the primary controller (conference creator) hangs up. Unified
Communications Manager releases all resources that are assigned to the
conference.
Note We recommend that you set this service parameter to Never. Any
other setting can result in unintentional termination of a conference.
The Drop Ad Hoc Conference service parameter works differently
for conference calls that are initiated from a Cisco Unified IP Phone
7940 or 7960 that is running SIP, or a third-party phone that is
running SIP
.
Maximum Ad Hoc This parameter specifies the maximum number of participants that are allowed
Conference in a single Ad Hoc conference.
Default Value: 4
222
Advanced Ad Hoc This parameter determines whether advanced Ad Hoc conference features are
Conference Enabled enabled. This includes the ability of non-controller participants to add and remove
other participants and the ability of all participants to link ad hoc conferences
together.
Non-linear Ad Hoc This parameter determines whether more than two Ad Hoc conferences can be
Conference Linking linked directly to an Ad Hoc conference in a non-linear fashion ( three or more
Enabled conferences linked to any one conference).
Choose Encrypted Audio This parameter determines whether Unified Communications Manager chooses
Conference Instead Of an encrypted audio conference bridge or an unencrypted video conference bridge
Video Conference for an Ad-Hoc conference call when the conference controller's Device Security
Mode is set to either Authenticated or Encrypted and at least two conference
participants are video-capable. Because encrypted video conference bridges are
not supported in this release, Unified Communications Manager must choose
between an encrypted audio conference bridge and an unencrypted video
conference bridge. The default value is True.
Minimum Video Capable This parameter specifies the number of video-capable conference participants
Participants To Allocate that must be present in an Ad Hoc conference to allocate a video conference
Video Conference bridge. If the number of video-capable participants is less than the number
specified in this parameter, Unified Communications Manager allocates an audio
conference bridge. If the number of video-capable participants is equal to, or
greater than, the number specified in this parameter, Unified Communications
Manager allocates a video conference bridge, when available, from the configured
media resource group list (MRGL). Specifying a value of zero means that video
conference bridges will always be allocated, even when none of the participants
on the conference are video-capable. When a conference has been established
using an audio bridge and then additional video-capable participants join the
conference, the conference will remain on the audio bridge and will not convert
to video. The default value is 2.
Allocate Video This parameter determines whether Unified Communications Manager chooses
Conference Bridge For a video conference bridge, when available, for an Ad Hoc audio-only conference
Audio Only Conferences call when the video conference bridge has a higher priority than an audio
When The Video conference bridge in the media resource group list (MRGL). If an audio conference
Conference Bridge Has bridge has higher priority than any video conference bridge in the MRGL, Unified
Higher Priority Communications Manager ignores this parameter. This parameter proves useful
in situations where the local conference bridge is a video bridge (and configured
in the MRGL with the highest priority) and audio conference bridges are only
available in remote locations; in that situation, enabling this parameter means
thatUnified Communications Manager would attempt to use the local video
conference bridge first, even for audio-only conference calls. The default value
is False.
223
Configure Join Across Lines
Enable This parameter determines whether the Click-to-Conference functionality over

Click-to-Conference for the SIP trunk is enabled on Unified Communications Manager. The
Third-Party Applications Click-to-Conference feature allows third-party applications to setup a conference
using the SIP out of dialog REFER method and subscribe to the SIP trunk for
Conference Event Package through SIP SUBSCRIBE/NOTIFY.
Warning Enabling this parameter could negatively affect CTI applications
that are not coded to support this feature.
Cluster Conferencing This parameter defines a number, up to 8 digits (e.g. 0001), that is prefixed to a
Prefix Identifier conference identifier generated for Adhoc and Meet-Me conferences that will be
hosted on a SIP conference bridge such as Cisco Telepresence MCU or Cisco
Telepresence Conductor. This field should be populated by the administrator
when there are multiple clusters in a network that will be sharing the SIP
conference bridges that Unified Communications Manager manages. Every cluster
should be configured with a unique prefix to ensure that the conference identifier
for Adhoc and Meet-Me conferences is unique. If conference resources are not
being shared across clusters, then this field may not be populated.
Configure Join Across Lines

The Join Across Lines feature allows a user to join calls on multiple phone lines (either on different directory
numbers or on the same directory number but on different partitions) to create a conference.
Before you begin

• Ensure the phone model supports Join Across Lines Generate a Phone Feature List, on page 1
• Configure Ad Hoc Conferencing, on page 221
Procedure
Step 1 From Cisco Unified CM Administration, choose Device > Device Settings > Default Device Profile.
The Default Device Profile Configuration window is displayed.
Step 2 From the Device Profile Type drop-down list, choose the phone model.
Step 3 From the Device Protocol drop-down list, choose the relevant SCCP or SIP protocol.
Step 4 Set the Join Across Lines to On.
Step 5 Click Save.
224
Conference Interactions
Conference Interactions
Feature Interaction
Conference by Using Initiate a conference by pressing the cBarge softkey, or if the Single Button
cBarge cBarge feature is enabled, by pressing the shared-line button of the active call.
When cBarge is initiated, a barge call gets set up by using the shared conference
bridge, if available. The original call gets split and then joined at the conference
bridge. The call information for all parties gets changed to Conference.
The barged call becomes a conference call with the barge target device as the
conference controller. It can add more parties to the conference or can drop any
party.
When any party releases from the call, leaving only two parties in the conference,
the remaining two parties experience a brief interruption and then get reconnected
as a point-to-point call, which releases the shared conference resource.
Interaction with Call Park, If the conference controller transfers, parks, or redirects the conference to another
Call Transfer, and party, the party that retrieves the call acts as the virtual controller for the
Redirect conference. A virtual controller cannot add new parties to the conference nor
remove any party that was added to the conference, but a virtual controller can
transfer, park, or redirect the conference to another party, who would, in turn,
become the virtual controller of the conference. When this virtual controller hangs
up the call, the conference ends.
Softkey display on SIP The ConfList and the Remove softkey feature is available only on SCCP phones.
phones The SIP phones have a Show Details button with similar functionality.
Conference Restrictions
The following restrictions apply to ad hoc conferencing:
Feature Restrictions
Ad Hoc conference Unified Communications Manager supports a maximum of 100 simultaneous Ad

Hoc conferences for each Unified Communications Manager server.
Cisco Unified Communications Manager supports a maximum of 64 participants
per Ad Hoc conference (provided adequate conference resources are available).
In the case of linked Ad Hoc conferences, the system considers each conference
as one participant.
225
Ad Hoc conference on Unified Communications Manager uses“ beep” and “beepbeep” tones when a
SIP phones: new party is added and when the new party drops from the Ad Hoc conference,
respectively. When a party is added to an Ad Hoc conference, a user on a phone
that is running SIP may not hear the beep; when a participant drops from the Ad
Phone 7911
Hoc conference, a user on a phone that is running SIP may not hear the
• Cisco Unified IP “beepbeep”. Users might not hear the beeps because of the time it takes Unified
Phone 7941 Communications Manager to set up and tear down connections during the
conferencing process.
Phone 7961 You can invoke Ad Hoc conference linking for phones that are running SIP only
by using the Conference and Transfer functions. The system does not support
Direct Transfer and Join. Supported phones that are running SIP comprise Cisco
Unified IP Phone 7911, 7941, 7961.
Ad Hoc conference on • Phones display individual calls as conference calls. Cisco Unified IP Phones
SIP phones: 7940 and 7960 can create local conference calls but not Ad Hoc conference
calls.
Phone 7940 • Conference list (ConfList), is not available.
• Cisco Unified IP • Remove last conference participant (RmLstC), is not available.
Phone 7960
• Drop Ad Hoc conference is not supported.
• Third-Party Phone
• The SIP Profile parameter Conference Join Enabled controls behavior of
the phone that is running SIP when the conference controller exits a locally
hosted conference. If the Conference Join Enabled check box is unchecked,
all legs disconnect when the conference controller exits the Ad Hoc
conference call. If the Conference Join Enabled check box is checked, the
remaining two parties stay connected.
• To achieve the same level of control that the Drop Ad Hoc Conference
parameter settings provide for conference calls that a phone that is running
SCCP initiates, the administrator can use a combination of the Conference
Join Enabled SIP profile parameter and the Block OffNet to OffNet Transfer
service parameter for conferences that are initiated on the phone that is
running SIP (Cisco Unified IP Phone 7940 or 60). (Because the phone that
is running SIP performs a transfer when it drops out of the conference call,
the Block OffNet to OffNet Transfer can prevent toll fraud by not allowing
two offnet phones to remain in the call.)
• Unified Communications Manager uses “beep” and “beepbeep” tones when
a new party is added and when the new party drops from the Ad Hoc
conference, respectively. When a party is added to an Ad Hoc conference,
a user on a phone that is running SIP may not hear the beep when a
participant drops from the Ad Hoc conference, a user on a phone that is
running SIP may not hear the “beepbeep”. Users might not hear the beeps
because of the time it takes Unified Communications Manager to set up and
tear down connections during the conferencing process.
226
Phone displaying "To Configure a Call Manager cluster with Publisher (CmA11) and Subscribers
Conference" even when (CmA2).
two parties are connected
Phones A, B, C are registered with CmA1. Phones D is registered with CmA2.
• Setup an consultative or blind ad-hoc conference between A(1000), B(4000),
C(5000), D(6000) with A as the controller.
• Shutdown Cma2.
• Phone D will go to Preservation mode & press end call softkey .
• Phone A,B & C are in conference.
• Phone A,B & C are in conference.
• Disconnect Phone A ,then Phone B & C should be in a Direct call. Issue:
Phone B & C are still in conference
• Disconnect Phone A ,then Phone B & C should be in a Direct call. Issue:
Phone B & C are still in conference
• Disconnect Phone B, there should be no call on phone C. Phone B & C are
still in conference. Issue: Phone C is still in Conference .
227
228
CHAPTER 19
Meet-Me Conferencing
• Meet-Me Conferencing Overview, on page 229
• Meet-Me Conferencing Task Flow, on page 229
• Meet-Me Conferencing Restrictions, on page 234
Meet-Me Conferencing Overview

Users can use Meet-Me Conferencing to set up or join conferences. A user that sets up a conference is called
the conference controller. A user that joins a conference is called a participant.
Meet-Me Conferencing Task Flow

Before you begin
• Refer to the configuration documentation which came with your router and check for any settings which
you may need to configure before proceeding with the Meet-Me Conferencing Task Flow.
Procedure

Step 1 Configure a Softkey Template for Meet-Me Add the Meet-Me softkey to a softkey template.
Conferencing, on page 230
complete the following subtasks: this step or the following step.
• Add a Softkey Template to a Common
• Associate a Common Device
Configuration with a Phone, on page 232
Step 3 Common Device ConfigurationAssociate a Optional. Use this procedure either as an

Softkey Template with a Phone, on page 232 alternative to associating the softkey template
229
Configure a Softkey Template for Meet-Me Conferencing

Step 4 Configure a Meet-Me Conferencing Number, Enable advanced conferencing, specify the
on page 233 maximum number of participants, and specify
when to drop a conference connection.
Configure a Softkey Template for Meet-Me Conferencing

Use this procedure to make the Meet Me softkey available in the off hook call state.
Procedure
a) Click Add New.
d) Click Save.
template.
click Go.
to display.
Step 9 Click Save.
devices.
230

To use the alternative method, see Associate a Softkey Template with a Phone, on page 232.
Before you begin

Configure a Softkey Template for Meet-Me Conferencing, on page 230
Procedure

Step 1 Add a Softkey Template to a Common Device
Procedure
a) Click Add New.
c) Click Save.
231
to make available.
Step 5 Click Save.
restart them.
Before you begin

Add a Softkey Template to a Common Device Configuration, on page 231
Procedure

Step 4 Click Save.

Before you begin

Configure a Softkey Template for Meet-Me Conferencing, on page 230
Procedure

Step 4 Click Save.
232
Configure a Meet-Me Conferencing Number
Configure a Meet-Me Conferencing Number

The Cisco Unified Communications Manager administrator provides the Meet-Me conference directory
number range to users, so that they can access the feature. The user chooses a directory number from the range
that is specified for the Meet-Me Number or Pattern to establish a Meet-Me conference and becomes the
conference controller.
Procedure
Step 1 From Cisco Unified CM Administration, choose Call Routing > Meet-Me Number/Pattern.
The Find and List Meet-Me Numbers window appears.
Step 2 Enter the appropriate search criteria and click Find.
All matching records are displayed.
Step 3 In the list of records, click the link for the record that you want to view.
• To copy a Meet-Me number or pattern, click the Meet-Me number or pattern that you want to copy. The
Meet-Me Number/Pattern Configuration window appears. Click Copy.
• To add a Meet-Me Number or Pattern, click the Add New button.
• To update an existing Meet-Me Number or Pattern, click the Meet-Me Number or Pattern that you want
to update.
Step 5 Enter the appropriate settings.

See the Related Topics section for information about the fields and their configuration options.
Step 6 Click Save.
Meet-Me Number and Pattern Settings

Field Description
Directory Number or Enter a Meet-Me number or a range of numbers.

Pattern
To configure a range, the dash must appear within brackets and follow a digit;
for example, to configure the range 1000 to 1050, enter 10[0-5]0.
Description The description can include up to 50 characters in any language, but it cannot
include double quotation marks ("), percentage sign (%), ampersand (&), or angle
brackets (<>).
233
Meet-Me Conferencing Restrictions
Field Description
Partition To use a partition to restrict access to the Meet-Me number or pattern, choose
the desired partition from the drop-down list box.
If you do not want to restrict access to the Meet-Me number or pattern, choose
<None> for the partition.
You can configure the number of partitions that are displayed in this drop-down
list box by using the Max List Box Items enterprise parameter. If more partitions
exist than the Max List Box Items enterprise parameter specifies, the Find button
is displayed next to the drop-down list box. Click the Find button to display the
Find and List Partitions window.
Note To set the maximum list box items, choose System > Enterprise
Parameters and update the Max List Box Items field under
CCMAdmin Parameters.
Note Make sure that the combination of Meet-Me number or pattern and
partition is unique within the Unified Communications Manager
cluster.
Minimum Security Level Choose the minimum Meet-Me conference security level for this Meet-Me number
or pattern from the drop-down list box.
• Choose Authenticated to block participants with nonsecure phones from
joining the conference.
• Choose Encrypted to block participants with authenticated or nonsecure
phones from joining the conference.
• Choose Non Secure to allow all participants to join the conference.
Note To use this feature, ensure that you have a secure conference bridge
that is configured and available.
Meet-Me Conferencing Restrictions

Unified Communications Manager supports a maximum of 100 simultaneous Meet-Me conferences for each
Unified Communications Manager server.
After the maximum number of participants that is specified for that conference is has been exceeded, no other
callers can join the conference.
234
CHAPTER 20
Conference Now
• Conference Now Overview, on page 235
• Conference Now Prerequisites, on page 235
• Activate Cisco IP Voice Media Streaming, on page 236
• Configure Conference Now Settings, on page 236
• Enable Conference Now for User, on page 237
• Enable Conference Now via LDAP, on page 237
• Conference Now Interactions, on page 238
• Conference Now Restrictions, on page 239
Conference Now Overview

Conference Now provides a basic audio conferencing solution for small business customers that allows internal
and external callers to join a conference via a centralized IVR.
To host a meeting, configured users must configure a meeting PIN that they will need to enter, along with the
meeting number, when starting the meeting. The host provides the other meeting participants with the relevant
meeting information, including the time slot, meeting number (which is usually the host’s primary extension)
and an optional access code for secure conferences. At the designated time, the other participants can join the
call by dialing the IVR, and entering the meeting information at the prompts.
Administrators must configure end users with the ability to host Conference Now conferences. After the
feature is configured, meeting hosts can edit their meeting access code from within the Self-Care Portal.
Note Cisco recommends that you use IPVMS software-based conference bridges for Conference Now. If you use
other conference bridges, the conference entry and exit tones may not play to participants.
Conference Now Prerequisites

To use Conference Now you must make sure that the following media resources are configured, and are
available to the devices that will be initiating conferences.
235
Activate Cisco IP Voice Media Streaming
• Conference Bridge—For the best user experience, we recommend using a software-based Cisco IPVMS
conference bridge. Using another conference bridge might not provide the conference party entry and
exit tone.
• Interactive Voice Response (IVR)
After you configure these resources, you can make them available to devices by configuring a media resource
group list that includes these resources and then associating that media resource group list to the device pools
that will be used by your devices, or to individual devices. For more information on configuring Conference
Bridges, Interactive Voice Response, and Media Resource Groups, see "Configure Media Resources" section
of the System Configuration Guide for Cisco Unified Communications Manager.

The Cisco IP Voice Media Streaming Service must be running in order to use IVR services and Conference
Now.
Procedure
Step 2 From the Server drop-down list, choose the Cisco Unified Communications Manager publisher node.
Step 3 If the Cisco IP Voice Media Streaming application is deactivated, check the corresponding check box and
click Save.
Configure Conference Now Settings

Use this procedure to configure Conference Now system settings on Unified Communications Manager.
Procedure
Step 1 From Cisco Unified CM Administration, choose Call Routing > Conference Now.
Step 2 In the Conference Now IVR Directory Number field, enter a DID (Direct Inward Dial) number for a Unified
Communications Manager cluster to provide access for external callers.
Step 3 From the Route Partition drop-down list, select a partition.
Note The combination of the number and the partition must be unique within a cluster.
Step 4 Complete the remaining fields in the Conference Now Configuration window. For more information on the
Step 5 Click Save.
236
Enable Conference Now for User
What to do next
Enable the feature for end users:
• If you haven't yet synced your LDAP directory, add Conference Now to your LDAP sync, so that newly
synced users will be able to host Conference Now meetings. See Enable Conference Now via LDAP, on
page 237.
• To enable the feature for an existing end user, see Enable Conference Now for User, on page 237.
Enable Conference Now for User

Use this procedure to configure an existing end user with the ability to host Conference Now meetings.
Note You can also use Bulk Administration's Update Users feature to enable Conference Now for a large number
of users via a csv file. You must ensure that the same settings as in the below task are configured. For more
information, on how to use Update Users, see Bulk Administration Guide for Cisco Unified Communications
Manager.
Procedure
Step 2 Click Find and select the user for whom you want to add Conference Now.
Step 3 Under Conference Now, check the Enable End User to Host Conference Now check box.
Step 4 (Optional) For secure conferencing, enter an Attendees Access Code. Note that end users will be able to
modify their access code setting within the Self-Care Portal.
Note If the user has a Self-Service User ID assigned, the Conference Now Meeting Number
prepopulates with the value of the Self-Service User ID, which defaults to the user's primary
extension.
Step 5 Complete any remaining fields within the End User Configuration window. For more information on the
Step 6 Click Save.
Enable Conference Now via LDAP

If you have not yet synced your LDAP directory, you can enable Conference Now for synced users by adding
the option to a feature group template and then adding that feature group template to the initial LDAP sync.
New users provisioned via the LDAP sync will have Conference Now enabled.
237
Conference Now Interactions
Note You cannot apply feature group template edits to an LDAP directory sync where the initial sync has already
occurred. To apply these edits to an LDAP sync, the initial sync must not yet have occurred.
Procedure
Step 1 From Cisco Unified CM Administration, choose User Management > User Phone/Add > Feature Group
Template.
• Select Find and select an existing template.
• Click Add New to create a new template.
Step 3 From the drop-down list, select a Service Profile.

Step 4 From the drop-down list, select a User Profile.
Step 5 Check the Enable End User to Host Conference Now check box.
Step 6 Click Save.
What to do next
Assign the template to an LDAP directory sync so that synced users will be configured for Conference Now.
For more information on configuring an LDAP sync, see the "Configure End Users" section of the System
Alternatively, you can add a new user with Conference Now functionality via the Quick User/Phone Add
menu. You would need to add a new user that uses this feature group template in addition to assigning a
primary extension.
Conference Now Interactions

Feature Interactions
Mobility EFA (Enterprise A mobility user dials an Enterprise Feature Access DID number from a remote
Feature Access) destination. After the call is connected, the remote destination phone is used to
send DTMF digits to Unified Communications Manager via the PSTN gateway.
The user PIN followed by the # key is first authenticated with Unified
Communications Manager. After the user PIN authentication is successful, press
1 and the # key, to indicate a two-stage dialed call, followed by the desired phone
number. If the dialed phone number is a Conference Now IVR Directory Number
and the user is a meeting host, then the user must enter the PIN again.
238
Conference Now Restrictions
Mobility MVA (Mobile A call is directed to Unified Communications Manager through the enterprise
Voice Access) PSTN H.323 or SIP gateway. The IVR prompts the user to enter the User ID, #
key, PIN, # key, number 1 (to make a Mobile Voice Access call) and then the
desired phone number. If the phone number is a Conference Now IVR Directory
Number and the user is a meeting host, then the user must enter the PIN again.
Note Users are not prompted for entering their PIN if they dial directly
from their remote destination. However, if they dial from a different
phone to Mobile Voice Access Directory Number, then they are
prompted to enter PIN before they can make the call. If the users
call Conference Now IVR Directory Number, they are prompted to
enter the PIN again.

The Conference Now feature has the following restrictions:
• The host cannot mute attendees.
• The attendee cannot mute the audio by entering DTMF digits.
• The list of Conference Now participants is not supported.
• Maximum number of participants in a conference is controlled by the existing CallManager service
parameter "Maximum MeetMe Conference Unicast". It applies to both internal and external callers.
• Maximum number of simultaneous Conference Now and MeetMe conference instances combined together
is 100 per Unified Communications Manager CallManager node.
• Video on hold is not supported.
• The IPVMS software conference bridge only supports codec G.711 (ALaw & ULaw) and Wide Band
256k. If there is a codec mismatch between the calling device and the software conference bridge, a
transcoder will be allocated.
• Ensure that at least one of the following conditions are met to play the conference party entry and exit
tone:
• At least one conference participant is using the Cisco IP Phone.
• IPVMS is the allocated software conference bridge.
• When the sets up a Conference Bridge, the conference will continue with the remaining attendees
irrespective whether the host is present or not. If the host wants to rejoin the conference, an announcement
to enter the Attendee Access Code is played if it is configured by host. The host cannot schedule or mute
attendees; therefore, the host status is no longer valid.
• No audio announcement will play if the host is the first person to join the conference. However, when
the host dials into Conference Now from an internal IP Phone, there is a visual display on the IP Phone
showing “To Conference”.
239
Note If the host joins the Conference Now from any external phone, then there will be
no visual display on the phone.
240
PA R T IX
Placing Calls
• Call Back , on page 243
• Hotline , on page 255
• Speed Dial and Abbreviated Dial, on page 269
• WebDialer , on page 273
• Paging , on page 289
• Intercom , on page 309
CHAPTER 21
Call Back
• Call Back Overview, on page 243
• Call Back Prerequisites, on page 243
• Call Back Configuration Task Flow, on page 244
• Call Back Interactions, on page 249
• Call Back Restrictions, on page 250
• Call Back Troubleshooting, on page 251
Call Back Overview

The CallBack feature allows you to receive notification when a busy extension is available to receive calls.
You can activate Call Back for a destination phone that is within the same Unified Communications Manager
cluster as your phone or on a remote Private Integrated Network Exchange (PINX) over QSIG trunks or
QSIG-enabled intercluster trunks.
To receive CallBack notification, press the CallBack softkey or feature button while receiving a busy or
ringback tone. You can activate Call Back during reorder tone, which is triggered when the No Answer timer
expires.
Suspend/Resume
The Call Back feature enables the system to suspend the call completion service if the user who originated
Call Back is busy. When the originating user then becomes available, the call completion service resumes for
that user.
Note Call Back supports Suspend/Resume CallBack notification for both intracluster and intercluster QSIG trunks
or QSIG-enabled intercluster trunks.
Call Back Prerequisites

To use the Call Back feature, the destination phone must be in one of the following locations:
• In the same Unified Communications Manager cluster as the user phone
243
Placing Calls
Call Back Configuration Task Flow
• On a remote PINX over QSIG trunks

• On a remote PINX over QSIG-enabled intercluster trunks
If you want to use non-English phone locales or country-specific tones, you must install locales.
• The following devices support the Call Back feature:
• Cisco Unified IP Phones 6900, 7900, 8900, and 9900 Series (except 6901 and 6911)
• Cisco IP Phones 7800 and 8800 Series
• Cisco VGC Phone (uses the Cisco VG248 Gateway)
• Cisco Analog Telephone Adapter (ATA) 186 and 188
• Busy Subscriber for Cisco VG224 endpoints
• No Answer for Cisco VG224 endpoints
• A CTI route point that forwards calls to any of the supported phones.
Call Back Configuration Task Flow

Complete one of the task flows depending on whether your phone supports softkey or buttons.
Use this table to determine whether to configure the CallBack softkey or the button for the Call Back supported
IP phones.
Table 23: Cisco IP Phones That Use CallBack Softkeys and Buttons
Cisco Phone Model CallBack Softkey CallBack Button
Cisco Unified IP Phone 6900 Series (except X X

6901 and 6911)
Cisco Unified IP Phone 7900 Series X
Cisco IP Phone 7800 and 8800 Series X X
Cisco Unified IP Phone 8900 Series X X
Cisco Unified IP Phone 9900 Series X X
Cisco IP Communicator X
Before you begin

• Review Call Back Prerequisites, on page 243.
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Placing Calls
Configure Softkey Template for CallBack
Procedure

Step 1 Configure Softkey Template for CallBack, on Perform this step to add CallBack softkey to
page 245 template and configure the softkey using the
Common Device Configuration or phone.
Step 2 Configure CallBack Button, on page 248 Perform this step to add and configure the
CallBack button to a phone.
Configure Softkey Template for CallBack

CallBack softkey has the following call states:
• On Hook
• Ring Out
• Connected Transfer
Use this procedure to make the CallBack softkey available:
Before you begin

Ensure your phone supports Call Back.
Procedure
a) Click Add New.
d) Click Save.
template.
click Go.
to display.
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Placing Calls
Associate CallBack Softkey Template with a Common Device Configuration
Step 9 Click Save.
devices.
What to do next
Perform one the following procedures:
• Associate CallBack Softkey Template with a Common Device Configuration, on page 246
• Associate CallBack Softkey Template with Phone, on page 247
Associate CallBack Softkey Template with a Common Device Configuration

To use the alternative method, see Associate CallBack Softkey Template with Phone, on page 247.
Procedure

Step 1 Add CallBack Softkey Template to the Perform this step to add CallBack softkey
Common Device Configuration, on page 246 template to the Common Device Configuration.
Step 2 Associate a Common Device Configuration Perform this step to link the CallBack softkey
with a Phone, on page 247 Common Device Configuration to a phone.
Add CallBack Softkey Template to the Common Device Configuration
Procedure
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Placing Calls
a) Click Add New.
c) Click Save.
to make available.
Step 5 Click Save.
restart them.
Procedure

Step 4 Click Save.
Associate CallBack Softkey Template with Phone

Optional: Use this procedure either as an alternative to associating the softkey template with the Common
Device Configuration, or in conjunction with the Common Device Configuration. Use this procedure in
conjunction with the Common Device Configuration if you need to assign a softkey template that overrides
the assignment in the Common Device Configuration or any other default softkey assignment.
Procedure

Step 4 Click Save.
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Placing Calls
Configure CallBack Button
Configure CallBack Button

The procedures in this section describe how to configure the CallBack button.
Procedure

Step 1 Configure Phone Button Template for Call Perform this step to assign CallBack button
Back, on page 248 features to line or speed dial keys.
Step 2 Associate a Button Template with a Phone, on Perform this step to configure the CallBack
page 249 button for a phone.
Configure Phone Button Template for Call Back

Follow this procedure when you want to assign features to line or speed dial keys.
Procedure
step.
c) Click Save.
Step 6 Click Save.
devices.
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Placing Calls
Associate a Button Template with a Phone
Procedure

button.
Step 5 Click Save.
Call Back Interactions

Feature Interaction
Call Forward Calls that are made from CallBack notification screen will override
all the Call Forward configured values on the target DN. The calls
should be made before CallBack recall timer expires otherwise the
calls will not override the Call Forward configured values.
CallBack notification with phones CallBack notification works differently only for Cisco Unified IP
running SIP Phones 7960 and 7940. All other SIP phones and all SCCP phones
support on-hook and off-hook notification.
The only way that Unified Communications Manager knows when
a line on a SIP 7960 or 7940 phone becomes available is by
monitoring an incoming SIP INVITE message that Unified
Communications Manager receives from the phone. After the phone
sends the SIP INVITE to Unified Communications Manager and
the phone goes on-hook, Unified Communications Manager sends
an audio and CallBack notification screen to the Cisco Unified IP
Phone 7960 and 7940 (SIP) user.
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Placing Calls
Call Back Restrictions
Feature Interaction
Do Not Disturb (DND) CallBack would work normally in case or when DND-Reject is set
to Off at the originating or the terminating end. The behavior differs
only when DND-Reject is set to On.
• DND-Reject On on Originating end—User A calls User B
and invokes Call Back. User A goes on DND-R. After User B
is available, the CallBack notification will still be displayed to
User A. That is, user will still be notified with the availability
of the other party irrespective of the DND status.
• DND-Reject On on Terminating end—User A calls User B,
and User B has set DND-Reject to On. User A will get a fast
busy tone. User A can initiate CallBack on a busy endpoint. If
User B is still on DND-Reject and goes Offhook and Onhook,
User A will get a notification “User B is available now but on
DND-R”, and it will not show the Dial option. If User A does
not choose to cancel, CallBack will still monitor User B until
User B sets DND-Reject to Off.
Cisco Extension Mobility When a Cisco Extension Mobility user logs in or logs out, any active
call completion that is associated with Call Back is automatically
canceled. If a called phone is removed from the system after Call
Back is activated on the phone, the caller receives a reorder tone
after pressing the Dial softkey. The user may cancel or reactivate
Call Back.
Call Back Restrictions

Feature Restriction
Call Back with video The Call Back feature does not work for video calls when the call is placed
across CUBE between two Unified CM clusters that are connected via CUBE with qsig-enabled
SIP trunks. For additional detail, see CSCun46243.
SIP Trunks Call Back is not supported over SIP trunks but is supported over QSIG-enabled
SIP trunks.
Supported characters for Call Back only supports spaces and digits 0 through 9 for the name or number of
name or number of the calling or called party. To work with CallBack, the name or number of the
calling or called party calling or called party cannot contain a pound sign (#) or asterisk (*).
Voicemail You cannot activate Call Back if you forward all calls to Voice-Messaging System.
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Call Back Troubleshooting
Call Back Troubleshooting

This section describes the problems, possible causes, and solutions for various scenarios, and error messages
that are displayed on the IP phone for Call Back.
Unplug/Reset Phone After Pressing CallBack Softkey but Before CallBack

Occurs
Problem
You have unplugged or reset the phone after pressing the CallBack Softkey but before activating CallBack.
Possible Cause
Unified Communications Manager cancels the Call Back activation.
Solution
After the caller phone registers, the caller phone does not display the Call Back activation window after the
reset. The caller must press the CallBack Softkey to view the active Call Back service. CallBack notification
occurs on the phone.
Caller Misses to View Availability Notification Before Phone Reset

Problem
In an intracluster or intercluster Call Back scenario, a caller initiates Call Back for a user, for example, User
B, who is unavailable. When User B becomes available, the availability notification screen displays on the
caller phone, and a tone plays. The caller misses the availability notification for some reason, and the phone
resets.
The caller contacts a different user, User C, for example, and presses the CallBack softkey because User C
appears busy. The replace/retain screen displays on the caller phone, but the screen does not state that the
availability notification already occurred for User B.
Possible Cause
The user reset the phone.
Solution
After a phone reset but not during an active call, review the Call Back notifications on the phone. Press the
CallBack softkey.
Call Back Error Messages

The following section describes the error messages that display on the IP phone screen.
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Placing Calls
CallBack Is Not Active
CallBack Is Not Active
Problem
The following error message is displayed:
CallBack is not active. Press Exit to quit this screen.
Possible Cause
User pressed the CallBack softkey during the idle state.
Solution
Follow the recommended action provided in the error message.
CallBack Is Already Active
Problem
CallBack is already active on xxxx. Press OK to activate on yyyy. Press Exit to quit this
screen.
Possible Cause
A user tried to activate Call Back, but it is already active.
Problem
Follow the recommended action provided in the error message.
CallBack Cannot Be Activated
Problem
CallBack cannot be activated for xxxx.
Possible Cause
When a user tried to activate Call Back, either the extension is not available in Unified Communications
Manager database or there is no QSIG route to the destination (that is, the extension belongs to remote Proxy
which is connected via non-QSIG trunk), and the extension is not found in the database.
Solution
The user must try again, or the administrator must add the directory number to the Cisco Unified CM
Administration.
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Key Not Active
Key Not Active
Problem
During a call, the CallBack softkey displays on the phone and the user presses the CallBack softkey before
the phone rings. But, the following error message is displayed on the phone:
Key Not Active
Possible Cause
User may not be pressing the CallBack softkey at the appropriate time.
Solution
Users must press the CallBack softkey after a ringing or busy signal is received. Pressing the softkey at the
wrong time may cause an error message to display on the phone.
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Key Not Active
254
CHAPTER 22
Hotline
• Hotline Overview, on page 255
• System Requirements for Hotline, on page 256
• Hotline Configuration Task Flow, on page 256
• Hotline Troubleshooting, on page 266
Hotline Overview
The Hotline feature extends the Private Line Automatic Ringdown (PLAR) feature, which allows you to
configure a phone so that when the user goes off hook (or the NewCall softkey or Line Key gets pressed), the
phone immediately dials a preconfigured number. The phone user cannot dial any other number from a phone
that is configured for PLAR
Hotline adds the following additional restrictions and administrator controls for phones that use PLAR:
• Hotline devices (devices configured to use hotline) that receive calls will receive calls only from other
hotline devices, and will reject non-hotline callers.
• You can configure a Hotline phone to call only, receive only, or both call and receive.
• You can restrict the features available on a Hotline phone by applying a softkey template to the phone.
• Analog hotline phones ignore inbound hookflash signals.
Route Class Signaling

Hotline uses route class signaling to allow Hotline phones to receive calls only from other Hotline phones. A
route class is a DSN code that identifies the class of traffic for a call. The route class informs downstream
devices about special routing or termination requirements. A Hotline phone can only accept calls from a
Hotline phone with the same route class.
Call Screening
Hotline also provides Configurable Call Screening based on caller ID. Configurable Call Screening allows a
receiving Hotline phone to screen calls based on caller ID information and allow only callers in a screening
list to connect.
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Placing Calls
System Requirements for Hotline
System Requirements for Hotline

The following hotline system requirements exist for Unified Communications Manager:
• Unified Communications Manager 8.0(1) or higher on each server in the cluster
• MGCP gateway POTS phones (FXS).
• SCCP gateway POTS phones (FXS).
Tip Cisco Feature Navigator allows you to determine which Cisco IOS and Catalyst OS software images support
a specific software release, feature set, or platform. To access Cisco Feature Navigator, go to
http://cfn.cloudapps.cisco.com/ITDIT/CFN/.
You do not need a Cisco.com account to access Cisco Feature Navigator.
Hotline Configuration Task Flow

Procedure

Step 1 Generate a Phone Feature List, on page 1 Log in to Cisco Unified Reporting and run a
phone feature list report to determine which
phones support Hotline.
Step 2 Create Custom Softkey Template, on page 256 Optional. If you want to restrict features on a
Hotline phone, create a softkey template that
allows only the features that you want.
Step 3 Configure Hotline on Phones, on page 257 Enable the phone as a Hotline device.
Step 4 Configure Route Class Signaling Task Flow, Configure route class signaling to support the
on page 258 Hotline feature.
Step 5 Configure Hotline to Call Only or Receive Only Optional. If you want to restrict a Hotline
Task Flow, on page 262 phone to either originating calls only or
terminating calls only, configure call and
receive settings.
Step 6 Configure Call Screening with a Calling Search Optional. Use calling search spaces and
Space , on page 264 partitions to configure a call screening list for
your Hotline phones.
Create Custom Softkey Template

When configuring Hotline, you can customize a softkey template to display only those features that you want
to make available to a Hotline phone.
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Placing Calls
Configure Hotline on Phones
Unified Communications Manager includes standard softkey templates for call processing and applications.
When creating custom softkey templates, copy the standard templates and make modifications as required.
Before you begin

Procedure
Step 1 Choose Device > Device Settings > Softkey Template.

Step 3 From the drop-down list, select a softkey template and click Copy to create a new template.
Step 4 In the Softkey Template Name field, enter a unique name to identify the softkey template.
Step 5 Enter a description that describes the use of the template. The description can include up to 50 characters in
any language, but it cannot include double-quotes ("), percentage sign (%), ampersand (&), backslash (\), or
angle brackets (<>).
Step 6 To designate this softkey template as the standard softkey template, check the Default Softkey Template
check box.
Note If you designate a softkey template as the default softkey template, you will not be able to delete
this softkey template unless you first remove the default designation.
Step 7 Click Save.

The softkey template gets copied, and the Softkey Template Configuration window redisplays.
Step 8 (Optional) Click the Add Application button.

Step 9 Configure the positions of the softkeys on the Cisco Unified IP Phone LCD screen.
Step 10 To save your configuration, click Save.
Configure Hotline on Phones

Use this procedure to enable the phone as a Hotline device.
Before you begin

Optional. If you want to create a custom softkey template to display only those features that you want to make
available to a Hotline phone, see Create Custom Softkey Template, on page 256.
Procedure

Step 2 Click Find and select the phone that you want to enable as a Hotline device.
Step 3 Check the Hotline Device check box.
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Configure Route Class Signaling Task Flow
Step 4 If you have created a custom softkey template specifically for the Hotline phone, from the Softkey Template
drop-down list, choose the softkey template.
Step 5 Click Save.
Note You can also assign a softkey template to a Device Pool and then assign that Device Pool to the
phone.
Configure Route Class Signaling Task Flow

Perform this task flow to configure route class signaling for Hotline calls.
Procedure

Step 1 Enable Route Class Signaling in the Cluster, on Set the route class signaling clusterwide defaults
page 258 for trunks and gateways to enabled.
Note The settings for individual trunks
and gateways override the
clusterwide defaults. If you use
this service parameter to enable
route class signaling across the
cluster, route class signaling can
still be disabled on an individual
trunk or gateway.
Step 2 Enable Route Class Signaling on Trunks, on Enable route class signaling on an individual
page 259 trunk.
Step 3 Enable Route Class Signaling on Gateways, on Enable route class signaling on an MGCP
page 259 T1/CAS or MGCP PRI gateway.
Step 4 Configure Signaling Labels for the Hotline Configure SIP signaling labels for Hotline route
Route Class, on page 260 classes.
Step 5 Configure the Route Class on Hotline Route Configure the route class on the route patterns
Patterns, on page 261 that are routing your Hotline calls.
Step 6 Configure the Route Class on Hotline Optional. If you use translation patterns on your
Translation Patterns, on page 261 Hotline calls, configure the route class on your
translation patterns.
Enable Route Class Signaling in the Cluster

When you set the Route Class Trunk Signaling Enabled service parameter to True, the default route class
signaling setting for all trunks or gateways in the cluster that support route class signaling is set to enabled.
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Enable Route Class Signaling on Trunks
Note The settings for individual trunks and gateways override the clusterwide defaults. If you use this service
parameter to enable route class signaling across the cluster, route class signaling can still be disabled on an
individual trunk or gateway.
Procedure
Step 2 Set the Route Class Trunk Signaling Enabled service parameter to True.
Step 3 Click Save.
What to do next
Use the following procedures to configure route class signaling on individual trunks or gateways.
Enable Route Class Signaling on Trunks, on page 259
Enable Route Class Signaling on Gateways, on page 259
Enable Route Class Signaling on Trunks

Use this procedure to enable route class signaling on an individual trunk. The configuration for individual
trunks overrides the clusterwide service parameter setting.
Before you begin

Follow the Enable Route Class Signaling in the Cluster, on page 258 procedure to use a clusterwide service
parameter to configure the default route class signaling settings for all trunks in the cluster.
Procedure
Step 1 From Cisco Unified CM Administration, choose Device > Trunks.

Step 2 Click Find and select the SIP trunk on which you want to enable route class signaling.
Step 3 From the Route Class Signaling Enabled drop-down list box, choose one of the following options:
• Default—This trunk uses the setting from the Route Class Signaling Enabled service parameter.
• Off—Route class signaling is disabled for this trunk.
• On—Route class signaling is enabled for this trunk.
Step 4 Click Save.
Enable Route Class Signaling on Gateways

Use this procedure to enable route class signaling on an individual MGCP PRI or MGCP T1/CAS gateway.
The configuration for individual gateways overrides the clusterwide service parameter setting.
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Configure Signaling Labels for the Hotline Route Class
Before you begin

Follow the Enable Route Class Signaling in the Cluster, on page 258 procedure to use a clusterwide service
parameter to set the default route class signaling setting for gateways in the cluster.
Perform the Enable Route Class Signaling on Trunks, on page 259 procedure to configure route class signaling
for individual trunks.
Procedure
Step 1 From Cisco Unified CM Administration, choose Device > Gateways.

Step 2 Click Find and select the gateway on which you want to configure route class signaling.
Step 3 From the Route Class Signaling Enabled drop-down list box, choose one of the following options:
• Default—This gateway uses the setting from the clusterwide Route Class Signaling Enabled service
parameter.
• Off—Route class signaling is disabled on this gateway.
• On—Route class signaling is enabled on this gateway.
Step 4 If you want to encode voice route class for voice calls, check the Encode Voice Route Class check box.
Step 5 Click Save.
Configure Signaling Labels for the Hotline Route Class

You must configure a SIP signaling label value for the Hotline route class that you want to use.
Before you begin

Enable route class signaling on your trunks and gateways. For details, see Enable Route Class Signaling in
the Cluster, on page 258.
Procedure
Step 5 In the SIP Route Class Naming Authority service parameter field, enter a value to represent the naming
authority and context for the labels used in SIP signaling to represent route class. The default value is cisco.com.
Step 6 In the SIP Hotline Voice Route Class Label service parameter field, enter a label to represent the Hotline
Voice route class. The default value is hotline.
Step 7 In the SIP Hotline Data Route Class Label service parameter field, enter a lable to represent the Hotline
Data route class. The default value is ccdata.
Step 8 Click Save.
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Configure the Route Class on Hotline Route Patterns
Configure the Route Class on Hotline Route Patterns

This procedure describes call routing instructions that are specific to Hotline devices. For more information
on how to configure route patterns and translation patterns in your network, see the System Configuration
Guide for Cisco Unified Communications Manager.
For each route pattern that you expect to route a Hotline call, you must set the route class for that route pattern
to Hotline Voice or Hotline Data.
Before you begin

Configure Signaling Labels for the Hotline Route Class, on page 260
Before you perform this procedure, it is expected that your network call routing is set up with route patterns.
Procedure
Step 1 From Cisco Unified CM Administration, choose Call Routing > Route/Hunt > Route Patterns.
Step 2 Click Find to display a list of route patterns in your network.
Step 3 For each T1/CAS route pattern that is used to route a Hotline call:
a) From the Find and List Route Patterns window, select the route pattern.
b) From the Route Class drop-down list box, choose either Hotline Voice or Hotline Data as the route class
for this route pattern.
c) Click Save.
Configure the Route Class on Hotline Translation Patterns
Before you begin

Before you perform this procedure, it is expected that you have set up network call routing with route patterns
and translation patterns.
Perform the Configure the Route Class on Hotline Route Patterns, on page 261 procedure.
Procedure
Step 1 From Cisco Unified CM Administration, choose Call Routing > Translation Pattern.
Step 2 Click Find to display the translation patterns in your cluster.
Step 3 For each translation pattern that you want to use on a Hotline number, perform the following steps:
a) From the Route Class drop-down list box, select either Hotline Voice or Hotline Data.
b) Click Save.
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Placing Calls
Configure Hotline to Call Only or Receive Only Task Flow
Configure Hotline to Call Only or Receive Only Task Flow

The configuration example in this task flow describes how to set up a Hotline phone to either place calls only
or receive calls only.
Procedure

Step 1 Configure Partitions for Hotline Call Only Create two partitions: one should be empty and
Receive Only, on page 262 the other will be assigned to a new CSS.
Step 2 Configure Calling Search Space for Hotline Create a new calling search space and assign
Call Only Receive Only, on page 262 one of the new partitions to this CSS. This CSS
will contain no other partition.
Step 3 Perform one of the following procedures: If you want to configure call only, assign the
empty partition to the phone line. If you want
• Configure Call Only on Hotline Phone, on
to configure receive only, assign the new CSS
page 263
to the phone.
• Configure Receive Only on Hotline Phone,
on page 263
Configure Partitions for Hotline Call Only Receive Only

If you want to configure a Hotline phone to either place calls only, or to receive calls only you must create
two partitions.
Procedure
Step 1 From Cisco Unified CM Administration, choose Call Routing > Class of Control > Partitions.
Step 3 Create a new partition.
Step 4 Enter a unique name and description for the partition. For example, IsolatedPartition.
Note This partition will not be assigned to any CSS.
Step 5 Click Save

Step 6 Repeat steps 2-5 and create a second partition. For example, EmptyPartition.
Note This partition will not be assigned to any phone line, but it will be assigned to the NoRouteCSS.
Configure Calling Search Space for Hotline Call Only Receive Only
You must create a calling search and assign one of the two partitions that you've created to the calling search
space.
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Placing Calls
Configure Call Only on Hotline Phone
Before you begin

Configure Partitions for Hotline Call Only Receive Only, on page 262
Procedure
Step 3 Enter a Name and Description for the calling search space.
Step 4 From the Available Partitions list box, use the arrows to select the EmptyPartition partition.
Note Make sure that the partition is assigned to only this calling search space and to no phone lines.
Step 5 Click Save
What to do next
Perform one of the following procedures:
• Configure Call Only on Hotline Phone, on page 263
• Configure Receive Only on Hotline Phone, on page 263
Configure Call Only on Hotline Phone

If you have set up your partitions and calling search spaces, perform these steps to configure the Hotline phone
to place calls only.
Before you begin

Configure Calling Search Space for Hotline Call Only Receive Only, on page 262
Procedure
Step 1 From Cisco Unified CM Administration, choose Call Routing > Phone.
Step 2 Click Find and select the Hotline phone.
Step 3 From the left navigation pane, click the phone line.
The Directory Number Configuration window displays.
Step 4 From the Route Partition drop-down list, select the empty partition that you created.
Step 5 Click Save.
Configure Receive Only on Hotline Phone

If you have created your calling search space and partitions already, perform these steps to configure the
Hotline phone to receive calls only.
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Placing Calls
Configure Call Screening with a Calling Search Space
Before you begin

Configure Calling Search Space for Hotline Call Only Receive Only, on page 262
Procedure

Step 3 From the Calling Search Space drop-down list, select the new CSS that you created in the previous procedure.
Step 4 Click Save.
Configure Call Screening with a Calling Search Space

Configure call screening for any intraswitched (line to line) Hotline calls by assigning a unique CSS where
the Hotline phones that are in the partitions are only those Hotline phones that you want to be able to call each
other.
Note You can also configure call screening by creating translation patterns where each pattern matches each number
pattern that you want to either allow or screen.
Procedure

Step 1 Configure Partitions for Hotline Call Screening, Create any new partitions for your Hotline
on page 264 phone lines.
Step 2 Create Calling Search Space for Hotline Call Create a new CSS for the screening list. The
Screening, on page 265 CSS must include partitions with only those
Hotline numbers that you want to allow.
Step 3 Configure Hotline Phones for Call Screening, Assign the new CSS and partition to the Hotline
on page 266 phone.
Configure Partitions for Hotline Call Screening

To configure call screening in Hotline phones using a calling search space, you must set up partitions where
the only Hotline numbers are those that you want to allow.
Perform the following procedure if you need to create a new partition for your Hotline call screening list.
Procedure
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Placing Calls
Create Calling Search Space for Hotline Call Screening
name in this field.
incoming call.
Step 8 Click Save.
Create Calling Search Space for Hotline Call Screening

Perform the following procedure to create a new calling search space for the Hotline phones in the call screening
list. Make sure that the only Hotline numbers in the partitions that you select for this CSS are those Hotline
numbers that you want to allow in the call screening list. No Hotline numbers that you want to screen out
should be included in the partitions for this CSS.
Before you begin

Configure Partitions for Hotline Call Screening, on page 264
Procedure

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Placing Calls
Configure Hotline Phones for Call Screening
Partitions box.
Step 8 Click Save.
Configure Hotline Phones for Call Screening

If you have already configured calling search spaces and partitions for Hotline call screening, perform this
procedure to assign the calling search spaces and partitions to your Hotline phones.
Before you begin

Create Calling Search Space for Hotline Call Screening, on page 265
Procedure

Step 3 From the Calling Search Space drop-down list, select the new calling search space that you created for the
Hotline call screening list.
Step 4 Click Save.
Step 5 From the left navigation pane, click the phone line that you want to use for Hotline calls.
The Directory Number Configuration window displays.
Step 6 From the Route Partition drop-down list, select a partition that is included in the calling search space that
you set up.
Step 7 Click Save.
Hotline Troubleshooting
The following table provides troubleshooting information for cases where hotline calls do not dial correctly.
Table 24: Troubleshooting Hotline—Calls Do Not Dial Correctly
Problem Solution
Dial tone Check PLAR configuration.
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Placing Calls
Problem Solution
Reorder tone or VCA (intracluster call) • Check PLAR configuration.

• Verify that the phones on both ends are
configured as hotline phones.
Reorder tone or VCA (intercluster or TDM call) • Check PLAR configuration.

• Verify that the phones on both ends are
configured as hotline phones.
• Verify that route class signalling is enabled on
trunks.
• Check the configuration of route class
translations on CAS gateways.
The following table provides troubleshooting information for cases where call screening based on caller ID
does not work.
Table 25: Troubleshooting Hotline—Call Screening Based on Caller ID Problems
Problem Solution
Call not allowed • Check Caller ID.

• Add pattern to screen CSS.
Call allowed Remove pattern from screen CSS.
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268
CHAPTER 23
Speed Dial and Abbreviated Dial
• Speed Dial and Abbreviated Dial Overview, on page 269
• Speed Dial and Abbreviated Dial Configuration Task Flow, on page 270
Speed Dial and Abbreviated Dial Overview

Administrators can configure speed dial numbers for phones to provide speed dial buttons for users or to
configure phones that do not have a specific user that is assigned to them. Users use the Cisco Unified
Communications Self Care Portal to change the speed dial buttons on their phones. When configuring speed
dial entries, some of the speed dial entries are assigned to the speed dial buttons on the IP phone; the remaining
speed dial entries are used for abbreviated dialing. When a user starts dialing digits, the AbbrDial softkey
displays, and the user can access any speed dial entry by entering the appropriate index (code) for abbreviated
dialing.
The speed dial settings on the phone are associated with a physical button on a phone, whereas the abbreviated
dial settings are not associated with a phone button.
Programming Speed Dials with Pauses

You can program commas in your speed dials to reach destinations that require a Forced Authorization Code
(FAC), Client Matter Code (CMC), dialing pause, or additional digits (such as a user extension, meeting
access number, or voice mail password). Within a speed dial, each comma (,) represents either:
• A delimiter that separates the destination call address from an FAC or CMC code
• A pause of 2 seconds prior to sending post-connect DTMF digits
For example, let’s say that you want a speed dial that includes FAC and CMC codes, followed by IVR prompts
where:
• The called number is 91886543.
• The FAC code is 8787.
• The CMC code is 5656.
• The IVR response is 987989#, which must be entered 4 seconds after the call connects.
In this case, you would program 91886543,8787,5656,,987989# as the speed dial.
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Speed Dial and Abbreviated Dial Configuration Task Flow
Speed Dial and Abbreviated Dial Configuration Task Flow

Procedure

support the Speed Dial and Abbreviated Dial
feature.
Step 2 Configure Speed Dial and Abbreviated Dial, Configure Speed Dial and Abbreviated Dial
on page 270 numbers.
Configure Speed Dial and Abbreviated Dial

You can configure a total of 199 speed dial and abbreviated dial settings. Configure speed dial settings for
the physical buttons on the phone. Configure abbreviated dial settings for the speed dial numbers that you
access with abbreviated dialing. You can configure speed dial entries and abbreviated dial indexes in the same
window.
You can also configure post connect DTMF digits as well as FAC , CMC codes as part of the speed dial.
Follow these steps to configure speed dial and abbreviated dial.
Note Not all Cisco Unified IP Phones support abbreviated dialing. See the phone user guide for information.
Before you begin

Procedure
Step 1 From Cisco Unified CM Administration, choose Device > Phone. Enter your search criteria and click Find.
Choose the phone for which you want to configure speed dial buttons.
Step 2 From the Phone Configuration window, choose Add/Update Speed Dials from the Related Links drop-down
list at the top of the window and click Go.
The Speed Dial and Abbreviated Dial Configuration window appears for the phone.
Step 3 In the Number field, enter the number that you want the system to dial when the user presses the speed dial
button or the abbreviated dial index for abbreviated dial. You can enter digits 0 through 9, *, #, and +, which
is the international escape character. To include dialing pauses in the speed dial, you can enter comma (,)
which can act as a delimiter before sending DTMF digits. Each comma you include represents an additional
pause of 2 seconds. For example, two commas (,,) represent a pause of 4 seconds. Use of commas also allows
you to separate FAC and CMC from the other digits in the speed dial string.
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Note Ensure that the following requirements are met when you include FAC and CMC in the speed
dial string:
• FAC must always precede CMC in the speed dial string.
• A speed dial label is required for speed dials with FAC and DTMF digits.
• Only one comma is allowed between FAC and CMC digits in the string.
Step 4 In the Label field, Enter the text that you want to display for the speed dial button or abbreviated dial number.
Note This field is not available for all the phones. To determine whether this field is available for your
Cisco Unified IP Phone, see the user documentation for your phone model.
Step 5 (Optional) If you are configuring a pause in speed dial, you must add a label so that FAC, CMC, and DTMF
digits are not displayed on the phone screen.
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272
CHAPTER 24
WebDialer
• WebDialer Overview, on page 273
• WebDialer Prerequisites, on page 273
• WebDialer Configuration Task Flow, on page 273
• WebDialer Interactions, on page 284
• WebDialer Restrictions, on page 285
• WebDialer Troubleshooting, on page 285
WebDialer Overview
Cisco WebDialer is installed on a Unified Communications Manager node and used along with Unified
Communications Manager. It allows Cisco Unified IP Phone users to make calls from web and desktop
applications.
Cisco WebDialer uses hyperlinked telephone numbers in a company directory to allow users to make calls
from a web page by clicking on the telephone number of the person that they are trying to call. Cisco WebDialer
supports both IPv4 and IPv6 addressing.
In the Cisco Unified Communications Self-Care Portal, from the Directory window, launch Cisco WebDialer
using a URL similar to the following:
https://<IP address of Cisco Unified Communications Manager server>:8443/webdialer/
Webdialer
WebDialer Prerequisites
Cisco WebDialer requires the following software components:
• CTI-supported Cisco Unified IP Phones
WebDialer Configuration Task Flow

Before you begin
• Review WebDialer Prerequisites, on page 273.
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WebDialer Configuration Task Flow
Procedure

Step 1 Activate WebDialer, on page 275 Activate the WebDialer service.
Step 2 (Optional) Enable WebDialer Tracing, on page To view WebDialer traces, enable tracing.
275
Step 3 (Optional) Configure WebDialer Servlet, on Configure the WebDialer servlet.
page 276
Step 4 (Optional) Configure Redirector Servlet, on If you have multi cluster applications that you
page 276 develop using HTML over HTTPS interfaces,
configure the Redirector servlet.
Step 5 (Optional) Configure WebDialer Application To configure Redirector for Cisco WebDialer.
Server, on page 277
Step 6 (Optional) To Configure Secure TLS WebDialer uses WDSecureSysUser application
Connection to CTI, on page 277, complete the user credentials to establish a secure TLS
following sub tasks: connection to CTI to make calls. Follow these
procedures if your system is running in mixed
• Configure WDSecureSysUser
mode.
Application User, on page 278
• Configure CAPF Profile, on page 176
• Configure Cisco WebDialer Web Service
, on page 177
Step 7 Configure Language Locale for WebDialer , Determine which language WebDialer displays
on page 280 by setting the locale field in the Cisco Unified
Communications Self Care Portal menu.
Step 8 Configure WebDialer Alarms, on page 281 If there are any issues with the Web Dialer
feature it alerts the administrator.
Step 9 (Optional) Configure Application Dial Rules, If your application requires multiple clusters,
on page 281 configure application dial rules.
Step 10 Add Users to Standard CCM End User Group, Add each WebDialer user to the Standard End
on page 282 User Group for Cisco Unified Communications
Manager.
Step 11 (Optional) To Configure Proxy User, on page If you use makeCallProxy HTML over HTTP
282, complete the following sub tasks: interface to develop an application for using
Cisco WebDialer, create a proxy user.
• Add a WebDialer End User, on page 283
• Assign Authentication Proxy Rights, on
page 283
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Activate WebDialer
Activate WebDialer
Procedure
Step 2 From the Servers drop-down list, choose the Unified Communications Manager server that is listed.
Step 3 From CTI Services, check the Cisco WebDialer Web Service check box.
Step 4 Click Save.
Step 5 From Cisco Unified Serviceability, choose Tools > Control Center - Feature Services to confirm that the
CTI Manager service is active and is in start mode.
For WebDialer to function properly, the CTI Manager service must be active and in start mode.
What to do next
Configure Language Locale for WebDialer , on page 280 or complete any or all of the following optional
tasks:
• Enable WebDialer Tracing, on page 275
• Configure WebDialer Servlet, on page 276
• Configure Redirector Servlet, on page 276
• Configure WebDialer Application Server, on page 277
• Configure Secure TLS Connection to CTI, on page 277
Enable WebDialer Tracing

To enable Cisco WebDialer tracing, use the Cisco Unified Serviceability Administration application. Trace
settings apply to both the WebDialer and Redirector servlets. To collect traces, use the Real Time Monitoring
Tool (RTMT).
To access the WebDialer trace files, use the following CLI commands:
• file get activelog tomcat/logs/webdialer/log4j
• file get activelog tomcat/logs/redirector/log4j
For more information about traces, see the Cisco Unified Serviceability Administration Guide.
Before you begin

Activate WebDialer, on page 275
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Configure WebDialer Servlet
Procedure
Step 1 From the navigation drop-down list of the Cisco Unified Communications Manager application, choose Cisco
Unified Serviceability and then click Go.
Step 2 Choose Trace > Configuration.
Step 3 From the Server drop-down list, choose the server on which to enable tracing.
Step 4 From the Service Group drop-down list, choose CTI Services.
Step 5 From the Service drop-down list, choose the Cisco WebDialer Web Service.
Step 6 In the Trace Configuration window, change the trace settings according to your troubleshooting requirements.
Note For more information about WebDialer trace configuration settings, see the Cisco Unified
Serviceability Administration Guide.
Step 7 Click Save.
Configure WebDialer Servlet

The WebDialer servlet is a Java servlet that allows Cisco Unified Communications Manager users in a specific
cluster to make and complete calls.
Before you begin

Procedure
Step 1 Choose System > Service Parameters.

Step 2 From the Server drop-down list, choose the Cisco Unified Communications Manager server on which to
configure Cisco WebDialer web service parameters.
Step 3 From the Service drop-down list, choose Cisco WebDialer Web Service.
Step 4 Configure the relevant WebDialer Web Service parameters. For detailed information about the parameters,
see online help.
Step 5 Restart the Cisco WebDialer Web Service for new parameter values to take effect.
Configure Redirector Servlet

Redirector servlet is a Java-based Tomcat servlet. When a Cisco WebDialer user makes a request, the Redirector
servlet looks for that request in the Cisco Unified Communications Manager cluster and redirects the request
to the specific Cisco WebDialer server that is located in the Cisco Unified Communications Manager cluster.
The Redirector servlet is available only for multi cluster applications that are developed by using HTML over
HTTPS interfaces.
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Configure WebDialer Application Server
Before you begin

Procedure
Step 2 From the Server drop-down list, choose the Cisco Unified Communications Manager server on which to
configure the Redirector Servlet.
Step 3 From the Service drop-down list, choose the Cisco WebDialer Web Service.
Step 4 Configure the relevant WebDialer Web Service parameters. For detailed information about the parameters,
see online help.
Step 5 Restart the Cisco WebDialer Web Service for new parameter values to take effect.
For more information on WebDialer Web Service, see the Cisco Unified Serviceability Administration Guide.
Configure WebDialer Application Server

Application server is required to configure the Redirector Servlet. Redirector is required only when you have
multiple Unified Communications Manager servers configured in a cluster.
Before you begin

Procedure
Step 1 From Cisco Unified Communications Manager Administration Application server window, choose System >
Application Server.
Step 2 From the Application Server Type drop-down list, choose a Cisco WebDialer application server.
The server appears in the List of WebDialers field in the Service Parameter Configuration window for the
Cisco WebDialer Web Service.
Configure Secure TLS Connection to CTI

WebDialer uses WDSecureSysUser application user credentials to establish a secure TLS connection to CTI
to make calls. To configure the WDSecureSysUser application user to establish a secure TLS connection,
complete the following tasks.
Before you begin

• Install and configure the Cisco CTL Client. For more information about CTL Client, see Security Guide
for Cisco Unified Communications Manager .
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Configure WDSecureSysUser Application User
• Verify that the Cluster Security Mode in the Enterprise Parameters Configuration window is 1 (mixed
mode). Operating the system in mixed mode impacts other security functions in your system. If your
system is not currently running in mixed mode, do not switch to mixed mode until you understand these
interactions. For more information, see Security Guide for Cisco Unified Communications Manager.
• Verify that the Cluster SIPOAuth Mode field is set to Enabled.
• Activate the Cisco Certificate Authority Proxy Function service on the first node.
• Activate WebDialer, on page 275
Procedure

Step 1 Configure WDSecureSysUser Application User, Configure a WDSecureSysUser application
on page 278 user.
Step 2 Configure CAPF Profile, on page 176 Configure a CAPF profile for the
WDSecureSysUser application user.
Step 3 Configure Cisco WebDialer Web Service , on Configure service parameters for the Cisco
page 177 WebDialer Web service.
Configure WDSecureSysUser Application User
Procedure
Step 2 Click Find.
Step 3 From the Find and List Application Users Application window, choose WDSecureSysUser.
Step 4 Configure the fields in the Application User Configuration window and click Save.
What to do next
Configure CAPF Profile, on page 176

Certificate Authority Proxy Function (CAPF) is a component that performs tasks to issue and authenticate
security certificates. When you create an application user CAPF profile, the profile uses the configuration
details to open secure connections for the application.
Procedure
Step 1 From Cisco Unified CM Administration, choose User Management > Application User CAPF Profile.
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• To add a new CAPF profile, click Add New in the Find window.
• To copy an existing profile, locate the appropriate profile and click the Copy icon for that record in the
Copy column.
To update an existing entry, locate and display the appropriate profile.
Step 3 Configure or update the relevant CAPF profile fields. See the Related Topics section information about the
Step 4 Click Save.
Step 5 Repeat the procedure for each application and end user that you want to use security.
Setting Description
Application User From the drop-down list, choose the application user for the CAPF operation. This set
This setting does not appear in the End User CAPF Profile window.
End User ID From the drop-down list, choose the end user for the CAPF operation. This setting dis
This setting does not appear in the Application User CAPF Profile window.
Instance ID Enter 1 to 128 alphanumeric characters (a-z, A-Z, 0-9). The Instance ID identifies the
You can configure multiple connections (instances) of an application. To secure the co
instance that runs on the application PC (for end users) or server (for application users
This field relates to the CAPF Profile Instance ID for Secure Connection to CTIManag
Certificate Operation From the drop-down list, choose one of the following options:
• No Pending Operation—This message is displayed when no certificate operatio
• Install/Upgrade—This option installs a new certificate or upgrades an existing l
Authentication Mode The authentication mode for the Install/Upgrade certificate operation specifies By Authe
a locally significant certificate only when the user or administrator enters the CAPF au
Authentication String To create your own authentication string, enter a unique string.
Each string must contain 4 to 10 digits.
To install or upgrade a locally significant certificate, the administrator must enter the auth
This string supports one-time use only; after you use the string for the instance, you ca
Generate String To automatically generate an authentication string, click this button. The 4- to10-digit
Key Size (bits) From the drop-down list, choose the key size for the certificate. The default setting is
Key generation, which is set at low priority, allows the application to function while th
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Configure Cisco IP Manager Assistant
Setting Description
Operation Completes by This field, which supports all certificate operations, specifies the date and time by which y
The values that are displayed apply for the first node.
Use this setting with the CAPF Operation Expires in (days) enterprise parameter, which s
must be completed. You can update this parameter at any time.
Certificate Operation Status This field displays the progress of the certificate operation, such as pending, failed, or suc
You cannot change the information that is displayed in this field.
Configure Cisco IP Manager Assistant
Procedure
Step 2 From the Server drop-down list, choose the server on which the Cisco WebDialer Web service is active.
Step 3 From the Service drop-down list, choose the Cisco WebDialer Web service.
A list of parameters appears.
Step 4 Navigate to and update the CTIManager Connection Security Flag and CAPF Profile Instance ID for Secure
Connection to CTIManager parameters.
To view parameter descriptions, click the parameter name link.
Note CTIManager supports IPv4 and IPv6 addresses.
Step 5 Click Save.

Step 6 Repeat the procedure on each server on which the service is active.
What to do next
Configure Language Locale for WebDialer

Use the Cisco Unified Communications Self Care Portal to configure a language locale for Cisco WebDialer.
The default language is English.
Before you begin

Procedure
Step 1 From the Cisco Unified Communications Self Care Portal, click the General Settings tab.
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Configure WebDialer Alarms
Step 2 Click Language.

Step 3 From the Display Language drop-down list, select a language local, and then click Save.
Configure WebDialer Alarms

Cisco WebDialer service uses Cisco Tomcat to generate alarms.
Before you begin

Configure Language Locale for WebDialer , on page 280
Procedure
Step 1 From Cisco Unified Serviceability, choose Alarm > Configuration.

Step 2 From the Server drop-down list, choose the server on which to configure the alarm and then click Go.
Step 3 From the Services Group drop-down list, choose Platform Services and then click Go.
Step 4 From the Services drop-down list, choose Cisco Tomcat and then click Go.
Step 5 If your configuration supports clusters, check the Apply to All Nodes check box to apply the alarm
configuration to all nodes in the cluster.
Step 6 Configure the settings, as described in Alarm configuration settings, which includes descriptions for monitors
and event levels.
Note For more information about the Alarm configuration settings, see the Cisco Unified Serviceability
Guide.
Step 7 Click Save.
What to do next
Add Users to Standard CCM End User Group, on page 282 or (optionally) if your application requires multiple
clusters, see Configure Application Dial Rules, on page 281.
Configure Application Dial Rules

Before you begin
Configure WebDialer Alarms, on page 281
Procedure
Step 1 From Cisco Unified CM Administration, choose Call Routing > Dial Rules > Application Dial Rules.
Step 2 In the Name field, enter a name for the dial rule.
Step 3 In the Description field, enter a description for the dial rule.
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Add Users to Standard CCM End User Group
Step 4 In the Number Begins With field, enter the initial digits of the directory numbers to which you want to apply
this application dial rule.
Step 5 In the Number of Digits field, enter the length of the dialed numbers to which you want to apply this application
dial rule.
Step 6 In the Total Digits to be Removed field, enter the number of digits that you want Unified Communications
Manager to remove from the beginning of dialed numbers that apply to this dial rule.
Step 7 In the Prefix With Pattern field, enter the pattern to prepend to dialed numbers that apply to this application
dial rule.
Step 8 For Application Dial Rule Priority, choose the dial rule priority as top, bottom, or middle.
Step 9 Click Save.
Add Users to Standard CCM End User Group

To use the Cisco WebDialer links in the User Directory windows in Unified Communications Manager, you
must add each user to the Standard Unified Communications Manager End Users Group.
Procedure
Step 1 Choose User Management > User Group.

Step 2 In the Find and List User Group window, click Find.
Step 3 Click Standard CCM End Users.
Step 4 In the User Group Configuration window, click Add End Users to Group.
Step 5 In the Find and List Users window, click Find. You can enter criteria for a specific user.
Step 6 To add one or more users to the user group, complete one of the following steps:
• To add one or more users, check the check box beside each user to add and then click Add Selected.
• To add all users, click Select All and then click Add Selected.
The users appear in the Users in Group table of the User Group Configuration window.
Configure Proxy User

If you use makeCallProxy HTML over HTTP interface to develop an application for using Cisco WebDialer,
create a proxy user. For information about the makeCallProxy interface, see the makeCallProxy section in
the Cisco WebDialer API Reference Guide.
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Add a WebDialer End User
Note MakeCallProxy HTTP Methods is a service parameter under WebDialer Service. This parameter controls the
HTTP methods that the MakeCallProxy API accepts. HTTP GET is considered insecure because the credentials
required by the API are included as parameters in HTTP GET requests. Hence these HTTP GET parameters
can be captured in the application logs and in the web browser's history.
When the service parameter MakeCallProxy HTTP Methods is set to Secure, request made by the HTTP GET
will be rejected. By default the parameter MakeCallProxy HTTP Methods is set to Insecure, so that the API
accepts both GET and POST methods and the backward compatibility is maintained.
Before you begin

Add Users to Standard CCM End User Group, on page 282
Procedure

Step 1 (Optional) Add a WebDialer End User, on page Add a new user. If the user exists, you can
283 proceed to the next task.
Step 2 Assign Authentication Proxy Rights, on page Assign authentication proxy rights to an end
283 user.
Add a WebDialer End User
Procedure
Step 3 Enter a Last Name.
Step 4 Enter and confirm a Password.
Step 5 Enter and confirm a PIN.
Step 6 Complete any remaining fields in the End User Configuration window. For more information on the fields
Step 7 Click Save.
Assign Authentication Proxy Rights

Perform the following procedure to enable authentication proxy rights for an existing user.
Procedure
Step 1 Choose User Management > User Group.

The Find and List User Group window appears.
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WebDialer Interactions
Step 2 Click Find.

Step 3 Click the Standard EM Authentication Proxy Rights link.
The User Group Configuration window appears.
Step 4 Click Add End Users to Group.
The Find and List Users window appears.
Step 5 Click Find. You can also add a criteria for a specific user.
Step 6 To assign proxy rights to one or more users, complete one of the following steps:
Step 7 To add a single user, select the user and then click Add Selected.
Step 8 To add all users that appear in the list, click Select All and then click Add Selected.
The user or users appear in the Users in Group table in the User Group Configuration window.
WebDialer Interactions
Feature Interaction
Client Matter Codes When you use CMCs, you must enter the proper code at the tone; otherwise, the IP
(CMC) phone disconnects and the user receives a reorder tone.
Forced Authorization When you use FACs, you must enter the proper code at the tone; otherwise, the IP
Codes (FAC) phone disconnects and the user receives a reorder tone.
ApplicationDialRule Cisco WebDialer uses change notifications on the ApplicationDialRule database table
table to track and use updated dial rules.
Client Matter Codes Web Dialer supports CMCs and FACs in the following ways:
and Forced
• A user can enter the destination number in the dial text box of the WD HTML
Authorization Codes
page or SOAP request, and then manually enter the CMC or FAC on the phone.
• A user can enter the destination number followed by the FAC or CMC in the
dial text box of the WD HTML page or SOAP request.
For example, if the destination number is 5555, the FAC is 111, and the CMC is 222,
a user can make a call by dialing 5555111# (FAC), 5555222# (CMC), or 5555111222#
(CMC and FAC).
Note • WebDialer does not handle any validation for the destination
number. The phone handles the required validation.
• If a user does not provide a code or provides the wrong code, the
call will fail.
• If a user makes a call from the WebApp with a DN that contains
special characters, the call goes successfully after stripping the
special characters. The same rules do not work in SOAP UI.
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WebDialer Restrictions
WebDialer Restrictions
Phones Cisco WebDialer supports phones that run Skinny

Client Control Protocol (SCCP) and Session Initiation
Protocol (SIP) that Cisco Computer Telephony
Integration (CTI) supports.
Note Few older phone models do not support
Cisco Web Dialer that run SIP.
WebDialer Troubleshooting
Authentication Error
Problem
Cisco WebDialer displays the following message:
Authentication failed, please try again.
Possible Cause
User entered wrong user ID or password.
Solution
Ensure that you use your Unified Communications ManagerCisco Unified Communications Manager user
ID and password to log in.
Service Temporarily Unavailable

Problem
Service temporarily unavailable, please try again later.
Possible Cause
The Cisco CallManager service became overloaded because it has reached its throttling limit of three concurrent
CTI sessions.
Solution
After a short time, retry your connection.
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Directory Service Down
Directory Service Down

Problem
Service temporarily unavailable, please try again later: Directory service down.
Possible Cause
The Cisco Communications Manager directory service may be down.
Solution
Cisco CTIManager Down

Problem
Service temporarily unavailable, please try again later: Cisco CTIManager down.
Possible Cause
Cisco CTIManager service that is configured for Cisco Web Dialer went down.
Solution
Session Expired, Please Login Again

Problem
Session expired, please login again.
Possible Cause
A Cisco Web Dialer session expires:
• After the WebDialer servlet gets configured
• If the Cisco Tomcat Service is restarted.
Solution
Log in by using your Unified Communications Manager User ID and Password.
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User Not Logged In on Any Device
User Not Logged In on Any Device

Problem
Cisco Web Dialer displays the following message:
User not logged in on any device.
Possible Cause
The user chooses to use Cisco Extension Mobility from the Cisco WebDialer preference window but does
not get log in to any IP phone.
Solution
• Log in to a phone before using Cisco WebDialer.
• Choose a device from the Cisco WebDialer preference list in the dialog box instead of choosing the
option Use Extension Mobility.
Failed to Open Device/Line

Problem
After a user attempts to make a call, Cisco WebDialer displays the following message:
User not logged in on any device.
Possible Cause
• The user chose a Cisco Unified IP Phone that is not registered with Unified Communications Manager.
For example, the user chooses a Cisco IP SoftPhone as the preferred device before starting the application.
• The user who has a new phone chooses an old phone that is no longer in service.
Solution
Choose a phone that is in service and is registered with Unified Communications Manager.
Destination Not Reachable

Problem
Cisco WebDialer displays the following message on the End Call window:
Destination not reachable.
Possible Cause
• User dialed the wrong number.
• The correct dial rules did not get applied. For example, the user dials 5550100 instead of 95550100.
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Destination Not Reachable
Solution
Check the dial rules.
288
CHAPTER 25
Paging
• Paging Overview, on page 289
• Paging Prerequisites, on page 290
• Cisco Unified Communications Manager Configuration for Basic Paging Task Flow, on page 291
• Advanced Notification Paging Configuration Task Flow, on page 301
• Paging Interactions, on page 307
Paging Overview
Unified Communications Manager can be configured to integrate with Cisco Paging Server to provide basic
paging services for Cisco Unified IP Phone and a variety of endpoints. The Cisco Paging Server product is
offered through the InformaCast Virtual Appliance and offers the following deployment options:
InformaCast Basic Paging

InformaCast Basic Paging provides phone-to-phone live audio paging to individual Cisco IP phones or groups
of up to 50 phones simultaneously. InformaCast Basic Paging is free to all Unified Communications Manager
customers and all Cisco Business Edition 6000 and Cisco Business Edition 7000 customers.
InformaCast Advanced Notification

InformaCast Advanced Notification is a full-featured emergency notification and paging solution that allows
you to reach an unlimited number of Cisco IP phones and various devices and systems with text and audio
messages.
To streamline the configuration process, Unified Communications Manager comes with a provisioning wizard
that allows you to quickly configure advanced notifications services.
Some of the features include:
• Text and audio (live or pre-recorded) to Cisco IP Phones and other endpoints
• Analog and IP overhead paging systems integration
• 911 or emergency call monitoring or alerting or recording
• Cisco Jabber integration
• Cisco Spark integration
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InformaCast Mobile
• Automated weather notifications

• Dynamically triggered emergency conference calls
• Pre-recorded or scheduled broadcasts (school bells or shift changes)
• Event accountability with message confirmation and reporting
• Notification to computer desktops (Windows and Mac OS)
• Facilities integration (control lighting, door locks)
• Security integration (panic or duress buttons, motion detectors, fire)
Purchase a license key to access InformaCast Advanced Notification features.
InformaCast Mobile
InformaCast Mobile is a cloud-based service that allows users to send images, text, and pre-recorded audio
to mobile devices running iOS or Android. It also has bi-directional integration with InformaCast Advanced
Notification.
Some of the features include:
• The ability to send and receive InformaCast messages via mobile devices running iOS or Android
• Bi-directional integration with InformaCast Advanced Notification
• Message confirmations and read receipts
• No calling or SMS messaging fees
InformaCast Mobile must be purchased direct from Singlewire Software. Please refer to the Singlewire website
for addtional details and downloads.
If you have already configured Unified Communications Manager to integrate with InformaCast Advanced
Notification, no further configuration of Unified Communications Manager is required.
Paging Prerequisites
Cisco Paging Server is designed to work in a multicast environment. You must configure your network for
multicast.
For a list of Cisco Unified IP Phones that support paging, refer to the Cisco Unified IP Phones section of the
Singlewire Compatibility Matrix at:
http://www.singlewire.com/compatibility-matrix.html.
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Cisco Unified Communications Manager Configuration for Basic Paging Task Flow
Cisco Unified Communications Manager Configuration for Basic

Paging Task Flow
Perform the following tasks to configure Unified Communications Manager to integrate with Cisco Paging
Server for an InformaCast Basic Paging deployment.
Before you begin

• Learn more about the feature by reviewing the following:
• Paging Overview, on page 289
• InformaCast Basic Paging, on page 289
• Review Paging Prerequisites, on page 290

• The configuration in this section is automated when using the Advanced Notification Paging Configuration
Task Flow wizard.
Procedure

Step 1 Enable SNMP Service, on page 292 Configure SNMP in Unified Communications
Manager.
Step 2 Set Default Codec to G.711, on page 293 Set the default codec to G.711.
Step 3 Configure a Device Pool for Paging, on page Configure a device pool.
294
Step 4 Configure Route Partition for InformaCast Configure a route partition for Basic Paging.
Paging, on page 295
Step 5 Configure Calling Search Space for Configure a calling search space for Basic
InformaCast Paging, on page 295 Paging.
Step 6 Configure CTI Ports for Paging, on page 296 Configure CTI ports.
Step 7 Configure Access Control Group with AXL Configure an AXL access control group.
Access, on page 296
Step 8 Configure Application User for Paging, on Configure an application user.
page 297
Step 9 Enable web access for the phone using one of You can enable web access on all phones
the following procedures: globally using Enterprise Phone Configuration,
a group of phones using a Common Phone
• Enable Web Access for a Phone, on page
Profile, or an individual phone.
298
• Enable Web Access for Common Phone
Profile, on page 298
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Configure SNMP for Paging

• Enable Web Access for Enterprise Phone
Step 10 Configure Authentication URL, on page 299 Configure the Unified Communications
Manager authentication URL to point to
InformaCast so that when InformaCast pushes
broadcasts to Cisco Unified IP Phone, the
phones will authenticate with InformaCast.
For detailed procedures on how to configure Cisco Unified Communications Manager and Cisco Paging
Server, refer to the InformaCast Virtual Appliance Basic Paging Installation and User Guide .
Configure SNMP for Paging

Perform the following tasks to configure SNMP services in the cluster.
Procedure

Step 1 Enable SNMP Service, on page 292 Enable the SNMP and other services in the
cluster.
Step 2 Create an InformaCast SNMP Community Configure an SNMP community string.

String, on page 293
Enable SNMP Service

To configure paging, you must enable SNMP on every node in the cluster. In addition, you must enable the
following services:
• Cisco CallManager SNMP Service—Enable on all nodes in the cluster.
• Cisco CallManager—Enable on at least one node.
• Cisco AXL Web Services—Enable on at least one node.
• Cisco CTIManager—Enable on at least one node.
Procedure
Step 2 From the Server drop-down list, choose the server on which you want to configure SNMP.
Step 3 Check the check boxes that correspond to the Cisco CallManager SNMP Service.
Step 4 For at least one server in the cluster, check the check boxes that correspond to Cisco CallManager, Cisco
CTIManager, and Cisco AXL Web Service services.
Step 5 Click Save.
Step 6 Click OK.
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Create an InformaCast SNMP Community String
Step 7 Repeat the previous steps for all nodes in the cluster.
Create an InformaCast SNMP Community String

Perform this procedure for Basic Paging to set up an SNMP community string.
Before you begin

Enable SNMP Service, on page 292
Procedure
Step 1 From Cisco Unified Serviceability, choose SNMP > V1/V2c > Community String.
Step 2 From the Server drop-down list, choose a server and click Find.
Step 4 In the Community String Name field, enter ICVA.
Step 5 From the Access Privileges drop-down list, select ReadOnly.
Step 6 Check the Apply to All Nodes check box if the check box is active.
Step 7 Click Save.
Step 8 Click OK.
What to do next
Set Default Codec to G.711, on page 293
Configure Region for Paging

For Basic Paging, you must set up a region for your paging deployment.
Procedure

Step 1 Set Default Codec to G.711, on page 293 Create a region that uses the G.711 codec for
calls to other regions.
Step 2 Configure a Device Pool for Paging, on page Set up a device pool for paging and assign the
294 region that you created to that device pool.
Set Default Codec to G.711

You must create an InformaCast region that uses G.711 as the default codec for calls to other regions.
Before you begin

Configure SNMP for Paging, on page 292
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Configure a Device Pool for Paging
Procedure
Step 1 From Cisco Unified CM Administration, choose System > Region Information > Region.
Step 3 In the Name field, enter ICVA.
Step 4 Click Save.
Step 5 In the Regions text box, select all regions by pressing the CTRL key and clicking all of the selected regions.
Step 6 From the Maximum Audio Bit Rate drop-down list, select 64 kbps (G.722, G.711).
Step 7 From the Maximum Session Bit Rate for Video Calls column click the None radio button.
Step 8 Click Save.
Configure a Device Pool for Paging

Perform this procedure to configure a device pool for your paging deployment.
Before you begin

Set Default Codec to G.711, on page 293
Procedure
Step 3 In the Device Pool Name field, enter ICVA.
Step 4 From the Cisco Unified Communications Manager Group drop-down list, select the group that contains
the Cisco Unified Communications Manager cluster with which the InformaCast Virtual Appliance will
communicate.
Step 5 From the Date/Time Group drop-down list, select a date/time group. Select CMLocal unless you are
performing dialing restrictions by the time of day.
Step 6 From the Region drop-down list, choose ICVA.
Step 7 From the SRST Reference drop-down list, select Disable.
Step 8 Click Save.
Configure Partitions and Calling Search Spaces for Paging

Perform the following tasks to configure a partition and calling search space (CSS) for paging as follows:
• For Basic Paging deployments, create a single partition and CSS for InformaCast paging.
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Configure Route Partition for InformaCast Paging
Procedure

Step 1 Configure Route Partition for InformaCast Configure a route partition for InformaCast
Paging, on page 295 paging.
Step 2 Configure Calling Search Space for Configure a calling search space for
InformaCast Paging, on page 295 InformaCast paging.
Configure Route Partition for InformaCast Paging

Create a route partition for InformaCast paging.
Before you begin

Configure a Device Pool for Paging, on page 294
Procedure
Step 1 From Cisco Unified CM Administration, choose Call Routing > Class of Control > Route Partitions.
Step 3 In the Name field, enter the following name and description for the partition:
ICVA-CTIOutbound,ICVA-Do not add to any phone CSS.
Step 4 Click Save.
Configure Calling Search Space for InformaCast Paging

Perform this procedure to configure a calling search space for InformaCast paging.
Procedure
Step 3 In the Name field, enter ICVA.
Step 4 In the Available Partitions list box, use the arrows to move the following partitions to the Selected Partitions
list box.
• The partition that you created for InformaCast paging
• The partitions that contain your users' extensions and any analog paging extensions
Step 5 Click Save.
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Configure CTI Ports for Paging
Configure CTI Ports for Paging

Perform this procedure to configure CTI ports for your paging deployment. The number of CTI ports that you
need depends on your deployment type and your applications' usage:
• For Basic Paging deployments, you must create a minimum of two CTI ports for InformaCast paging.
Before you begin

Configure Calling Search Space for InformaCast Paging, on page 295
Procedure

Step 3 From the Phone Type drop-down list, choose CTI Port.
Step 4 In the Device Name field, enter a name for the CTI Port. For example, ICVA-IC-001 for an InformaCast
port.
Step 5 In the Description field, enter a description for the port. For example, InformaCast Recording Port
for Call Monitoring.
Step 6 From the Device Pool drop-down list, select ICVA.
Step 7 From the Calling Search Space drop-down list, select ICVA.
Step 8 From the Device Security Profile drop-down list, select Cisco CTI Port - Standard SCCP Non-Secure
Profile.
Step 9 Click Save.
Step 10 Click OK.
Step 11 In the left association area, click Line [1] - Add a new DN.
Step 12 In the Directory Number field, enter a directory number. This directory number should not be used for any
purpose other than making paging calls. It should not be assigned to a phone and should not be within a
direct-inward-dialing range.
Step 13 In the Route Partition drop-down list, select the following ports:
• For InformaCast ports, select ICVA-CTIOutbound.
Step 14 In the Display (Internal Caller ID) text box, enter InformaCast.
Step 15 In the ASCII Display (Internal Caller ID) text box, enter InformaCast.
Step 16 Click Save.
Step 17 Repeat this procedure for each CTI port that you need.
What to do next
Configure Access Control Group with AXL Access

Perform this procedure to create an access control group that includes AXL access.
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Configure Application User for Paging
Procedure
Step 1 From Cisco Unified CM Administration, choose User Management > User Settings > Access Control
Group.
Step 3 In the Name text box, enter ICVA User Group.
Step 4 Click Save.
Step 5 From the Related Links drop-down list, select Back to Find/List and click Go.
Step 6 In the Roles column, click the i icon that corresponds to the new access control group.
Step 7 Click Assign Role to Group.
Step 8 Click Find.
Step 9 Select Standard AXL API Access check box, and click Add Selected.
Step 10 Click Save.
Configure Application User for Paging

Perform this procedure to configure an application user:
• For Basic Paging, configure an InformaCast application user.
Procedure
Step 3 In the User ID text box, enter a user ID for the application user. For example, ICVA InformaCast.
Step 4 Enter a password in the Password and Confirm Password fields.
Step 5 In the Available Devices list box, click the CTI ports that you created for your deployment and use the arrows
to move the devices to the Controlled Devices list box. For example, select ICVA-IC-001 for InformaCast
and ICVA-CA-001 for CallAware.
Step 6 Click the Add to Access Control Group.
Step 7 Click Find.
Step 8 Check the following check boxes (unless otherwise indicated, select these permissions for all application
users):
• ICVA User Group
• Standard CTI Allow Control of All Devices
• Standard CTI Allow Control of Phones supporting Connected Xfer and conf
• Standard CTI Allow Control of Phones supporting Rollover Mode
• Standard CTI Enabled
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Enable Web Access for a Phone
Step 10 Click Save.
Enable Web Access for a Phone

Perform this procedure in Basic Paging to enable web access for a Cisco Unified IP Phone. You can also use
a Common Phone Profile to enable web access for a group of phones that use that profile. For details, see
Enable Web Access for Common Phone Profile, on page 298.
Before you begin

Configure Application User for Paging, on page 297
Procedure

Step 2 Click Find and select the phone for which you want to enable web access.
Step 3 In the Product Specific Configuration Layout area, from the Web Access drop-down list, select Enabled.
Step 4 Click Save.
What to do next
Configure Authentication URL, on page 299
Enable Web Access for Common Phone Profile

Perform this procedure in Basic Paging to enable web access for a group of Cisco Unified IP Phones that use
a Common Phone Profile. You can also enable web access on an individual phone. For details, see Enable
Web Access for a Phone, on page 298.
Before you begin

Configure Application User for Paging, on page 297
Procedure
Step 1 From Cisco Unified CM Administration, chooseI Device > Device Settings > Common Phone Profile.
Step 2 Click Find and select the profile that applies to the group of phones for which you want to enable web access.
Step 3 In the Product Specific Configuration Layout area, from the Web Access drop-down list, select Enable.
Step 4 Click Save.
Step 5 Click Apply Config to reset the phones that use the Common Phone Profile.
Step 6 Click OK.
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Enable Web Access for Enterprise Phone Configuration
What to do next
Configure Authentication URL, on page 299
Enable Web Access for Enterprise Phone Configuration

Perform this procedure in Unified Communications Manager to enable web access for a group of Cisco Unified
IP Phone that use a Common Phone Profile. You can also enable web access on an individual phone. For
more details, see Enable Web Access for a Phone, on page 298.
Before you begin

Configure Application User for Paging, on page 297.
Procedure
Step 1 From Cisco Unified CM Administration, choose System > Enterprise Phone Configuration.
Step 2 From the Web Access drop-down list, select Enable.
Step 3 Click Save.
Step 4 Click Apply Config to reset the phones that use the Common Phone Profile.
Step 5 Click OK.
Configure Authentication URL

Perform the following tasks to configure an authentication URL that points to InformaCast so that when
InformaCast pushes broadcasts to Cisco Unified IP Phones, the phones authenticate with InformaCast instead
of Unified Communications Manager.
Procedure

Step 1 Set Authentication URL, on page 299 Set the Unified Communications Manager
authentication URL to point InformaCast.
Step 2 Reset Your Phones, on page 300 Reset the phones in your deployment so that
your phones use the new settings.
Step 3 Test Your Phones, on page 300 Verify that the phones in your deployment use
the new authentication URL settings.
Set Authentication URL

Perform this procedure to set the Unified Communications Manager authentication URL to point to the
InformaCast Virtual Appliance.
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Reset Your Phones
Procedure
Step 1 From Cisco Unified CM Administration, choose System > Enterprise Parameters.
Step 2 Scroll to the Phone URL Parameters area, and in the URL Authentication field, enter http://<IP
Address>:8081/InformaCast/phone/auth where <IP Address> is the IP Address of the InformaCast
Virtual Appliance.
Note Make a note of the existing URL in the URL Authentication field. You may need this when you
configure InformaCast. See your InformaCast documentation for details.
Step 3 Scroll to the Secured Phone URL Parameters area, and in the Secured Authentication URL field, enter
http://<IP Address>:8081/InformaCast/phone/auth where <IP Address> is the IP Address
of the InformaCast Virtual Appliance.
Step 4 Click Save.
Reset Your Phones

After you set the authentication URL to point to the InformaCast Virtual Applicance, you must reset your
phones. This procedures describes how to manually reset the phones in device pools. There are many methods
for resetting your phones. For example, you can also use Bulk Administration Tool to schedule the reset during
off hours. See the Cisco Unified Communications Manager Bulk Administration Guide for information on
the Bulk Administration Tool.
Procedure

Step 2 In the From Phone Where box, select Device Pool.
Step 3 Set the other drop-down menus and field items to settings that will bring up the device pools that you contain
your phones.
Step 4 Click Find.
Step 5 Select the device pools that you want to reset.
Step 6 Click Reset Selected.
Step 7 Click Reset.
Test Your Phones

Verify that your phones are authenticating with the InformaCast Virtual Appliance.
Procedure

Step 2 Use the drop-down list and fields in the Find and List Phones window to filter your search for a phone that
should be using the new authentication URL, and click Find.
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Advanced Notification Paging Configuration Task Flow
Step 3 For the phone that should be using the new settings, click the IP Address link in the IPv4 Address column.
Step 4 Click Network Configuration.
The Network Configuration page appears.
Step 5 Verify that the Authentication URL field displays the InformaCast Virtual Appliance IP address that you
entered for the URL Authentication enterprise parameter. If the correct URL does not appear, you will need
to set the authentication URL.
Advanced Notification Paging Configuration Task Flow

Perform the following tasks to integrate InformaCast Paging Server with Unified Communications Manager
for IP paging and emergency call alerting. It includes the following features:
• InformaCast advanced notification
• Panic button configuration
• Text and audio notification to IP phones when a user dials an emergency services number (CallAware)
Procedure

Step 1 Install the InformaCast Virtual Appliance, on Download the InformaCast OVA file from the
page 301. Singlewire website and upload it to vSphere.
Step 2 Configure Connection to InformaCast, on page Configure Unified Communications Manager

303. and InformaCast.
Step 3 Configure Panic Button, on page 304. Configure a panic button to send a text and
audio notification to IP phones.
Step 4 Configure CallAware Emergency Call Alerting, Configure emergency call text and audio
on page 306. notifications.
Install the InformaCast Virtual Appliance

Singlewire supports InformaCast Virtual Appliance on the VMware ESXi platform, which is managed through
the vSphere client.
Note To view a list of Singlewire-supported VMware ESXi versions, go to this URL: https://www.singlewire.com/
compatibility-matrix and click the Server Platforms link under InformaCast Platform section.
Note If you have purchased a license, refer to https://www.singlewire.com/icva-kb-activate to activate your license.
This will ensure that Emergency Notifications stay active after the 90-day trial.
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Install the InformaCast Virtual Appliance
Note For more details on the installation, including InformaCast screen captures, go to this URL:
https://www.singlewire.com/icva-kb-install.
Before you begin

Import InformaCast Virtual Appliance using the vSphere client. This can be downloaded from your VMware
server.
Procedure
Step 1 Download the OVA file from the Singlewire website and then log in to the vSphere client.
Note If you are using InformaCast on the Communications Manager Business Edition 6000, you are
supplied with a DVD in a package with an OVA on it (physical media).
The vSphere Client window appears.

Step 2 From the vSphere Client window, choose File > Deploy OVF Template.
The Deploy OVF Template dialog box appears.
Step 3 Click the Deploy from File radio button and then click Browse to select the saved the OVA file (or to the
OVA file on the supplied DVD). After you select the OVA file, click Open.
The Source location is selected in the Deploy OVF Template dialog box.
Step 4 Click Next to continue.
The Deploy OVF Template dialog box refreshes and OVF Template Details appears.
Step 5 Click Next to verify the Name and Location, and then click Next to select the network to store the new
virtual machine files.
Tip It is good practice to place the Virtual Appliance on the same VLAN as your Cisco Unified
Step 6 Click Next to continue, and then click Finish.

The InformaCast Virtual Appliance begins importing.
Step 7 From the vSphere Client window, click Hosts and Clusters icon and then select your host server.
The vSphere Client window refreshes.
Step 8 Click the Configuration tab and select the Virtual Machine Startup/Shutdown link in the Software section.
Step 9 Click the Properties link.
The Virtual Machine Startup and Shutdown dialog box appears.
Step 10 Check the Allow virtual machines to start and stop automatically with the system check box under System
Settings.
Step 11 Under Startup Order, scroll to the Manual Startup section and select your virtual machine (by default, this
is Singlewire InformaCast VM), and then move it from the Manual Startup section to the Automatic Startup
section, by using the Move Up button. After moving it, click OK.
The InformaCast Virtual Appliance starts and stops automatically with the server on which it is hosted. Now
you can turn on InformaCast's virtual machine and set its network configuration.
Step 12 Choose View > Inventory > VMs and Templates and then select your virtual machine.
Step 13 Choose the Inventory > Virtual Machine > Open Console
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Configure Connection to InformaCast
The Singlewire InformaCast VM console window appears.

Step 14 InformaCast configuration starts for the first time. During this configuration, perform the following tasks for
the InformaCast Virtual Appliance:
a) Accept Cisco End User License Agreement (EULA)
b) Accept Singlewire EULA
c) Set up hostname
d) Set up IP address, subnet mask, and default gateway
e) Set up DNS server IP address and domain name
f) Set up NTP server IP address or hostname
g) Set up time zone
h) Set up Secure Socket Layer (SSL) certificate parameters
i) Set up SSL subject alternate names (optional)
j) Set up the OS admin password
k) Set up the InformaCast and PTT (PushToTalk) admin password. This password is required to connect the
Cisco Unified Communications Manager and InformaCast in the Cisco Unified CM Administration,
Advanced Features > Emergency Notifications Paging.
l) Set up security passphrase for backup and communication
When your configuration is successful, the “Welcome to Singlewire InformaCast” message is displayed.
Step 15 Click Continue to work with Singlewire InformaCast.
Configure Connection to InformaCast

Use this procedure to load the InformaCast certificate to the Unified Communications Manager Tomcat trust
store.
Before you begin

Install the InformaCast Virtual Appliance, on page 301.
Procedure
Step 1 From Cisco Unified CM Administration, choose Advanced Features > Emergency Notifications Paging.
Step 2 In the Introduction to InformaCast Emergency Notifications page, click Next to continue.
The Installing the InformaCast Virtual Appliance page appears.
Step 3 In the Installing the InformaCast Virtual Appliance page, click Next to continue.
Note You should have successfully installed InformaCast Virtual Appliance to configure with the
The Connecting Cisco Unified Communications Manager and InformaCast page appears.
Step 4 In the IP address of InformaCast VM field, enter either IP address or Hostname.
Note By default, the username is stated as admin in the Username to use in InformaCast field, and
it is not editable.
Step 5 In the Password for admin app user field, enter the administrator password of the InformaCast application.
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Configure Panic Button
The dialog box displaying the thumbprint of InformaCast certificate is displayed.

Step 6 Click OK to load the InformaCast certificate to the Unified Communications Manager Tomcat trust store.
Configuration process starts.
Note When the configuration is successful, the Status field displays the completion status.
Step 7 Click Next.

The wizard performs the following tasks:
• Activates SNMP service
• Configures SNMP Service with locally generated random credentials
• Activates CTI Manager Service
• Configures Unified Communications Manager for InformaCast
• Creates new region (1 per cluster)
• Creates new device pool (1 per cluster)
• Creates SIP trunk (1 per cluster)
• Creates route group (1 per cluster)
• Creates route list
• Creates role
• Creates app user
• Configures InformaCast for Unified Communications Manager

• Creates a cluster
• Refreshes recipient groups
• Sets SIP access to deny
• Creates SIP access

Use this procedure to configure a panic button to send a text and audio notification to IP phones. This allows
you to initiate a one click alarm if there is emergency.
Before you begin

Configure Connection to InformaCast, on page 303.
Procedure
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Step 4 In the Connecting Cisco Unified Communications Manager and InformaCast page, click Next to continue.
The Configuring a Panic Button page appears.
Step 5 From the Choose pre-recorded message by name drop-down list, select the pre-recorded message to be
displayed on Cisco Unified IP phones and various devices and systems in emergency.
Note You can change the pre-recorded message in InformaCast administration, as required.
Step 6 In the Enter DN to trigger the panic button field, enter the Directory Number (DN), which includes the
digits 0 to 9, asterisks (*), and pound signs (#). Default value is ***5.
Step 7 From the Route Partition drop-down list, select a partition to restrict access to the route pattern.
Note If you do not want to restrict access to the route pattern, select <None> for the partition.
Step 8 Click Choose Phones to Send Notification button.

The Phones to Send Notification dialog box appears.
Step 9 From the Phones to Send Notification dialog box, select the Cisco Unified IP phones to send the pre-recorded
message. The dial pattern entered by you (for example, ***5) is configured as speed dial on the selected
phones.
The selected Cisco Unified IP Phone are displayed in the Selected Phones to Send Notification list box.
Step 10 Click Add Rules, to create a new rule for the selected Cisco Unified IP Phone to receive notifications.
a) Select one of the parameters from the drop-down list. The available options are Device Pool, Description,
and Directory Number.
b) In the second drop-down list, select a criteria from the following options:
• Does
• Does not
c) In the third drop-down list, select a criteria from the following options:
• Begins with
• Ends with
• Contains
d) In the text box, enter the search criterion.

Note Minimum of one new rule and maximum of new five rules can be created. The Add Rules
button gets disabled when five rules are created.
Note To delete a rule, click Delete Rules.
e) Click Test Rules, to validate the created rules. When the test rule is completed with more than zero phones,
the Next button is enabled.
Note Phones added to Cisco Unified Communications Manager at a later date that match this rule will
be included as recipients in notifications to this group.
Step 11 Click Next.
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Configure CallAware Emergency Call Alerting

• Adds a speed dial for the entered DN to the selected phones. If the selected phones have unused speed
dials assigned to existing phone button templates, this speed dial appears directly on the selected phones.
If the selected phones do not have unused speed dial buttons, the panic button speed dial is created, but
it does not appear on the phone.
• Adds route pattern for entered DN in selected partition using created route list.
• Creates an InformaCast DialCast entry for the entered DN to send the selected message to the phones
matching the selected rules.
Configure CallAware Emergency Call Alerting

Use this procedure to configure the CallAware emergency call alerting details. This sends a text and audio
notification to IP phones when an emergency number is dialed. It can also detect calls to numbers other than
911.
Before you begin

Configure Panic Button, on page 304.
Procedure
Step 4 In the Connecting Cisco Unified Communications Manager and InformaCast page, click Next to continue.
Step 5 In the Configuring a Panic Button page, click Next to continue.
The Configuring CallAware Emergency Call Alerting page appears.
Step 6 From the Choose pre-recorded message by name drop-down list, select the pre-recorded message to be
displayed on Cisco Unified IP phones and various devices and systems in emergency.
Note You can change the pre-recorded message in InformaCast administration, as required.
Step 7 Click Choose Emergency Route Patterns button.

The Route Patterns dialog box appears.
Step 8 From the Route Patterns dialog box, select the route patterns by checking the box next to the desired patterns.
a) Click the Save Selected/Changes button.
The selected route patterns are displayed in the Selected Route Patterns list box.
Step 9 Click Add Rules, to create a new rule for the selected Cisco Unified IP Phone to receive notifications.
a) Select one of the parameters from the drop-down list. The available options are Device Pool, Description,
and Directory Number.
b) In the second drop-down list, select a criteria from the following options:
• Does
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Paging Interactions
• Does not
c) In the third drop-down list, select a criteria from the following options:
• Begins with
• Ends with
• Contains
d) In the text box, enter the search criterion.

Note Minimum of one new rule and maximum of five new rules can be created. The Add Rules
button gets disabled when five rules are created.
Note To delete a rule, click Delete Rules.
e) Click Test Rules, to validate the created rules. When the test rule is completed with more than zero phones,
the Finish button is enabled.
Note Phones added to Unified Communications Manager at a later date that match this rule will be
included as recipients in notifications to this group.
Step 10 Click Finish.

• Adds External Call Control profile for InformaCast
• For each selected route pattern, modify that route pattern to reference the External Call Control profile
• Creates a recipient group with rules that match phones to receive the notification
• Creates an InformaCast routing request with the selected message and recipient group
The Summary page appears and confirms the successful configuration of InformaCast with Unified
Communications Manager. For more information, see https://www.singlewire.com.
Paging Interactions
• Advanced Notification Paging Interactions, on page 308
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Advanced Notification Paging Interactions
Advanced Notification Paging Interactions

Table 26: Advanced Notification Paging Interactions
Feature Interaction
Emergency Notifications Paging You can configure the Emergency Notifications Paging wizard
using InformaCast Release 11.5(1)SU3 and later versions in
basic paging mode only.
You can configure call monitoring to route patterns that
contain digits only in the Emergency Notifications Paging
wizard. For route patterns that contain wildcard characters,
configure in InformaCast.
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CHAPTER 26
Intercom
• Intercom Overview, on page 309
• Intercom Prerequisites, on page 310
• Intercom Configuration Task Flow, on page 310
• Intercom Interactions, on page 313
• Intercom Restrictions, on page 315
• Intercom Troubleshooting, on page 316
Intercom Overview
Intercom is a type of phone line that combines the functionality of a traditional line and a speed dial. With an
intercom line, a user can call the intercom line of another user, which answers automatically to one-way audio
whisper. The recipient can then acknowledge the whispered call and initiate a two-way intercom call.
You can use an intercom line to dial any other intercom line in the intercom partition, or you can preconfigure
the line to target an intercom line outside the intercom partition.
Intercom allows a user to place a call to a predefined target. The called destination answers the call automatically
in speakerphone mode with mute activated. This sets up a one-way voice path between the initiator and the
destination, so the initiator can deliver a short message, regardless of whether the called party is busy or idle.
To ensure that the voice of the called party does not get sent back to the caller when the intercom call is
automatically answered, Unified Communications Manager implements whisper intercom. Whisper intercom
ensures that only one-way audio exists from the caller to the called party. The called party must manually
press a key to talk to the caller.
An auto-answer tone indicates the beginning of the whisper intercom state for both the sender and the recipient.
Intercom and Default Devices

Each intercom line needs a default device. The intercom line is displayed only on the designated default
device.
When the administrator assigns an intercom line to a device, the system sets the device as the default device
for the intercom line if not set previously. The administrator can modify the default device for the intercom
line. When the administrator changes the default device to a different device, the intercom line gets removed
from the original device, even though the intercom line may still be assigned to the original device.
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Intercom Prerequisites
You can assign an intercom line to a device profile. Only when a user uses a device profile to log in to the
default device that matches the default device of the intercom line does the intercom line become available.
Otherwise, no intercom line is displayed when the user logs in.
Intercom Prerequisites
The intercom feature has the following system requirements:
• Cisco Unified IP Phones Firmware Release 8.3(1) or later
Intercom Configuration Task Flow

Before you begin
• Review Intercom Prerequisites, on page 310.
Procedure

Step 1 Configure Intercom Partition, on page 310 To add a new Intercom partition or configure
an existing partition.
Step 2 Configure an Intercom Calling Search Space, To add a new Intercom Calling Search Space.
on page 311
Step 3 Configure an Intercom Translation Pattern, on To add a new Intercom Translation Pattern or
page 312 to configure an existing Intercom Translation
Pattern .
Step 4 Configure an Intercom Directory Number, on To add or update an Intercom Directory
page 312 Number.
Step 5 Intercom Line and Speed Dial Configuration, Configure Intercom Line and Speed Dial.
on page 313
Configure Intercom Partition

Before you begin
Ensure the phone model supports the Intercom feature for a particular release and device pack Generate a
Phone Feature List, on page 1
Procedure
Step 1 From Cisco Unified CM Administration, choose Call Routing > Intercom > Intercom Route Partition.
The Find and List Intercom Partitions window appears.
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An Add New Intercom Partition window appears.
Step 3 Under the Intercom Partition Information section, in the Name box, enter the name and description of the
intercom partition that you want to add.
Note To enter multiple partitions, use one line for each partition entry. You can enter up to 75 partitions;
the names and descriptions can have up to a total of 1475 characters. The partition name cannot
exceed 50 characters. Use a comma (,) to separate the partition name and description on each
line. If a description is not entered, Unified Communications Manager uses the partition name
as the description.
Step 4 Click Save.

Step 5 Locate the partition that you want to configure.
Intercom Partition Configuration window is displayed
Step 6 Configure the fields in the Intercom Partition Configuration field area. See the online help for more information
Step 7 Click Save.
The Intercom Partition Configuration window appears.
Step 8 Enter the appropriate settings. For detailed information about the Intercom Partition Configuration parameters,
see online help.
Step 9 Click Save.

Before you begin
Configure Intercom Partition, on page 310
Procedure
Step 1 In the menu bar, choose Call Routing > Intercom > Intercom Calling Search Space.
Step 2 Click the Add New.
Step 3 Configure the fields in the Intercom Calling Search Space field area. For more information on the fields and
Step 4 Click Save.
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Placing Calls

Before you begin
Configure an Intercom Calling Search Space, on page 311
Procedure
Step 1 Choose Call Routing > Intercom > Intercom Translation Pattern.
The Find and List Intercom Translation Patterns window appears.

a) To copy an existing intercom translation pattern, locate the partition to configure, click Copy eside the
intercom translation pattern to copy.
b) To add a new intercom translation pattern, click the Add New.
Step 3 Configure the fields in the Intercom Translation Pattern Configuration field area. For more information on
Step 4 Click Save.
Ensure that the intercom translation pattern that uses the selected partition, route filter, and numbering plan
combination is unique. if you receive an error that indicates duplicate entries, check the route pattern or hunt
pilot, translation pattern, directory number, call park number, call pickup number, or meet-me number
configuration windows.
The Intercom Translation Pattern Configuration window displays the newly configured intercom translation
pattern.
Configure an Intercom Directory Number

You can assign patterns to intercom directory numbers; for example, 352XX. To avoid user confusion, when
you assign a pattern to an intercom directory number, add text or digits to these intercom DN configuration
fields, Line Text Label, Display (Internal Caller ID), and External Phone Number Mask. These fields are
displayed for an intercom directory number only after you add the intercom directory number and you associate
the intercom directory number with a phone.
For example, add the username to the line text label and internal caller ID, and add the outside line number
to the external number mask, when the calling information is displayed, it says John Chan, not 352XX.
Procedure
Step 1 Choose Call Routing > Intercom > Intercom Directory Number.
The Find and List Intercom Directory Numbers window is displayed.
Step 2 To locate a specific intercom directory number, enter search criteria and click Find.
A list of intercom directory numbers that match the search criteria displayed.
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Placing Calls
Intercom Line and Speed Dial Configuration

a) To add an intercom directory number, click Add New.
b) To update an intercom directory number, click the intercom directory number to update.
The Intercom Directory Number Configuration window displayed.
Step 4 Configure the fields in the Intercom Directory Number Configuration field area. For more information on the
Step 5 Click Save.
Step 7 Click Reset Phone.
Step 8 Restart devices.
During the restart, the system may drop calls on gateways.
Intercom Line and Speed Dial Configuration

Before you begin
Configure an Intercom Directory Number, on page 312
Procedure
Step 1 Choose Device > Device Settings > Phone Button Template and add the intercom line to an existing phone
button template or create a new template.
Note The intercom line cannot be configured as the primary line.
Step 2 From the Button Information area, from Feature drop-down list, choose Intercom.
Step 3 From the Button Information area, from Feature drop-down list, choose Speed Dial.
Note You can configure the intercom line with a predefined destination (speed dial) to allow fast access.
Step 4 Click Save.

Intercom Interactions
Feature Interaction
Bulk Administration Tool The Unified Communications Manageradministrator can use the Bulk
Administration Tool to add many intercom users at once instead of adding users
individually. For more information, see Bulk Administration Guide for Cisco
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Placing Calls
Intercom Interactions
Feature Interaction
Barge When the intercom destination is a barge target, the Cisco Unified IP Phone can
still support whisper intercom.
When the destination caller chooses to talk to the intercom caller by pressing the
intercom button, the original call is put on hold, and the barge initiator is released.
Do Not Disturb (DND) The intercom call will override DND on the destination phone.
Call Preservation When a call is preserved, the end user must hang up before the phone can
reregister with Unified Communications Manager.
When the intercom call is in whisper mode, it represents a one-way medium, and
the terminating side might have no user at all; therefore, only the intercom call
in talkback mode will get preserved. (Whisper intercom will not get preserved.)
Cisco Unified Survivable When Cisco Unified IP Phones register with SRST, the phones do not register
Remote Site Telephony intercom lines; therefore, the feature will not be available when the phones are
(SRST) registered with SRST.
Cisco Unified With the Unified Communications Manager Assistant Configuration Wizard,
Communications Manager Cisco Unified Communications Manager Assistant configuration takes less
Assistant time and eliminates errors. The partitions, calling search spaces, route point, and
translation pattern automatically get created when the administrator successfully
runs and completes the configuration wizard.
CTI You can use CTI/JTAPI/TSP to set or modify the preconfigured target directory
number for an intercom line. You will receive notification if the target directory
number is updated or reconfigured through Cisco Unified Communications
Manager Administration.
Be aware that CTI/JTAPI/TSP is backward compatible if the intercom line is not
configured to be controlled by the application. If the intercom line is configured
in the application user list, you may have to make changes and test the
compatibility.
Cisco Extension Mobility The intercom feature interacts with Cisco Extension Mobility. The system presents
an intercom line to a user who uses Cisco Extension Mobility to log in to a phone
that supports the feature if the device profile that the user uses to log in has an
intercom line that is provisioned. The phone must be the default device for that
intercom line.
Internet Protocol Version Intercom can support phones with an IP Addressing Mode of IPv4 Only or IPv4
6 (IPv6) and IPv6. During an intercom call, the talkback mode establishes media streams
with the same IP version as the media stream that is used when the caller initiates
intercom.
Intercom directory Intercom directory numbers (lines) are restricted to one device per intercom line.
numbers (lines) Cisco Extension Mobility is widely used; mobile users need the intercom feature
but need it to be available only on a single device. You can assign intercom lines
to either a regular device or to an extension mobility profile, but the system needs
to enforce that an intercom line gets associated to either a regular device or to an
extension mobility profile.
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Placing Calls
Intercom Restrictions
Feature Interaction
Extension mobility profile An extension mobility profile can be used on more than one phone simultaneously,
use the Default Activated Device field in the Intercom Directory Number
Configuration window ( Cisco Unified CM Administration > Call Routing >
Intercom > ntercom Directory Number Configuration) to specify which device
can display this intercom line. Intercom lines that are not used for Extension
Mobility also require configuration of the Default Activated Device field.
Intercom Restrictions
The following restrictions apply to the Intercom feature:
Hold The system does not allow intercom calls to be placed on hold.
Call Forwarding Intercom calls cannot be forwarded.
Transfer The system does not allow an intercom call to be transferred.
iDivert The system does not allow an intercom call to be diverted.
Call Pickup/Directed Call The call pickup groups do not include intercom calls.
Pickup
DND Intercom overrides Do Not Disturb (DND).
Bandwidth If sufficient bandwidth does not exist, the intercom call fails.
Call Target If two intercom calls are directed to a target, the first one goes through; the second
fails with a busy tone.
Barge and cBarge Intercom does not work with Barge and cBarge.
Conferencing The system does not allow intercom calls to be in conference.
Monitoring and Recording When an active call is being monitored or recorded, the user cannot receive nor
place intercom calls.
Video Video is not supported with intercom.
Intercom Partition An intercom partition assigned to an item such as calling search space or to a
route pattern cannot be deleted.
Intercom Calling Search Intercom calling search spaces that devices, lines (DNs), translation patterns, or
Spaces other items are using cannot be deleted.
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Placing Calls
Intercom Troubleshooting
Intercom Troubleshooting
Busy Tone When Dialing Out of Intercom Line
Problem
Phone plays busy tone when user is dialing out of intercom line.
Possible Cause
The DN is not in the same intercom partition as the calling number.
Solution
• Ensure that the DN is in the same intercom partition as the calling number.
•I
f it is, ensure that the dialed-out DN is configured on another phone and that the phone is registered with
the same Unified Communications Manager cluster.
Intercom Calls cannot use Talkback with Speaker, Handset or Headset

Problem
User cannot go into talkback mode for intercom calls by using headset, handset, or speaker.
Possible Cause
This situation exists by design. The only way to go into the connected state for intercom calls is by pressing
the corresponding line button.
Solution
User can end call by using speaker, handset, or headset.
Troubleshooting SCCP
Intercom Lines Not Showing Up on Phone
Problem
Intercom lines do not display on the phone.
Possible Cause
The phone version may be earlier than 8.3(1), or the button template may not be assigned to the phone.
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Placing Calls
Intercom Lines Not Showing Up When Phone Falls Back to SRST
Solution
• Check the phone version. Ensure that it is 8.3(1) or later.
• Determine whether the button template is assigned to the phone.
• Capture the sniffer trace between Cisco Unified Communications Manager and the phone. In the button
template response, see whether intercom lines get sent to the phone (button definition = Ox17).
Intercom Lines Not Showing Up When Phone Falls Back to SRST
Problem
A phone that was configured with Unified Communications Manager Release 6.0(x) or later, includes two
intercom lines. Unified Communications Manager stops and falls back to SRST. The intercom lines do not
display.
Possible Cause
The SCCP version of SRST does not support SCCP Version 12.
Solution
• Check the SCCP Version of SRST. If SRST supports SCCP Version 12, it will support intercom lines.
• If SRST supports SCCP Version 12, capture a sniffer trace and ensure that the button template that the
phone sent includes intercom lines.
Troubleshooting SIP
Debug Phones That Are Running SIP
Use this debug command: Debug sip-messages sip-task gsmfsmIsm sip-adapter.
Configuration of Phones That Are Running SIP

Show config —The command on the phone is displayed if intercom lines are configured as regular lines with
featureid-->23.
Cisco Extension Mobility User Is Logged In But Intercom Line Does Not Display
Problem
The Cisco Extension Mobility user is logged in to a phone, but the user intercom line does not display.
Possible Cause
Default Activated Device is configured incorrectly.
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Placing Calls
Intercom Line Fails to Display on Phone
Solution
• Check that the Default Activated Device is configured on the intercom directory number.
• Check that the Default Activated Device matches the device to which the user is logged in.
Intercom Line Fails to Display on Phone

Problem
An intercom line has been configured and assigned to a phone but fails to display on the phone.
Possible Cause
Default Activated Device value is set to the intercom line of this device.
Solution
If the configuration has been done, reset the phone.
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PA R T X
Receiving Calls
• Prime Line Support , on page 321
• Call Forwarding , on page 325
• Call Pickup , on page 349
• Call Park and Directed Call, on page 371
• Extension Mobility , on page 397
• Extension Mobility Cross Cluster , on page 413
• Extension Mobility Roaming Across Clusters, on page 445
• Hold Reversion , on page 459
• Accessing Hunt Groups , on page 465
• Malicious Call Identification , on page 473
• Call Transfer , on page 483
• External Call Transfer Restrictions , on page 497
CHAPTER 27
Prime Line Support
• Prime Line Support Overview, on page 321
• Prime Line Support Prerequisites, on page 321
• Prime Line Support Configuration Task Flow, on page 321
• Prime Line Support Interactions, on page 323
• Prime Line Support Troubleshooting, on page 324
Prime Line Support Overview

You can configure the Prime Line Support in Cisco Unified CM Administration so that when the phone is
off-hook and receives a call on any line, the system always chooses the primary line for the call.
Prime Line Support Prerequisites

The following devices are compatible with the Prime Line Support feature:
Cisco Unified IP Phone 7900 Series, 8900 Series, and 9900 Series
For more information on the supported devices, see the latest version of Cisco Unified IP Phone Guide and
Cisco Unified IP Phone Administration Guide.
Prime Line Support Configuration Task Flow

To configure the Prime Line Support feature for either the Cisco CallManager service or devices and device
profiles, perform one of the following procedures.
Before you begin

• Review Prime Line Support Prerequisites, on page 321.
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Receiving Calls
Configure Clusterwide Prime Line Support
Procedure

Step 1 Configure Clusterwide Prime Line Support, on (Optional). Configure the Prime Line Support
page 322 feature for the Cisco CallManager service,
which applies to the entire cluster.
Step 2 Configure Prime Line Support for Devices, on (Optional). Configure the Prime Line Support
page 323 feature for specific devices within the cluster,
if you do not want to enable the feature
clusterwide.
Note When you configure this
parameter, going off-hook makes
only the first line active on the
phone, even when a call rings on
another line on the phone. So the
call does not get answered on the
other line.
Configure Clusterwide Prime Line Support

Procedure
Step 4 From the Always Use Prime Line clusterwide service parameter, choose one of the following options from
the drop-down list:
• True- When a phone goes off-hook, the primary line gets chosen and becomes the active line.
• False- When a phone goes off-hook, the IP phone automatically chooses an available line as the active
line.
The default value for this service parameter is False.
Step 5 For this change to take effect on the SIP phones, click the ApplyConfig button in Cisco Unified CM
Administration (for example, on the Device Configuration window, the Device Pool Configuration window,
or any other window on which ApplyConfig is an option).
Note If the new configuration is not applied on the SIP phones, the SIP Prime Line Support feature
changes will not be implemented until the next reset of the Cisco CallManager service or reset
of each affected device.
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Configure Prime Line Support for Devices
Configure Prime Line Support for Devices

Procedure
Step 1 From Cisco Unified CM Administration, choose Device > Common Phone Profile.
Step 2 From the Find and List window, choose the phone for which you want to change the Always Use Prime Line
setting.
The Phone Configuration window appears.
Step 3 From the Always Use Prime Line drop-down list, choose one of the following options:
• Off- When the phone is idle and receives a call on any line, the phone user answers the call from the line
on which the call is received.
• On- When the phone is idle (off hook) and receives a call on any line, the primary line is chosen for the
call. Calls on other lines continue to ring, and the phone user must select those other lines to answer these
calls.
• Default- Unified Communications Manager uses the configuration from the Always Use Prime Line
service parameter, which supports the Cisco CallManager service.
Step 4 Click Save.
Prime Line Support Interactions

Feature Interaction
Always Use Prime Line If you select On for the Always Use Prime Line
parameter in the Device Profile or Default Device
Profile Configuration window, a Cisco Extension
Mobility user can use this feature after logging in to
the device that supports Cisco Extension Mobility.
Maximum Number of Calls and Busy Trigger Settings When the phone already has a call on a line, Unified
Communications Manager uses the configuration for
the Maximum Number of Calls and Busy Trigger
settings to determine how to route the call.
Auto Answer If you choose the Auto Answer with Headset option
or Auto Answer with Speakerphone option from the
Auto Answer drop-down list in Cisco Unified CM
Administration, the Auto Answer configuration
overrides the configuration for the Always Use Prime
Line parameter.
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Prime Line Support Troubleshooting
Prime Line Support Troubleshooting

Prime Line Support Does Not Work When Set To True
Problem When the cluster-wide service parameter Always use Prime Line is set to True and the IP phone
goes off-hook, the primary line becomes the active line. Even if a call rings on the second line, when the user
goes off-hook, it activates only the first line. The phone does not answer the call on the second line. However,
when IP phones with multiple line appearances are used with the 7.1.2 phone load, the phone does not use
the primary line when a second line rings. If the user picks up the handset, the phone answers the call on the
second line.
Solution Press the line button for the primary line so that the secondary line is not engaged when a call is
initiated.
Unable To Answer Inbound Calls

Problem The users are unable to automatically answer inbound calls after they go off-hook on IP phones, and
must press the Answer softkey to answer the calls.
Solution To resolve the problem, perform the following procedure:
1. From Cisco Unified CM Administration, choose System > Service Parameters.

2. From the Server drop-down list, choose the server that is running the Cisco CallManager service.
3. From the Service drop-down list, choose Cisco CallManager.
4. In Cluster wide parameters (Device - phone), set Always Use Prime Line to False.
Inbound Calls Are Answered Automatically

Problem When an inbound call is received on a shared line of an IP phone, the call is answered immediately
as the handset is lifted, without the option to either answer the call or make an outbound call. This behavior
does not change even though Auto Line Select is set to disabled.
Solution To resolve the problem, perform the following procedure:
1. From Cisco Unified CM Administration, choose System > Service Parameters.

2. From the Server drop-down list, choose the server that is running the Cisco CallManager service.
3. From the Service drop-down list, choose Cisco CallManager.
4. In Cluster wide parameters (Device - phone), set Always Use Prime Line to False.
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CHAPTER 28
Call Forwarding
• Call Forwarding Overview, on page 325
• Call Forwarding Configuration Task Flow, on page 327
• Call Forwarding Interactions, on page 343
• Call Forwarding Restrictions, on page 347
Call Forwarding Overview

As a user, you can configure a Cisco Unified IP Phone to forward calls to another phone. The following call
forwarding types are supported:
• Call Forward No Bandwidth—Forwards calls when a call to a directory number fails due to insufficient
bandwidth, and provides forwarding functionality to an Automated Alternate Routing (AAR) destination
using public switched telephone network (PSTN) as the alternate route or to a voicemail system.
• Call Forward with Alternate Destination—Forwards calls when a call to a directory number and the
forwarded destination are not answered. The call gets diverted to an alternate destination as a last resort.
This Call Forwarding type is also referred to as “MLPP Alternate Party destination.”
• Call Forward All (CFA)—Forwards all calls to a directory number.
• Call Forward Busy (CFB)—Forwards calls only when the line is in use and the configured Call Forward
Busy trigger value is reached.
• Call Forward No Answer (CFNA)—Forwards calls when the phone is not answered after the configured
No Answer Ring Duration timer is exceeded or the destination is unregistered.
• Call Forward No Coverage (CFNC)—Forwards calls when the hunt list is exhausted or timed out, and
the associated hunt-pilot for coverage specifies “Use Personal Preferences” for its final forwarding.
• Call Forward Unregistered (CFU)—Forwards calls when the phone is unregistered due to a remote
WAN link failure, and provides automated rerouting through the Public Switched Telephone Network
(PSTN). Calls can also be forwarded based on the type of caller: internal or external.
• CFA Destination Override—Forwards calls when the user to whom calls are being forwarded (the
target) calls the user whose calls are being forwarded (the initiator). The phone of the initiator rings
instead of call forwarding back to the target.
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Receiving Calls
Call Forward All, Including CFA Loop Prevention and CFA Loop Breakout
Call Forward All, Including CFA Loop Prevention and CFA Loop Breakout
Call Forward All (CFA) allows a phone user to forward all calls to a directory number.
You can configure CFA for internal and external calls and can forward calls to a voicemail system or a dialed
destination number by configuring the calling search space (CSS). Unified Communications Manager includes
a secondary Calling Search Space configuration field for CFA. The secondary CSS for CFA combines with
the existing CSS for CFA to allow support of the alternate CSS system configuration. When you activate
CFA, only the primary and secondary CSS for CFA are used to validate the CFA destination and redirect the
call to the CFA destination. If these fields are empty, the null CSS is used. Only the CSS fields that are
configured in the primary CSS for CFA and secondary CSS for CFA fields are used. If CFA is activated from
the phone, the CFA destination is validated by using the CSS for CFA and the secondary CSS for CFA, and
the CFA destination gets written to the database. When a CFA is activated, the CFA destination always gets
validated against the CSS for CFA and the secondary CSS for CFA.
Unified Communications Manager prevents CFA activation on the phone when a CFA loop is identified. For
example, Unified Communications Manager identifies a call forward loop when the user presses the CFwdALL
softkey on the phone with directory number 1000 and enters 1001 as the CFA destination, and 1001 has
forwarded all calls to directory number 1002, which has forwarded all calls to directory number 1003, which
has forwarded all calls to 1000. In this case, Unified Communications Manager identifies that a loop has
occurred and prevents CFA activation on the phone with directory number 1000.
Tip If the same directory number exists in different partitions, for example, directory number 1000 exists in
partitions 1 and 2, Unified Communications Manager allows the CFA activation on the phone.
CFA loops do not affect call processing because Unified Communications Manager supports CFA loop
breakout, which ensures that if a CFA loop is identified, the call goes through the entire forwarding chain,
breaks out of the Call Forward All loop, and the loop is completed as expected, even if CFNA, CFB, or other
forwarding options are configured along with CFA for one of the directory numbers in the forwarding chain.
For example, the user for the phone with directory number 1000 forwards all calls to directory number 1001,
which has forwarded all calls to directory number 1002, which has forwarded all calls to directory number
1000, which creates a CFA loop. In addition, directory number 1002 has configured CFNA to directory number
1004. The user at the phone with directory number 1003 calls directory number 1000, which forwards to 1001,
which forwards to 1002. Unified Communications Manager identifies a CFA loop, and the call, which breaks
out of the loop, tries to connect to directory number 1002. If the No Answer Ring Duration timer expires
before the user for the phone with directory number 1002 answers the call, Unified Communications Manager
forwards the call to directory number 1004.
For a single call, Unified Communications Manager may identify multiple CFA loops and attempt to connect
the call after each loop is identified.
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Receiving Calls
Call Forwarding Configuration Task Flow
Call Forwarding Configuration Task Flow

Procedure

Step 1 Configure Partitions for Call Forwarding, on Administrators can configure partitions to
page 327 restrict Call Forwarding to certain numbers
based on the design criteria and requirements.
Step 2 Configure Calling Search Space for Call Administrators can configure calling search
Forwarding, on page 329 spaces to restrict Call Forwarding to certain
numbers based on the design criteria and
requirements.
Step 3 Configure Call Forwarding when Hunt List is You can forward a call when hunting fails (that
Exhausted or Hunt Timer Expires, on page 330 is, when hunting is terminated without any hunt
party answering, either because no hunt number
from the list picked up or because the hunt timer
timed out).
Step 4 Configure Call Forward No Bandwidth, on page You can forward a call to an Automated
332 Alternate Routing (AAR) destination using
public switched telephone network (PSTN) as
the alternate route or to a voicemail system
when a call to a called directory number fails
due to insufficient bandwidth.
Step 5 Configure Call Forward Alternate Destination, You can forward calls that go unanswered to
on page 333 the directory number and the forwarded
destination. Calls will get diverted to an
alternate destination as a last resort.
Step 6 Configure Other Call Forwarding Types, on You can configure additional forwarding types
page 334 such as CFA, CFB, CFNA, CFNC, and CFU.
You can configure all these forwarding types
from the Directory Number Configuration
window.
Step 7 Enable Destination Override for Call Administrators can override the CFA when the
Forwarding, on page 343 target of the CFA calls the initiator of the CFA.
This allows the CFA target can reach the
initiator for important calls.
Configure Partitions for Call Forwarding

Configure partitions to create a logical grouping of directory numbers (DNs) and route patterns with similar
reachability characteristics. Partitions facilitate call routing by dividing the route plan into logical subsets that
are based on organization, location, and call type. You can configure multiple partitions.
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Receiving Calls
Partition Name Guidelines for Call Forwarding
Configure partitions to restrict call forwarding to certain numbers based on your design criteria and
requirements.
Procedure
name in this field.
incoming call.
Step 8 Click Save.
Partition Name Guidelines for Call Forwarding

2 characters 340
3 characters 256
4 characters 204
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Receiving Calls
Configure Calling Search Space for Call Forwarding
5 characters 172
... ...
10 characters 92
15 characters 64
Configure Calling Search Space for Call Forwarding

Configure calling search spaces to restrict Call Forwarding to certain numbers based on your design criteria
and requirements.
Before you begin

Configure Partitions for Call Forwarding, on page 327
Procedure

Partitions box.
Step 8 Click Save.
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Receiving Calls
Configure Call Forwarding when Hunt List is Exhausted or Hunt Timer Expires
Configure Call Forwarding when Hunt List is Exhausted or Hunt Timer Expires
The concept of hunting differs from that of call forwarding. Hunting allows Unified Communications Manager
to extend a call to one or more lists of numbers, where each list specifies a hunting order that is chosen from
a fixed set of algorithms. When a call extends to a hunt party from these lists and the party fails to answer or
is busy, hunting resumes with the next hunt party. (The next hunt party varies depending on the current hunt
algorithm.) Hunting then ignores the Call Forward No Answer (CFNA), Call Forward Busy (CFB), or Call
Forward All (CFA) configured values for the attempted party.
Call Forwarding allows detailed control as to how to extend (divert or redirect) a call when a called party fails
to answer, or is busy and hunting is not taking place. For example, if the CFNA value for a line is set to a
hunt-pilot number, a call to that line that is not answered diverts to the hunt-pilot number and begins hunting.
Before you begin

Configure Calling Search Space for Call Forwarding, on page 329
Procedure
Step 1 From Cisco Unified CM Administration, choose Call Routing > Route/Hunt > Hunt Pilot.
The Find and List Hunt Pilots window is displayed.
Step 2 Click Find.
A list of configured Hunt Pilots is displayed.
Step 3 Choose the pattern for which you want to configure call treatment when hunting fails.
The Hunt Pilot Configuration window is displayed.
Step 4 Configure the fields in the Hunt Pilot Configuration for the Hunt Call Treatment Settings area. For more
information on the fields and their configuration options, see Online Help.
Step 5 Click Save.
Hunt Call Treatment Fields for Call Forwarding

Field Description
Hunt Call Treatment Settings

Note Forward Hunt No Answer or Forward Hunt Busy fields are designed to move calls through
the route list. Queuing is used to hold callers in a route list. Therefore, if queuing is enabled,
both Forward Hunt No Answer and Forward Hunt Busy are automatically disabled.
Conversely, if Forward Hunt No Answer or Forward Hunt Busy are enabled, queuing is
automatically disabled.
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Hunt Call Treatment Fields for Call Forwarding
Field Description
Forward Hunt No Answer When the call that is distributed through the hunt list is not answered in a specific
period of time, this field specifies the destination to which the call gets forwarded.
Choose one of the following options:
• Do Not Forward Unanswered Calls
• Forward Unanswered Calls to
• Destination—Enter a directory number to which calls must be
forwarded to.
• Calling Search Space—Choose a calling search space from the
drop-down list which applies to all devices that use this directory
number.
• Maximum Hunt Timer—Enter a value (in seconds) that specifies the

maximum time for hunting without queuing.
Valid values are 1 to 3600. The default value is 1800 seconds (30 minutes).
Caution Do not specify the same value for the Maximum Hunt Timer
and the RNA Reversion Timeout on the associated line group.
The forward no answer timer should be greater than the RNA
timer of the line group.
The forward no answer timer should not be multiples of RNA
timer of line group.
This timer is canceled if either a hunt member answers the call or the hunt
list gets exhausted before the timer expires. If you do not specify a value
for this timer, hunting continues until a hunt member answers or the hunt
list is exhausted. If neither event takes place, hunting continues for 30
minutes, after which the call is received for final treatment.
Note If hunting exceeds the number of hops that the Forward Maximum
Hop Count service parameter specifies, hunting expires before the
30 minute maximum hunt timer value, and the caller receives a
reorder tone.
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Receiving Calls
Configure Call Forward No Bandwidth
Field Description
Forward Hunt Busy When the call that is distributed through the hunt list is not answered in a specific
period of time, this field specifies the destination to which the call gets forwarded.
Choose one of the following options:
• Do Not Forward Unanswered Calls
• Use Forward Settings of Line Group Member
• Forward Unanswered Calls to
• Destination—Enter a directory number to which calls must be
forwarded to.
• Calling Search Space—Choose a calling search space from the
drop-down list which applies to all devices that use this directory
number.
Configure Call Forward No Bandwidth

Before you begin
Configure Call Forwarding when Hunt List is Exhausted or Hunt Timer Expires, on page 330
Procedure
Step 1 From Cisco Unified CM Administration, choose Call Routing > Directory Number Configuration.
The Find and List Directory Numbers window is displayed.
Step 2 Click Find.
A list of configured directory numbers is displayed.
Step 3 Choose the directory number for which you want to configure call forward when there is insufficient bandwidth.
Step 4 Configure the fields In the AAR Settings area. See Directory Number Configuration Fields for Call Forwarding,
on page 332 for more information about the fields and their configuration options.
Step 5 Click Save.
Directory Number Configuration Fields for Call Forwarding

Field Description
Voice Mail Check this check box to forward the call to the
voicemail.
Note When you check this check box, Unified
Communications Manager ignores the
values in the Destination and Calling
Search Space fields.
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Configure Call Forward Alternate Destination
Field Description
AAR Destination Mask Enter a destination mask to determine the AAR

destination to dial instead of using the external phone
number mask.
AAR Group Choose an AAR group from the drop-down list. It

provides the prefix digits that are used to route calls
that are otherwise blocked due to insufficient
bandwidth. If you choose None, the server does not
attempt to reroute the blocked calls.
You can also configure this value in the Precedence
Alternate Party Timeout service parameter from
System > Service Parameters.
Retain this destination in the call forwarding By default, the directory number configuration retains
history the AAR leg of the call in the call history, which
ensures that the AAR forward to voicemail system
will prompt the user to leave a voice message.
If you check the check box, the AAR leg of the call
will be present in the call forwarding history.
Configure Call Forward Alternate Destination

Before you begin
Configure Call Forward No Bandwidth, on page 332
Procedure
Step 2 Click Find.
A list of configured directory numbers is displayed.
Step 3 Choose the directory number for which you want to configure an alternate destination.
Step 4 Configure the fields in the MLPP Alternate Party And Confidential Access Level Settings area. See MLPP
Alternate Party And Confidential Access Level Settings Fields for Call Forwarding, on page 334 for more
information about the fields and their configuration options.
Step 5 Click Save.
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MLPP Alternate Party And Confidential Access Level Settings Fields for Call Forwarding
MLPP Alternate Party And Confidential Access Level Settings Fields for Call Forwarding
Field Description
Target (Destination) Enter the number to which MLPP precedence calls

should be diverted if this directory number receives
a precedence call and neither this number nor its Call
Forward destination answers the precedence call.
Values can include numeric characters, octothorpe
(#), and asterisk (*).
MLPP Calling Search Space From the drop-down list, choose a calling search space
to associate with the MLPP alternate party target
(destination) number.
MLPP No Answer Ring Duration (seconds) Enter the number of seconds (between 4 and 60) after
which an MLPP precedence call will be directed to
this directory number alternate party, if this directory
number and its Call Forward destination have not
answered the precedence call.
You can also configure this value in the Precedence
Alternate Party Timeout service parameter from
System > Service Parameters from Cisco Unified
CM Administration.
Configure Other Call Forwarding Types

You can configure Call Forward All (CFA), Call Forward Busy (CFB), Call Forward No Answer (CFNA),
Call Forward No Coverage (CFNC), and Call Forward Unregistered (CFU) from the Directory Number
Configuration window.
Before you begin

• For Call Forwarding functionality to work as intended, Cisco recommends that for the configured phones
and the directory numbers in various partitions, the Call Forward Calling Search Spaces also be configured
or else the forwarding may fail. When a call is forwarded or redirected to the Call Forward destination,
the configured Call Forward Calling Search Space is used to forward the call.
• Configure Call Forward Alternate Destination, on page 333
Procedure
Step 2 Configure the Call Forwarding and Call Pickup Settings fields in the Directory Number Configuration
window to configure CFA, CFB, CFNA, CFNC, and CFU. See Call Forwarding Fields, on page 335 for
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Call Forwarding Fields
Step 3 Click Save.

Field Description
Call Forward and Call Pickup Settings
Calling Search Space Activation Three possible values exist for this option:
Policy
• Use System Default—The CFA CSS Activation Policy service parameter determines which Forward All Calling Search
Space to use for Call Forwarding. If the CFA CSS Activation Policy service parameter is set to With Configured CSS,
then Forward All Calling Search Space and secondary Calling Search Space for Forward All will be used for Call
Forwarding. This is the default setting.
• With Configured CSS—The Forward All Calling Search Space that is explicitly configured in the Directory Number
Configuration window controls the Forward All activation and Call Forwarding.
If the Forward All Calling Search Space is set to None, no CSS is configured for Forward All. A Forward All activation
attempt to any directory number with a partition will fail. No change in the Forward All Calling Search Space and secondary
Calling Search Space for Forward All occurs during the Forward All activation.
• With Activating Device/Line CSS—A combination of the Directory Number Calling Search Space and Device Calling
Search Space controls the Forward All activation and Call Forwarding without explicitly configuring a Forward All
Calling Search Space.
When Forward All is activated from the phone, the Forward All Calling Search Space and secondary Calling Search
Space for Forward All automatically gets populated with the Directory Number Calling Search Space and Device Calling
Search Space for the activating device.
If the Forward All Calling Search Space is set to None, and when Forward All is activated through the phone, the
combination of Directory Number Calling Search Space and activating Device Calling Search Space controls the Forward
All attempt.
CFA CSS Activation Policy—Ensure that you configure this service parameter correctly for Forward All to work as intended
in the Service Parameter Configuration window. The service parameter includes two possible values:
• With Configured CSS—The primary and secondary CFA Calling Search Space controls the Call Forwarding attempt.
• With Activating Device/Line CSS—The primary and secondary CFA Calling Search Space is updated with primary
Line Calling Search Space and activating Device Calling Search Space.
Roaming—When a device is roaming in the same device mobility group, Cisco Unified Communications Manager uses the
Device Mobility CSS to reach the local gateway. If a user sets Call Forward All at the phone, the CFA CSS is set to None, and
the CFA CSS Activation Policy is set to With Activating Device/Line CSS, then:
• The Device CSS and Line CSS is used as the CFA CSS when the device is in its home location.
• If the device is roaming within the same device mobility group, the Device Mobility CSS from the Roaming Device Pool
and the Line CSS is used as the CFA CSS.
• If the device is roaming within a different device mobility group, the Device CSS and Line CSS is used as the CFA CSS.
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Field Description
Forward All The fields in this row of fields specify the Call Forwarding treatment for calls to this directory number if the directory number
is set to forward all calls. The value in the Calling Search Space field is used to validate the Forward All destination that is
entered when the user activates Call Forward All from the phone. This field is also used to redirect the call to the Call Forward
All destination.
Configure the following values:
• Voice Mail—Check this check box to use the value that is set in the Voice Mail Profile Configuration window.
Note When this check box is checked, Unified Communications Manager ignores the values in the Destination
and Calling Search Space fields.
• Destination—This field indicates the directory number to which all calls are forwarded. Use any dialable phone number,
including an outside destination.
• Calling Search Space—This value applies to all devices that use this directory number.
• Forward Maximum Hop Count—Configure this parameter from the Cisco Unified CM Administrator, choose System >
Service Parameters.
This service parameter specifies the maximum number of times that a single call can be forwarded or diverted, and has
special considerations for QSIG calls. For an incoming QSIG call, the maximum value is 15 (per ISO specifications); if
you specify a greater value in this field, the specified value will apply to non-QSIG calls and for an incoming QSIG call,
the call will only divert a maximum of 15 times. When QSIG trunks are configured, Cisco recommends setting this
parameter to 15.
For example, if the value of this parameter is seven, and a Call Forward All chain occurs consecutively from directory
numbers 1000 to 007, which comprises seven hops, Cisco Unified Communications Manager prevents a phone user with
directory number 2000 from activating CFA to directory number 1000, because no more than seven forwarding hops are
supported for a single call.
Secondary Calling Search Space Because Call Forwarding is a line-based feature, in cases where the Device Calling Search Space is unknown, the system uses
for Forward All only the Line Calling Search Space to forward the call. If the Line Calling Search Space is restrictive and not routable, the
forward attempt fails.
Addition of a secondary calling search space for Call Forward All provides a solution to enable forwarding. The primary calling
search space for Call Forward All and secondary calling search space for Call Forward All get concatenated (primary CFA CSS
+ secondary CFA CSS). Unified Communications Manager uses this combination to validate the CFA destination and to forward
the call.
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Field Description
Forward Busy Internal The fields in this row of fields specify the forwarding treatment for internal calls to this directory number if the directory number
is busy. The values in the Destination and the Calling Search Space fields are used to redirect the call to the forward destination.
• Voice Mail—Check this check box to use the configured values in the Voice Mail Profile Configuration window for
internal calls.
Note When this check box is checked, the calling search space of the voicemail pilot is used. Unified
Communications Manager ignores the values in the Destination and the Calling Search Space fields.
Note When this check box is checked for internal calls, the system automatically checks the Voice Mail check
box for external calls. If you do not want external calls to be forwarded to the voicemail system, you must
uncheck the Voice Mail check box for external calls.
• Destination—This field indicates the Call Forward Busy destination for internal calls. Use any dialable phone number,
Note When you enter a destination value for internal calls, the system automatically copies this value to the
Destination field for external calls. If you want external calls to be forwarded to a different destination, you
must enter a different value in the Destination field for external calls.
• Calling Search Space—The Forward Busy Internal Calling Search Space is used to forward the call to the Forward Busy
Internal destination. It applies to all devices that use this directory number.
Note If the system is using partitions and calling search spaces, Cisco recommends that you configure the Call
Forward Calling Search Spaces. When a call is forwarded or redirected to the Call Forward destination, the
configured Call Forward Calling Search Space is used to forward the call. If the Calling Search Space field
is set to None, the forward operation fails if the system uses partitions and calling search spaces. For example,
if you configure the Forward Busy destination, you should also configure the Forward Busy Calling Search
Space. If you do not configure the Forward Busy Calling Search Space and the Forward Busy destination
is in a partition, the forward operation fails.
Note When you choose a calling search space for internal calls, the system automatically copies this value to the
calling search space setting for external calls. If you want external calls to be forwarded to a different calling
search space, you must choose a different value in the Calling Search Space field for external calls.
The Call Forward Busy trigger is configured for each line appearance and cannot exceed the maximum number of calls that are
configured for a line appearance. The Call Forward Busy trigger determines how many active calls exist on a line before the
Call Forward Busy setting is activated (for example, ten calls).
Tip Keep the busy trigger slightly lower than the maximum number of calls so that users can make outgoing calls and
perform transfers.
Tip If a call gets forwarded to a directory number that is busy, the call is not completed.
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Field Description
Forward Busy External The fields in this row of fields specify the forwarding treatment for external calls to this directory number if the directory number
is busy. The Destination and Calling Search Space fields is used to redirect the call to the forward destination.
• Voice Mail—Check this check box to use the configured values in the Voice Mail Profile Configuration window for
external calls.
• Destination—This field indicates the Call Forward Busy destination for external calls. Use any dialable phone number,
• Calling Search Space—The Forward Busy External Calling Search Space forwards the call to the Forward Busy External
destination. It applies to all devices that use this directory number.
if you configure the Forward Busy destination, you should also configure the Forward Busy Calling Search
Space. If you do not configure the Forward Busy Calling Search Space and the Forward Busy destination
is in a partition, the forward operation fails.
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Field Description
Forward No Answer Internal The fields in this row of fields specify the forwarding treatment for internal calls to this directory number if the directory number
does not answer. The Destination and Calling Search Space fields are used to redirect the call to the forward destination.
• Voice Mail—Check this check box to use the configured values in the Voice Mail Profile Configuration window.
• Destination—This field indicates the directory number to which an internal call is forwarded when the call is not answered.
Use any dialable phone number, including an outside destination.
• Calling Search Space—The Forward No Answer Internal Calling Search Space is used to forward the call to the Forward
No Answer Internal destination. It applies to all devices that use this directory number.
if you configure the Forward No Answer destination, you should also configure the Forward No Answer
Calling Search Space. If you do not configure the Forward No Answer Calling Search Space and the Forward
No Answer destination is in a partition, the forward operation fails.
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Field Description
Forward No Answer External The fields in this row of fields specify the forwarding treatment for external calls to this directory number if the directory number
does not answer. The Destination and Calling Search Space fields are used to redirect the call to the forward destination.
• Destination—This field indicates the directory number to which an external call is forwarded when the call is not answered.
Use any dialable phone number, including an outside destination.
• Calling Search Space—The Forward No Answer External Calling Search Space is used to forward the call to the Forward
No Answer External destination. It applies to all devices that use this directory number.
if you configure the Forward Busy destination, you should also configure the Forward No Answer Calling
Search Space. If you do not configure the Forward No Answer Calling Search Space and the Forward No
Answer destination is in a partition, the forward operation fails.
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Field Description
Forward No Coverage Internal The Destination and Calling Search Space fields are used to redirect the call to the forward destination.
and Calling Search Space fields. When this check box is checked for internal calls, the system automatically
checks the Voice Mail check box for external calls. If you do not want external calls to forward to the
voicemail system, you must uncheck the Voice Mail check box for external calls.
• Destination—This field specifies the directory number to which an internal nonconnected call is forwarded when an
application that controls that directory number fails. Use any dialable phone number, including an outside destination.
• Calling Search Space—The Forward No Coverage Internal Calling Search Space is used to forward the call to the
Forward No Coverage Internal destination. This value applies to all devices that use this directory number.
if you configure the Forward Busy destination, you should also configure the Forward No Coverage Calling
Search Space. If you do not configure the Forward No Coverage Calling Search Space and the Forward
Busy destination is in a partition, the forward operation fails.
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Field Description
Forward No Coverage External The Destination and Calling Search Space fields are et used to redirect the call to the forward destination.
Specify the following values:
and the Calling Search Space fields. When this check box is checked for internal calls, the system
automatically checks the Voice Mail check box for external calls. If you do not want external calls to forward
to the voicemail system, you must uncheck the Voice Mail check box for external calls.
• Calling Search Space—The Forward No Coverage External Calling Search Space is used to forward the call to the
Forward No Coverage External destination. This value applies to all devices that use this directory number.
configured call forward calling search space is used to forward the call. If the Calling Search Space is
None, the forward operation may fail if the system is using partitions and calling search spaces. For example,
if you configure the Forward No Coverage destination, you should also configure the Forward No Coverage
Calling Search Space. If you do not configure the Forward No Coverage Calling Search Space, and the
Forward No Coverage destination is in a partition, the forward operation may fail.
search space, choose a different value in the Calling Search Space field for external calls.
Forward on CTI Failure This field applies only to CTI route points and CTI ports. The fields in this row specify the forwarding treatment for external
calls to this CTI route point or CTI port if the CTI route point or CTI port fails.
and Calling Search Space fields.
• Calling Search Space—This value applies to all devices that use this directory number.
Forward Unregistered Internal This field applies to unregistered internal DN calls. The calls are rerouted to a specified destination or voicemail.
Note You must also specify the maximum number of forwards in the Service Parameters Configuration window for
a directory number in the Max Forward UnRegistered Hops to DN service parameter.
This parameter specifies the maximum number of forward unregistered hops that are allowed for a directory
number at the same time. This parameter limits the number of times the call can be forwarded due to unregistered
DN when a forwarding loop occurs. Use this count to stop forward loops for external calls that have been Call
Forward Unregistered. Unified Communications Manager terminates the call when the value that is specified in
this service parameter is exceeded.
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Enable Destination Override for Call Forwarding
Field Description
Forward Unregistered External This field applies to unregistered external DN calls. The calls are rerouted to a specified destination or voicemail.
Note You must also specify the maximum number of forwards in the Service Parameters Configuration window for
a directory number in the Max Forward UnRegistered Hops to DN service parameter.
This parameter specifies the maximum number of forward unregistered hops that are allowed for a directory
number at the same time. This parameter limits the number of times the call can be forwarded due to unregistered
DN when a forwarding loop occurs. Use this count to stop forward loops for external calls that have been Call
Forward Unregistered. Unified Communications Manager terminates the call when the value that is specified in
this service parameter is exceeded.
No Answer Ring Duration This field specifies the seconds to wait before forwarding the unanswered call to the Call Forward No Answer destination, if
(seconds) specified. Make sure the value that is specified in this parameter is less than the value that is specified in the T301 Timer service
parameter. If the value in the Forward No Answer Timer service parameter is greater than the value that is specified in the
T301 Timer service parameter, the call is not forwarded and the caller receives a busy signal.
Leave this field empty if you want to set the value in the Cisco Unified Communications Manager Forward No Answer Timer
service parameter.
Enable Destination Override for Call Forwarding

Enable the destination override for call forwarding, Unified Communications Manager ignores the CFA
destination when it matches the calling party number. The override applies to both internal and external calls.
In cases where the calling party number has been transformed, the calling party number does not match the
CFA destination, no override occurs.
Before you begin

Configure Other Call Forwarding Types, on page 334
Procedure
The Service Parameter Configuration window is displayed.
Step 2 In the Clusterwide Parameters (Feature - Hold Reversion) area, set the CFA Destination Override service
parameter value to True.
Call Forwarding Interactions

Feature Interaction
Call Back Calls that are made from the CallBack notification screen will override all
the Call Forward configured values on the target DN. The calls should be
made before the CallBack recall timer expires, otherwise the calls will not
override the Call Forward configured values.
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Feature Interaction
Call Display Restrictions The Connected Number Display restriction applies to all calls that originate
in the system. When this value is set to True, this field interacts
transparently with existing Unified Communications Manager applications,
features, and call processing. The value applies to all calls that terminate
inside or outside the system. The Connected Number Display is updated to
show the modified number or redirected number when a call is routed to a
Call Forward All or Call Forward Busy destination, or gets redirected
through a call transfer or CTI application.
Do Not Disturb On Cisco Unified IP Phones, the message that indicates that the Do Not
Disturb (DND) feature is active takes priority over the message that indicates
that the user has new voice messages. However, the message that indicates
that the Call Forward All feature is active has a higher priority than DND.
External Call Control External Call Control intercepts calls at the translation pattern level, while
Call Forward intercepts calls at the directory number level. External Call
Control has higher priority; for calls where call forward is invoked, Unified
Communications Manager sends a routing query to the adjunct route server
if the translation pattern has an External Call Control profile assigned to it.
Call Forwarding is triggered only when the adjunct route server sends a
Permit decision with a Continue obligation to the Unified Communications
Manager.
Note The Call Diversion Hop Count service parameter that
supports External Call Control, and the Call Forward Call
Hop Count service parameter that supports Call Forwarding
are independent of each other; they work separately.
Extension Mobility Cross Cisco Extension Mobility Cross Cluster supports Call Forwarding.
Cluster
Extend and Connect Extend and Connect supports Call Forward All.
Immediate Divert When the Forward No Answer field in the Directory Number
Configuration window is not configured, Call Forward uses the clusterwide
CFNA timer service parameter, Forward No Answer Timer.
If a user presses the iDivert softkey at the same time as the call is being
forwarded, the call gets diverted to an assigned call forward directory
number (because the amount of time set on the timer was too short), not
the voicemail. To resolve this situation, set the CFNA timer service
parameter to enough time (for example, 60 seconds).
Logical Partitioning Unified Communications Manager performs logical partitioning policy

check using the geolocation identifier information that associates with the
incoming and forwarded devices. This handling applies to all types of call
forwarding.
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Feature Interaction
Multilevel Precedence and Call Forward Busy

Preemption (MLPP)
• You can optionally configure a preconfigured Precedence Alternate
Party target for any MLPP-enabled station.
• Cisco Unified Communications Manager applies the Call Forward
Busy feature to forward a precedence call in the usual manner before
it applies to any Precedence Alternate Party Diversion procedures to
the call.
• The system preserves precedence of calls across multiple forwarded
calls.
• If the incoming precedence call is of higher precedence than the
existing call, preemption occurs. Both the preempted parties in the
active call receive a continuous preemption tone until the station to
which the precedence call is directed hangs up. After hanging up, the
station to which the precedence call is directed receives precedence
ringing. The destination station connects to the preempting call when
the station goes off hook.
Call Forward No Answer

• For calls of Priority precedence level and above, call processing
preserves the precedence level of calls during the forwarding process
and may preempt the forwarded-to user.
• If an Alternate Party is configured for the destination of a precedence
call, call processing diverts the precedence call to the Alternate Party
after the Precedence Call Alternate Party timeout expires. If no
Alternate Party value is configured for the destination of a precedence
call, call processing diverts the precedence call to the Call Forward
No Answer value.
• Normally, precedence calls are routed to users and not to the voicemail
system. The administrator sets the Use Standard VM Handling For
Precedence Calls enterprise parameter to avoid routing precedence
calls to voicemail systems.
If the incoming precedence call is of equal or lower precedence than the

existing call, call processing invokes normal call-forwarding behavior. If
the destination station for a precedence call is nonpreemptable (that is, not
MLPP-configured), call processing invokes call-forwarding behavior.
Alternate Party Diversion (APD) comprises a special type of call forwarding.
If users are configured for APD, APD takes place when a precedence call
is directed to a directory number (DN) that is busy or does not answer.
MLPP APD applies only to precedence calls. An MLPP APD call disables
the DN Call Forward No Answer value for precedence calls.
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Feature Interaction
Originally called party name in When privacy is configured only in the SIP profile of the called party device
Placed Call History and Call Forward All (CFA), or Call Forward Busy (CFB), or Call Forward
Unregistered (CFUR) is enabled, the configured alerting name is displayed
instead of “private”. To ensure that “private” is displayed for call forwarding,
Cisco recommends that you configure the name presentation restriction in
the translation pattern or the route pattern rather than in the SIP profile.
Rollover Lines By using call forwarding settings, you can create rollover lines for a shared
line. This could be a useful for some call center situations.
With rollover lines, when someone dials a number (e.g. 1-800-HOTLINE),
the call always is routed to a specific phone line. This may be a shared line
that is shared by multiple phones. If line 1 is busy, the call rolls over to line
2, if line 2 is busy it rolls over to line 3, and so on. Line 2 or 3 become
available only if line 1 is busy.
This type of call functionality is possible via call forwarding busy settings
and the Busy Trigger as follows:
• On line 1, set the Busy Trigger to 1 and configure Call Forward Busy
to the second line in the chain.
• On line 2, set the Busy Trigger to 1 and configure Call Forward Busy
to the third line in the chain
• Continue this for as many lines as meets your needs.
Secure Tone Call Forward All is supported on protected phones.
Session Handoff When the user hands off a call, a new call gets presented on the desk phone.
While the desk phone is flashing, Call Forward All is not triggered on the
desk phone for the call that was handed off.
Shared Lines When you use a shared line with Call Forward All (CFA) setting and choose
the 'Calling Number' as 'Redirected Party's External Phone Number'
presentation in the outgoing trunk, then the redirected number that is
displayed may not be consistent when shared lines have different E164
numbers configured. So we recommend using the same E164 number across
Shared lines.
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Call Forwarding Restrictions

Feature Restriction
Call Forwarding • If Call Forward All activation occurs in Unified Communications

Manager or the Cisco Unified Communications Self Care Portal,
Unified Communications Manager does not prevent the CFA loop.
• Unified Communications Manager prevents Call Forward All loops
if CFA is activated from the phone, if the number of hops for a Call
Forward All call exceeds the value that is specified for the Forward
Maximum Hop Count service parameter, and if all phones in the
forwarding chain have CFA activated (not CFB, CFNA, or any other
call forwarding options).
For example, if the user with directory number 1000 forwards all calls
to directory number 1001, which has CFB and CFNA configured to
directory number 1002, which has CFA configured to directory
number 1000, Unified Communications Manager allows the call to
occur because directory number 1002 acts as the CFB and CFNA (not
CFA) destination for directory number 1001.
• You cannot activate Call Back if you forward all calls to voicemail
system.
• An uncommon condition in connection with the Forward No Answer
Timeout exists when you press the iDivert softkey. For example, if a
manager presses the iDivert softkey immediately after the Forward
No Answer timeout, Call Forward forwards the call to a preconfigured
directory number. However, if the manager presses the iDivert softkey
before the Forward No Answer timeout, Immediate Divert diverts the
call to the voicemail of the manager.
Immediate Divert When Call Forward All (CFA) and Call Forward Busy (CFB) are activated,
the system does not support Immediate Divert (CFA and CFB have
precedence over Immediate Divert).
Intercom You cannot forward Intercom calls.
Log Out of Hunt Group When a phone that is running SIP (7906, 7911, 7941, 7961) is logged in
to hunt groups and Call Forward All is activated, the call gets presented
to the phone that is running SIP.
When 7940 and 7960 IP phones that are running SIP are logged in to hunt
groups and Call Forward All is activated, the phone gets skipped and the
next phone in the line group is rung.
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Feature Restriction
Logical Partitioning Logical partitioning handling does not take place in the following
circumstances:
• When both the caller and forwarded devices are Voice over IP (VoIP)
phones.
• When geolocation or a geolocation filter is not associated with any
device.
Multilevel Precedence and Multilevel Precedence and Preemption (MLPP) support for supplementary
Preemption (MLPP) services specifies the following restrictions for Call Forwarding:
• Call Forward All (CFA) support for inbound MLPP calls always
forwards the call to the MLPP Alternate Party (MAP) target of the
called party, if the MAP target is configured. In the event of an
incorrect configuration (that is, if no MAP target is specified), the
call gets rejected, and the calling party receives reorder tone.
• Call Forward No Answer (CFNA) support for inbound MLPP calls
forwards the call once to a CFNA target. After the first hop, if the
call remains unanswered, the call is sent to the MAP target of the
original called party, if the MAP target is configured. In the event of
an incorrect configuration (that is, if no MAP target is specified), the
call gets rejected, and the calling party receives reorder tone.
• Call Forward Busy (CFB) support for inbound MLPP calls forwards
the call up to the maximum number that is configured for forwarding
hops. If the maximum hop count is reached, the call is sent to the
MAP target of the original called party, if the MAP target is
configured. In the event of an incorrect configuration (that is, no MAP
target is specified), the call gets rejected, and the calling party receives
reorder tone.
Call Forward Classification with When a call is transferred, the call classification takes on the classification
Call Transfer of the transferred leg, rather than the original leg. For example:
• Incoming call from PSTN is received by a receptionist. This is an
external call.
• The receptionist transfers the call to extension 3100. The transferred
call is now an internal call.
• The user at extension 3100 is busy, but has Call Forward External
configured to send external calls back to the receptionist. However,
because the call takes on the classification of the second leg (internal),
the call goes to voicemail.
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CHAPTER 29
Call Pickup
• Call Pickup Overview, on page 349
• Call Pickup Configuration Task Flow, on page 351
• Call Pickup Interactions, on page 368
• Call Pickup Restrictions, on page 368
Call Pickup Overview

The Call Pickup feature allows users to answer calls that come in on a directory number other than their own.
Group Call Pickup Overview

The Group Call Pickup feature allows users to pick up incoming calls in another group. Users must dial the
appropriate call pickup group number when this feature is activated from a Cisco Unified IP Phone. Use the
softkey, GPickUp, for this type of call pickup. When the user invokes the Group Call Pickup phone feature
while multiple calls are incoming to a pickup group, the user gets connected to the incoming call that has been
ringing the longest. Depending on the phone model, the users can either use the Group Pickup programmable
feature button or the Group Pickup softkey to pick up an incoming call. If Auto Group Call Pickup is not
enabled, the user must press the GPickUp softkey, dial the group number of another pickup group, and answer
the call to make the connection.
Other Group Pickup Overview

The Other Group Pickup feature allows users to pick up incoming calls in a group that is associated with their
own group. The Unified Communications Manager automatically searches for the incoming call in the
associated groups to make the call connection when the user activates this feature from a Cisco Unified IP
Phone. Users use the softkey, OPickUp, for this type of call pickup. If Auto Other Group Pickup is not enabled,
the user must press the softkeys, OPickUp and Answer, to make the call connection. Depending on the phone
model, the users can either use the Call Pickup programmable feature button or the Call Pickup softkey to
pick up an incoming call.
When more than one associated group exists, the first associated group has the highest the priority of answering
calls for the associated group. For example, groups A, B, and C associate with group X, the group A has the
highest priority and the group C has the lowest priority of answering calls. The group X picks up incoming
call in group A, though a call may have come in earlier in group C than the incoming call in group A.
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Directed Call Pickup Overview
Note The longest alerting call (longest ringing time) gets picked up first if multiple incoming calls occur in that
group. For other group call pickup, priority takes precedence over the ringing time if multiple associated
pickup groups are configured.
Directed Call Pickup Overview

The Directed Call Pickup feature allows a user to pick up a ringing call on a DN directly by pressing the
GPickUp or Group Pickup softkeys and entering the directory number of the device that is ringing. If Auto
Directed Call Pickup is not enabled, the user must press the GPickUp softkey, dial the DN of the ringing
phone, and answer the call that will now ring on the user phone to make the connection. Unified
Communications Manager uses the associated group mechanism to control the privilege of a user who wants
to pick up an incoming call by using Directed Call Pickup. The associated group of a user specifies one or
more call pickup groups that are associated to the pickup group to which the user belongs.
If a user wants to pick up a ringing call from a DN directly, the associated groups of the user must contain
the pickup group to which the DN belongs. If two users belong to two different call pickup groups and the
associated groups of the users do not contain the call pickup group of the other user, the users cannot invoke
Directed Call Pickup to pick up calls from each other.
When the user invokes the Directed Call Pickup feature and enters a DN to pick up an incoming call, the user
connects to the call that is incoming to the specified phone whether or not the call is the longest ringing call
in the call pickup group to which the DN belongs. If multiple calls are ringing on a particular DN and the user
invokes Directed Call Pickup to pick up a call from the DN, the user connects to the incoming call that has
been ringing the specified DN the longest.
BLF Call Pickup Overview

The BLF Call Pickup feature allows Unified Communications Manager to notify a phone user when a call is
waiting to be picked up from a BLF DN. The BLF call pickup initiator (the phone that picks up the call) is
selected as the next available line or as a specified line. To use a specified line, the line must remain off hook
before the BLF SD button is pressed. You can configure a hunt list member DN as the BLF DN to allow an
incoming call to a hunt list member to be picked up by the BLF call pickup initiator. The incoming call on
the hunt list member can come from the hunt list or be a directed call. The behavior in each case depends on
how you configure call pickup for the hunt list member DN, the BLF DN, and the hunt pilot number. When
a Call Pickup occurs with the service parameter Auto Call Pickup Enabled set to false, the phone must remain
off hook or the user must press the answer key to pick up the call.
The BLF SD button on the phone can exist in any of the following states:
• Idle—Indicates that no call exists on the BLF DN.
• Busy—Indicates that at least one active call exists on the BLF DN, but no alerts exist.
• Alert— Indicates by flashing that at least one incoming call exists on the BLF DN.
When there is an incoming call to the BLF DN, the BLF SD button flashes on the BLF call pickup initiator
phone to indicate that an incoming call to the BLF DN exists. If Auto Call Pickup is configured, the user
presses the BLF SD button on the call pickup initiator phone to pick up the incoming call. If auto call pickup
is not configured, the phone must remain off hook, or the user must press the answer key to pick up the call.
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Call Pickup Configuration Task Flow

Procedure

Step 1 Configure a Call Pickup Group, on page 353 Configure a call pickup group for each of the
call pickup features that you want to use:
• Call Pickup
• Group Call Pickup
• Other Call Pickup
• Directed Call Pickup
• BLF Call Pickup
You must define groups with unique names

and numbers.
Step 2 Assign a Call Pickup Group to Directory Assign each of the call pickup groups that you
Numbers, on page 353 created to the directory numbers that are
associated with phones on which you want to
enable call pickup. Directory numbers must
be assigned to a call pickup group to use this
feature.
Repeat this procedure for each call pickup
group that you create.
Step 3 Create another call pickup group and associate Perform this step if you are configuring BLF
it with the BLF call pickup group that you Call Pickup.
created in Step 1, on page 351. You can
Note You do not always need to create
associate a call pickup group with multiple
another call pickup group. For
BLF DN call pickup groups.
example, you can have a single
call pickup group that includes
both the initiator DN and the
destination DN. In such cases,
associate the BLF call pickup
group with itself.
Step 4 Configure Partitions for Call Pickup, on page Configure partitions to create a logical
354 grouping of directory numbers (DN) with
similar reachability characteristics. You can
use partitions to restrict access to call pickup
groups. If you assign call pickup group
numbers to a partition, only those phones that
can dial numbers in that partition can use the
call pickup group.
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You must complete this procedure for directed
call pickup. It is optional for other types of call
pickup.
Step 5 Configure Calling Search Space, on page 355 If you configure partitions, you must also
configure calling search spaces. Configure
calling search spaces to identify the partitions
that calling devices can search when they
attempt to complete a call.
You must complete this procedure for directed
all pickup. It is optional for other types of call
pickup.
Step 6 Assign a Call Pickup Group to Hunt Pilots, on (Optional). Assign a call pickup group to a
page 356 hunt pilot DN so that users can pick up calls
that are alerting in the line group members.
Hunt lists that are assigned to a call pickup
group can use Call Pickup, Group Call pickup,
BLF Call Pickup, Other Group Pickup, and
Directed call pickup.
Step 7 Configure notifications: (Optional). Configure notifications when other

members of a pickup group receive a call. You
• Configure Call Pickup Notification, on
can configure audio or visual notifications, or
page 356
both.
• Configure Call Pickup Notification for a
Directory Number, on page 358
• Configure BLF Call Pickup Notification,
on page 358
Step 8 Configure Directed Call Pickup: Before you configure directed call pickup, you
must configure partitions and calling search
• Configure a Time Period, on page 360
spaces. With directed call pickup, the calling
• Configure Time Schedule, on page 360 search space of the user who requests the
• Associate a Time Schedule with a Directed Call Pickup feature must contain the
Partition, on page 360 partition of the DN from which the user wants
to pick up a call.
Time periods and time schedules specify the
times when members in the associated group
are available to accept calls.
Step 9 Configure automatic call answering: (Optional). Enable automatic call answering
and configure timers for automatic call
• Configure Auto Call Pickup, on page 361
answering.
• Configure BLF Auto Pickup, on page 362
Step 10 Configure phone button templates: Configure phone button templates for any of
the call pickup features that you want to use:
• Configure Call Pickup Phone Button
Template, on page 363 • Speed Dial BLF
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Configure a Call Pickup Group

• Associate Call Pickup Button Template • Call Pickup
with Phone, on page 363
• Group Call Pickup
• Configure BLF Speed Dial Number for
the BLF Call Pickup Initiator, on page • Other Group Pickup
364
For Directed Call Pickup, use the Group Call
Pickup button.
Step 11 Configure Softkeys for Call Pickup, on page Configure softkeys for any of the call pickup
364 features that you want to use:
• Configure a Softkey Template for Call • Call Pickup (Pickup)
Pickup, on page 365
• Group Call Pickup (GPickup)
Common Device Configuration, on page • Other Group Pickup (OPickup)
366
• Associate a Softkey Template with a For Directed Call Pickup, use the Group Call
Phone , on page 367 Pickup softkey.
Configure a Call Pickup Group

Procedure
Step 1 From Cisco Unified CM Administration, choose Call Routing > Call Pickup Group.
The Find and List Call Pickup Groups window appears.
The Call Pickup Group Configuration window appears.
Step 3 Configure the fields in the Call Pickup Group Configuration window. For more information on the fields
Assign a Call Pickup Group to Directory Numbers

This section describes how to assign a call pickup group to a directory number. Only directory numbers that
are assigned to a call pickup group can use call pickup, group call pickup, BLF call pickup, other group pickup,
and directed call pickup. If partitions are used with call pickup numbers, make sure that the directory numbers
that are assigned to the call pickup group have a calling search space that includes the appropriate partitions.
Before you begin

Configure a Call Pickup Group, on page 353
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Configure Partitions for Call Pickup
Procedure
Step 1 Choose Device > Phone or Call Routing > Directory Number.
Step 2 Enter the appropriate search criteria to find the phone or directory number that you want to assign to a call
pickup group and click Find.
A list of phones or directory numbers that match the search criteria displays.
Step 3 Choose the phone or directory number to which you want to assign a call pickup group.
Step 4 From the Association Information list in the Phone Configuration window, choose the directory number
to which the call pickup group will be assigned.
Step 5 From the Call Pickup Group drop-down list that displays in the Call Forward and Call Pickup Settings area,
choose the desired call pickup group.
Step 6 To save the changes in the database, click Save.
What to do next
Perform the following tasks:
• Configure Partitions for Call Pickup, on page 354
• Configure Calling Search Space, on page 355
Configure Partitions for Call Pickup

You can restrict access to call pickup groups by assigning a partition to the call pickup group number. When
this configuration is used, only the phones that have a calling search space that includes the partition with the
call pickup group number can participate in that call pickup group. Make sure that the combination of partition
and group number is unique throughout the system. You can create multiple partitions.
If you assign call pickup group numbers to a partition, only those phones that can dial numbers in that partition
can use the call pickup group. If partitions represent tenants in a multitenant configuration, make sure that
you assign the pickup groups to the appropriate partition for each tenant.
Before you begin

Assign a Call Pickup Group to Directory Numbers, on page 353
Procedure
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Configure Calling Search Space
name in this field.
incoming call.
Step 7 Click Save.
Configure Calling Search Space

Before you begin

Configure Partitions for Call Pickup, on page 354
Procedure

Partitions box.
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Assign a Call Pickup Group to Hunt Pilots
Step 8 Click Save.
Assign a Call Pickup Group to Hunt Pilots

Only hunt lists that are assigned to a call pickup group can use Call Pickup, Group Call Pickup, BLF Call
Pickup, Other Group Pickup, and Directed Call Pickup. Follow these steps to assign a call pickup group to
hunt pilots:
Before you begin

Configure Calling Search Space, on page 355
Procedure
Step 2 Enter the appropriate search criteria to find the hunt pilot that you want to assign to a call pickup group and
click Find. A list of hunt pilots that match the search criteria appears.
Step 3 Choose the hunt pilot to which you want to assign a call pickup group.
Step 4 From the Call Pickup Group drop-down list that appears in the Hunt Forward Settings area, choose the
desired call pickup group.
Step 5 Click Save.
Configure Call Pickup Notification

You can configure Call Pickup Notification at the system level, call pickup group level, or individual phone
level.
Before you begin

Assign a Call Pickup Group to Hunt Pilots, on page 356
Procedure

Step 1 Configure Call Pickup Notification for a Call To allow the original called party to pick up the
Pickup Group, on page 357 call prior to the audio and/or visual alert being
sent to the pickup group.
Step 2 Configure Call Pickup Notification for a To configure the type of audio alert to be
Directory Number, on page 358 provided when phone is idle or has an active
call.
Step 3 Configure BLF Call Pickup Notification, on
page 358
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Configure Call Pickup Notification for a Call Pickup Group
Configure Call Pickup Notification for a Call Pickup Group
Before you begin

Assign a Call Pickup Group to Hunt Pilots, on page 356
Procedure
Step 1 From Cisco Unified CM Administration, choose Call Routing > Call Pickup Group.
The Call Pickup Group window appears.
Step 2 Configure the fields in the Call Pickup Group Notification Settings section in the Call Pickup Group
Configuration window. See Call Pickup Notification Fields for Call Pickup, on page 357 for details about
the fields and their configuration options.
Note Refer to Call Pickup Interactions and Restrictions for feature interactions and restrictions that
will affect your Call Pickup configuration.
Call Pickup Notification Fields for Call Pickup
Field Description
Call Pickup Group Notification Policy From the drop-down list, select the notification policy.
The available are options are No Alert, Audio Alert,
Visual Alert, and Audio and Visual Alert.
Call Pickup Group Notification Timer Enter the seconds of delay (integer in the range of 1
to 300) between the time that the call first comes into
the original called party and the time that the
notification to the rest of the call pickup group is sent.
Calling Party Information Check the check box if you want the visual
notification message to the call pickup group to
include identification of the calling party. The system
only makes this setting available when the Call Pickup
Group Notification Policy is set to Visual Alert or
Audio and Visual Alert.
Note The notification is sent only to the
primary line of a device.
Called Party Information Check the check box if you want the visual
notification message to the call pickup group to
include identification of the original called party. The
system makes this setting available when the Call
Pickup Group Notification Policy is set to Visual Alert
or Audio and Visual Alert.
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Configure Call Pickup Notification for a Directory Number
Configure Call Pickup Notification for a Directory Number

Perform these steps to configure the type of audio notification that is provided when a phone is idle or in use.
Before you begin

Configure Call Pickup Notification for a Call Pickup Group, on page 357
Procedure
Step 1 From Cisco Unified CM Administration, choose Call Routing > Directory Number.
The Find and List Directory Numbers window appears.
Step 2 Enter the search criteria and click Find.
Step 3 Click the directory number for which you want to configure the Call Pickup Notification.
Step 4 Choose a device name in the Associated Devices pane and click the Edit Line Appearance button.
The Directory Number Configuration window refreshes to show the line appearance for this DN on the
device that you choose.
Step 5 From the Call Pickup Group Audio Alert Setting(Phone Idle) drop-down list, choose one of the following:
• Use System Default
• Disable
• Ring Once
Step 6 From the Call Pickup Group Audio Alert Setting(Phone Active) drop-down list, choose one of the following:
• Use System Default
• Disable
• Beep Only
Step 7 Click Save.
Configure BLF Call Pickup Notification
Before you begin

Configure Call Pickup Notification for a Directory Number, on page 358
Procedure
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Service Parameter Fields for BLF Call Pickup Notification
Step 4 Configure the fields from Clusterwide Parameters (Device - Phone section in the Service Parameter
Configuration window. See Service Parameter Fields for BLF Call Pickup Notification, on page 359 for more
Service Parameter Fields for BLF Call Pickup Notification
Field Description
Call Pickup Group Audio Alert Setting of Idle Station This parameter determines the kind of audio
notification that is provided when a phone is idle (not
in use) and it needs to be alerted regarding an
incoming call on its Call Pickup Group. Valid values
are as follows:
• Disable
• Ring Once
Call Pickup Group Audio Alert Setting of Busy This parameter determines the kind of audio
Station notification that is provided when a phone is busy (in
use) and it needs to be alerted regarding an incoming
call on its Call Pickup Group. Valid values are as
follows:
• Disable
• Beep Only
BLF Pickup Group Audio Alert Setting of Idle Station This parameter determines the kind of audio
notification that is provided when a phone is idle and
it needs to be alerted regarding an incoming call on
the BLF Pickup Button. Valid values are as follows:
• No Ring
• Ring Once
BLF Pickup Group Audio Alert Setting of Busy This parameter determines the kind of audio
Station notification that is provided when a phone is busy and
it needs to be alerted regarding an incoming call on
the BLF Pickup Button. Valid values are as follows:
• No Ring
• Beep Only
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Configure Directed Call Pickup
Configure Directed Call Pickup

Procedure

Step 1 Configure a Time Period, on page 360 Configure time period for members of the
associated groups to your group.
Step 2 Configure Time Schedule, on page 360 Configure time schedule for members of the
associated groups to your group.
Step 3 Associate a Time Schedule with a Partition, on Associate time schedules with partitions to
page 360 determine where calling devices search when
they are attempting to complete a call during a
particular time of a day.
Configure a Time Period

Use this procedure to define time periods. You can define a start time and an end time, and also specify
repetition interval either as days of the week or a specified date on the yearly calendar.
Procedure
Step 1 From Cisco Unified CM Administration, choose Call Routing > Class of Control > Time Period.
Step 2 Configure the fields in the Time Period Configuration window. For more information on the fields and their
configuration options, see the system Online Help.
Step 3 Click Save.
Configure Time Schedule
Before you begin

Configure a Time Period, on page 360
Procedure
Step 1 From Cisco Unified CM Administration, choose Call Routing > Class of Control > Time Schedule.
Step 2 Configure the fields in the Time Schedule Configuration window. For more information on the fields and
Associate a Time Schedule with a Partition

Associate time schedules with partitions to determine where calling devices search when they are attempting
to complete a call during a particular time of day.
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Configure Automatic Call Answering
Before you begin

Configure Time Schedule, on page 360
Procedure
Step 3 Click Save.
Configure Automatic Call Answering

Procedure

Step 1 Configure Auto Call Pickup, on page 361 You can automate call pickup, group pickup,
other group pickup, directed call pickup, and
BLF call pickup. If you do not enable automatic
call answering, users must press additional
softkeys or dial group numbers to complete the
connection.
Step 2 Configure BLF Auto Pickup, on page 362
Configure Auto Call Pickup

Auto call pickup connects the user to an incoming call. When the user presses the softkey on the phone,
Unified Communications Manager locates the incoming call in the group and completes the call connection.
You can automate call pickup, group pickup, other group pickup, directed call pickup, and BLF call pickup.
If you do not enable automatic call answering, users must press additional softkeys or dial group numbers to
complete the connection.
Before you begin

Associate a Time Schedule with a Partition, on page 360
Procedure
Step 4 In the Clusterwide Parameters (Feature – Call Pickup) section, select True or False from the Auto Call
Pickup Enabled drop-down list to enable or disable automatic call answering for call pickup groups.
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Configure BLF Auto Pickup
Step 5 If the Auto Call Pickup Enabled service parameter is False, enter a value from 12 to 300 in the Call Pickup
No Answer Timer field. This parameter controls the time that a call takes to get restored if the call is picked
up but not answered by using call pickup, group call pickup, or other group call pickup.
Step 6 In the Pickup Locating Timer field, enter a value from 1 to 5. This service parameter specifies the maximum
time, in seconds, for Cisco Unified Communications Manager to identify all alerting calls from all nodes in
the cluster. This information is then used to help ensure that the call that has been waiting longest in the queue
is delivered to the next user who presses the PickUp, GPickUp, or OPickUp softkey.
Step 7 Click Save.
Configure BLF Auto Pickup
Before you begin

Configure Auto Call Pickup, on page 361
Procedure
Step 4 Configure values for the following clusterwide service parameters.
• BLF Pickup Audio Alert Setting of Idle Station—Select True or False from the drop-down list to enable
or disable automatic call answering for call pickup groups. The default value for this service parameter
is False.
• BLF Pickup Audio Alert Setting of Busy Station—If the Auto Call Pickup Enabled service parameter is
False, enter a value from 12 to 300 (inclusive). This parameter controls the time that a call takes to get
restored if the call is picked up but not answered by using call pickup, group call pickup, or other group
call pickup.
Configure Call Pickup Phone Buttons

Procedure

Step 1 Configure Call Pickup Phone Button Template, Add Call Pickup feature to the phone button
on page 363 template.
Step 2 Associate Call Pickup Button Template with

Phone, on page 363
Step 3 Configure BLF Speed Dial Number for the BLF
Call Pickup Initiator, on page 364
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Configure Call Pickup Phone Button Template
Configure Call Pickup Phone Button Template

Follow these steps to add Call Pickup feature to the phone button template.
Before you begin

Configure Automatic Call Answering, on page 361
Procedure
step.
c) Click Save.
Step 6 Click Save.
devices.
Associate Call Pickup Button Template with Phone
Before you begin

Configure Call Pickup Phone Button Template, on page 363
Procedure

button.
Step 5 Click Save.
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Configure BLF Speed Dial Number for the BLF Call Pickup Initiator
Configure BLF Speed Dial Number for the BLF Call Pickup Initiator
Before you begin

Associate Call Pickup Button Template with Phone, on page 363
Procedure

Step 2 Select the phone that you want to use as the BLF call pickup initiator.
Step 3 In the Association pane, Add a new BLF SD link.
The Busy Lamp Field Speed Dial Configuration window appears.
Step 4 Select a Directory Number (BLF DN) that should be monitored by the BLF SD button.
Step 5 Check the Call Pickup check box to use the BLF SD button for BLF Call Pickup and BLF Speed Dial. If you
do not check this check box, the BLF SD button will be used only for BLF Speed Dial.
Step 6 Click Save.
Configure Softkeys for Call Pickup

Procedure

Step 1 Configure a Softkey Template for Call Pickup, Add the Pickup, GPickup, and OPickup softkeys
on page 365 to a softkey template.
• Add a Softkey Template to Common
Configuration to apply configuration options to
phones. This is the most commonly used
• Associate a Common Device method for making a softkey template available
Configuration with a Phone, on page 367 to phones.
Step 3 Associate a Softkey Template with a Phone , Optional. Use this procedure either as an
on page 367 alternative to associating the softkey template
364
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Configure a Softkey Template for Call Pickup

default softkey.
Configure a Softkey Template for Call Pickup

Use this procedure to make the following call pickup softkeys available:
Softkey Description Call States
Call Pickup (Pickup) Allows you to answer a call on On Hook

another extension in your group.
Off Hook
Group Call Pickup (GPickup) Allows you to answer a call on On Hook

extension outside your group.
Off Hook
Other Group Pickup (OPickup) Allows you to answer a call On Hook

ringing in another group that is
Off Hook
associated with your group.
Before you begin

Configure Call Pickup Phone Buttons, on page 362
Procedure
a) Click Add New.
d) Click Save.
template.
click Go.
to display.
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Step 9 Click Save.
devices.
What to do next
Perform one of the following tasks:
• Associate a Softkey Template with a Common Device Configuration, on page 366
• Associate a Softkey Template with a Phone , on page 367

To use the alternative method, see Associate a Softkey Template with a Phone , on page 367.
Procedure
Step 1 Add a Softkey Template to Common Device Configuration, on page 366

Step 2 Associate a Common Device Configuration with a Phone, on page 367
Add a Softkey Template to Common Device Configuration
Procedure
366
Receiving Calls
a) Click Add New.
c) Click Save.
to make available.
Step 5 Click Save.
restart them.
Procedure

Step 4 Click Save.
Procedure

Step 4 Click Save.
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Call Pickup Interactions
Call Pickup Interactions

Feature Interaction
Route Plan Report The route plan report displays the patterns and DNs that are configured in Unified
Communications Manager. Use the route plan report to look for overlapping
patterns and DNs before assigning a DN to call pickup group.
Calling search space and Assigning a partition to the Call Pickup Group number limits call pickup access
partitions to users on the basis of the device calling search space.
Time of Day (TOD) Time of Day (TOD) parameter for members in the associated group enable them
to accept calls within the same time period as their own group. TOD associates
a time stamp to the calling search space and partition.
Call Accounting When a call pickup occurs through auto call pickup, the system generates two
call detail records (CDRs). One CDR applies to the original call that is cleared,
and another CDR applies to the requesting call that is connected.
When a call pickup occurs via non-auto call pickup, the system generates one
call detail record, which applies to the requesting call that is connected.
A CDR search returns all CDRs that match a specific time range and other search
criteria. You can also search for a type of call that is associated with a particular
CDR. The search result displays a call type field that indicates whether the call
is a pickup call.
Call Forwarding When a call pickup occurs with the service parameter Auto Call Pickup Enabled
set to false, the call forward that is configured on the phone gets ignored when
one of the pickup softkeys is pressed. If the call pickup requestor does not answer
the call, the original call gets restored after the pickup no answer timer expires.
Call Pickup Restrictions

Different phone lines to Although you can assign different lines on a phone to different call pickup groups,
different call pickup Cisco does not recommend this setup because it can be confusing to users.
groups
Call Pickup Group • You cannot delete a call pickup group number when it is assigned to a line
Number or DN. To determine which lines are using the call pickup group number,
use Dependency Records in Call Pickup Configuration window. To delete
a call pickup group number, reassign a new call pickup group number to
each line or DN.
• When you update a call pickup group number, Cisco Unified
Communications Manager automatically updates all directory numbers that
are assigned to that call pickup group.
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SIP Phones • The system does not support Call Pickup Notification on a few Cisco Unified
IP Phones that run SIP.
• Call Pickup Notification is only supported on licensed, third-party phones
that run SIP.
Directed Call Pickup • If a device that belongs to a hunt list rings due to a call that was made by
calling the hunt pilot number, users cannot use the Directed Call Pickup
feature to pick up such a call.
• Users cannot pick up calls to a DN that belongs to a line group by using the
Directed Call Pickup feature.
BLF Pickup The system does not support Call Pickup Notification on a few Cisco Unified IP
Phones that run SIP.
Incoming Calling Party If you have configured a prefix in the “Incoming Calling Party International
International Number Number Prefix - Phone ” service parameter, and an international call is placed
Prefix - Phone to a member in the Call Pickup Group, the prefix does not get invoked in the
calling party field if the call gets picked up by another member of the Call Pickup
Group.
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370
CHAPTER 30
Call Park and Directed Call
• Call Park Overview, on page 371
• Call Park Prerequisites, on page 372
• Call Park Configuration Task Flow, on page 372
• Call Park Interactions, on page 386
• Call Park Restrictions, on page 387
• Troubleshooting Call Park, on page 387
• Directed Call Park Overview, on page 388
• Directed Call Park Prerequisites, on page 388
• Directed Call Park Configuration Task Flow, on page 388
• Directed Call Park Interactions, on page 392
• Directed Call Park Restrictions, on page 394
• Troubleshooting Directed Call Park, on page 394
Call Park Overview

The Call Park feature allows you to place a call on hold so that can be retrieved from another phone in the
Unified Communications Manager system (for example, a phone in another office or in a conference room).
If you are on an active call, you can park the call to a call park extension by pressing the Park softkey. Another
phone in your system can then dial the call park extension to retrieve the call.
You can define either a single directory number or a range of directory numbers for use as Call Park extension
numbers. You can park only one call at each Call Park extension number.
The Call Park feature works within a Unified Communications Manager cluster, and each Unified
Communications Manager node in a cluster must have Call Park extension numbers defined. You can define
either a single directory number or a range of directory numbers for use as Call Park extension numbers.
Ensure that the directory number or range of numbers is unique.
Users can dial the assigned route pattern (for example, a route pattern for an intercluster trunk could be 80XX)
and the Call Park number (for example, 8022) to retrieve parked calls from another Unified Communications
Manager cluster. You must ensure that calling search spaces and partitions are properly configured. Call Park
works across clusters.
Valid Call Park extension numbers comprise integers and the wildcard character X. You can configure a
maximum of XX in a Call Park extension number (for example, 80XX), which provides up to 100 Call Park
extension numbers. When a call gets parked,Unified Communications Manager chooses the next Call Park
extension number that is available and displays that number on the phone.
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Call Park Prerequisites
Park Monitoring
Park Monitoring is an optional Call Park feature where Cisco Unified Communications Manager monitors
the status of a parked call until a timer expires. After the timer expires, the call is forwarded to a preassigned
number, sent to voicemail, or returned to the call parker. You can apply park monitoring to phone lines and
to hunt pilots.
Call Park Prerequisites

If you are using call park across clusters, you must have partitions and calling search spaces configured.
Table 28: Cisco Unified IP Phones that Support Park Softkey Template and Call Park Button Template
Phone Model Supported in Softkey Template Supported in Phone Button

Template
Cisco Unified IP Phones 6900 X X

series (except 6901 and 6911)
Cisco IP Phone 7800 Series X X
Cisco Unified IP Phones 7900 X

series (except 7921, 7925, 7936,
7937)
Cisco IP Phone 8800 Series X X

series

series
Cisco Unified IP Phones 7900 X

series (except 7906, 7911, 7921,
7925, 7936, 7937)
Note You can configure Call Park on any line (except line 1) or button by using the programmable line key feature.
Call Park Configuration Task Flow

Before you begin
• Review Call Park Prerequisites, on page 372
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Configure Clusterwide Call Park
Procedure

Step 1 Configure Clusterwide Call Park, on page 373 (Optional). Configure Call Park for the entire
cluster, or use the procedure in step 3 to
configure Call Park on servers within the cluster
Step 2 Configure a Partition for Call Park, on page 374 Create a partition to add a Call Park Number
Step 3 Configure a Call Park Number, on page 375 Configure a Call Park Number to use Call Park
across servers in a cluster.
Step 4 Configure a Softkey Template for Call Park, on Add the Park softkey to a softkey template.
page 377
Common Device Configuration , on page 378, available to phones, you must complete either
Device Configuration , on page 378
to phones. This is the most commonly used
Step 6 Associate a Softkey with a Phone, on page 379 Optional. Use this procedure either as an
alternative to associating the softkey template
Step 7 To Configure Call Park Button, on page 380,

complete the following subtasks:
• Configure a Phone Button Template for
Call Park, on page 380
• Associate a Button Template with a Phone,
on page 380
Step 8 Configure Park Monitoring, on page 381 Complete this optional task flow to add Park
Monitoring to your Call Park configuration.
Configure Clusterwide Call Park

Procedure
Step 1 Choose System > Service Parameters.
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Configure a Partition for Call Park
Step 2 Select the desired node as Server and the service as Cisco CallManager (active).
Step 3 Click the Advanced.
The advanced service parameters are displayed in the window.
Step 4 In Clusterwide Parameter(Feature- General) section set the Enable cluster-wide Call Park Number/Ranges
to True.
The default value is False. This parameter determines whether the Call Park feature is implemented clusterwide
or restricted to a specific Unified CM node.
Step 5 Set the Call Park Display Timer for each server in a cluster that has the Cisco CallManager service and Call
Park configured.
The default is 10 seconds. This parameter determines how long a Call Park number displays on the phone
that parked the call.
Step 6 Set the Call Park Reversion Timer for each server in a cluster that has the Unified Communications Manager
service and Call Park configured.
The default is 60 seconds. This parameter determines the time that a call remains parked. When this timer
expires, the parked call returns to the device that parked the call. If a hunt group member parks a call that
comes through a hunt pilot, the call goes back to the hunt pilot when the Call Park Reversion Timer expires.
Note If you enter a Call Park Reversion Timer value that is less than the Call Park Display Timer, Call
Park numbers may not display on the phone.
Step 7 Click Save.

Step 8 Restart all Unified Communications Manager and CTI Manager services.
Configure a Partition for Call Park

are based on organization, location, and call type. You can configure multiple partitions.
Before you begin

(Optional) Configure Clusterwide Call Park, on page 373
Procedure
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Configure a Call Park Number
name in this field.
incoming call.
Step 8 Click Save.
Configure a Call Park Number

If you want to use Call Park across servers in a cluster, you must configure Call Park extension numbers on
each server.
Ensure that each Call Park directory number, partition, and range is unique within the Unified Communications
Manager. Each Unified Communications Manager to which devices are registered requires its own unique
Call Park directory number and range. Cisco Unified Communications Manager Administration does not
validate the Call Park numbers or range that you use to configure Call Park. To help identify invalid numbers
or ranges and potential range overlaps, use the Unified Communications Manager Dialed Number Analyzer
tool.
Before you begin

Configure a Partition for Call Park, on page 374
Procedure
Step 1 Choose Call Routing > Call Park.

• To add a new Call Park number, click Add New.
• To copy a Call Park number, find the Call Park number or range of numbers and then click the Copy
icon.
• To update a Call Park number, find the Call Park number or range of numbers.
The Call Park number configuration window displays.
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Call Park Configuration Fields
Step 3 Configure the fields in the Call Park configuration fields. See Call Park Configuration Fields, on page 376 for
more information about the fields and their configuration options.
Step 4 To save the new or changed Call Park numbers in the database, click Save.
Call Park Configuration Fields

Field Description
Call Park Number/Range Enter the Call Park extension number. You can enter
digits or the wildcard character X (the system allows
one or two Xs). For example, enter 5555 to define a
single Call Park extension number of 5555 or enter
55XX to define a range of Call Park extension
numbers from 5500 to 5599.
Note You can create a maximum of 100 Call
Park numbers with one call park range
definition. Make sure that the call park
numbers are unique.
Note You cannot overlap call park numbers

between Unified Communications
Manager servers. Ensure that each
Unified Communications Manager
server has its own number range.
Note The call park range is selected from the

list of servers where the call originates.
For example, if phoneA (registered to
nodeA) calls phone B (registered to
nodeB) and the phoneB user presses
Park, phoneB requires a call park range
in the CSS that resides on nodeA. In a
multinode environment where phones
and gateways communicate with various
nodes and where calls that originate
from any server may need to be parked,
the phones require a CSS that contains
call park ranges from all servers.
Description Provide a brief description of this call park number.

The description can include up to 50 characters in any
language, but it cannot include double-quotes (“),
percentage sign (%), ampersand (&), or angle brackets
(<>).
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Configure a Softkey Template for Call Park
Field Description
Partition If you want to use a partition to restrict access to the

call park numbers, choose the desired partition from
the drop-down list. If you do not want to restrict
access to the call park numbers, choose <None> for
the partition.
Note Make sure that the combination of call
park extension number and partition is
unique within the Unified
Unified Communications Manager Using the drop-down list, choose the Cisco Unified
Communications Manager to which these call park
numbers apply.
Configure a Softkey Template for Call Park

Use this procedure to make the Park softkey available.
Park softkey has the following call states:
• On Hook
• Ring Out
• Connected Transfer
Procedure
a) Click Add New.
d) Click Save.
template.
click Go.
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to display.
Step 9 Click Save.
devices.

To use the alternative method, see the section Associate a Softkey Template with a Phone .
Procedure
Step 1 Add a Softkey Template to a Common Device Configuration , on page 378

Procedure
a) Click Add New.
c) Click Save.
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to make available.
Step 5 Click Save.
restart them.
Procedure

Step 4 Click Save.
Associate a Softkey with a Phone

Procedure

Step 4 Click Save.
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Configure Call Park Button
Configure Call Park Button

Configure a Phone Button Template for Call Park
Procedure
step.
c) Click Save.
Step 6 Click Save.
devices.
Before you begin

Configure a Phone Button Template for Call Park, on page 380
Procedure

button.
Step 5 Click Save.
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Configure Park Monitoring
Configure Park Monitoring

Complete these optional tasks to add Park Monitoring to your Call Park configuration.
Before you begin

Park Monitoring is supported on only a subset of phones that support Call Park. The following Cisco Unified
IP Phones support Park Monitoring:
• Cisco IP Phone 8811
• Cisco IP Phone 8851NR
• Cisco IP Phone 8865NR
• Cisco Unified IP Phone 8961
Procedure

Step 1 Configure Park Monitoring System Timers, on Configure system-level timers for the Park
page 381 Monitoring feature.
Step 2 Configure Park Monitoring for Hunt Pilots, on Optional. If you have hunt pilots deployed,
page 382 assign a Park Monitoring destination to a hunt
pilot.
Step 3 Configure Park Monitoring for a Directory Assign a Park Monitoring destination for an
Number, on page 383 individual phone line.
Step 4 Configure Park Monitoring via Universal Line If you have an LDAP directory sync configured,
Template, on page 384 you can use universal line templates to
provision directory number settings for multiple
users with park monitoring configured.
Configure Park Monitoring System Timers

Use this procedure to configure system-level timers for the Park Monitoring feature.
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Configure Park Monitoring for Hunt Pilots
Procedure
Step 2 From the Server drop-down list, select the publisher node.
Step 3 From the Service drop-down list, select Cisco CallManager.
Step 4 Configure values for the following service parameters:
• Park Monitoring Reversion Timer—The number of seconds that Cisco Unified Communications
Manager waits before prompting the user to retrieve a parked call. For individual phone lines, this setting
can be overridden by the same setting in the Directory Number Configuration window. When the call
park reversion timer expires, the call will be forwarded to the hunt pilot.
• Park Monitoring Periodic Reversion Timer—The number of seconds between reversion attempts
when a call has been parked. Cisco Unified Communications Manager prompts the user about the parked
call by ringing, beeping, or flashing the parker's phone. When the park monitoring reversion timer expires,
the call will be forwarded to the parked party and not the hunt pilot.
• Park Monitoring Forward No Retrieve Timer—The number of seconds that park reminder notifications
occur before the parked call is forwarded to the Park Monitoring Forward No Retrieve destination
specified in the call parker's Directory Number configuration. When park monitoring forward no retrieve
timer expires, the call will be forwarded to the hunt pilot.
Note For additional details on these fields, see the service parameter online help.
Step 5 Click Save.
What to do next
Use any of these optional tasks to assign how expired timers get handled for individual phones lines and hunt
pilots:
• Configure Park Monitoring for Hunt Pilots, on page 382
• Configure Park Monitoring for a Directory Number, on page 383
• Configure Park Monitoring via Universal Line Template, on page 384
Configure Park Monitoring for Hunt Pilots

If your deployment uses hunt pilots, use this optional procedure to assign a Park Monitoring destination to a
hunt pilot.
Note For general information on setting up hunt pilots, see the "Configure Hunt Pilots" chapter of the System
Before you begin

Configure Park Monitoring System Timers, on page 381
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Configure Park Monitoring for a Directory Number
Procedure
Step 2 Click Find and select the hunt pilot on which you want to configure a Park Monitoring destination.
Step 3 In the Park Monitoring No Retrieve Destination field, assign a Destination directory number and Calling
Search Space.
Step 4 Complete any remaining fields in the Hunt Pilot Configuration window. For more information on the fields
Step 5 Click Save.
Configure Park Monitoring for a Directory Number

Use this procedure to assign a Park Monitoring destination for an individual phone line. You can forward
calls to another number, send to voicemail, or return to the call parker.
Note The following tools are available to provision settings for multiple phone lines:
• Use a universal line template to provision park monitoring settings for multiple phone lines via an LDAP
directory sync. For details, see Configure Park Monitoring via Universal Line Template, on page 384.
• Use the Bulk Administration Tool to import a CSV file with settings for a large number of phone lines.
For more information, see the Bulk Administration Guide for Cisco Unified Communications Manager.
Before you begin

Procedure
Step 2 Click Find and select the directory number that you want to configure.
Step 3 Enter values for the following Park Monitoring fields:
• Park Monitoring Forward No Retrieve Destination External—When the Park Monitoring Forward
No Retrieve Timer expires, and the parkee is an external party, the call is forwarded either to voicemail
or to a specified directory number. If this field s empty, the call is redirected to the call parker’s line.
• Park Monitoring Forward No Retrieve Destination Internal—When the Park Monitoring Forward
No Retrieve Timer expires, and the parkee is an internal party, the call is forwarded either to voicemail
or to a specified directory number. If this field s empty, the call is redirected to the call parker’s line.
• Park Monitor Reversion Timer—The number of seconds that Cisco Unified Communications Manager
waits before prompting the user to retrieve a call parked on this phone line. If the value is 0 or empty,
then Cisco Unified Communications Manager uses the value of the Park Monitor Reversion Timer
service parameter.
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Configure Park Monitoring via Universal Line Template
Step 4 Complete any remaining fields in the Directory Number Configuration window.For more information on
Step 5 Click Save.
Configure Park Monitoring via Universal Line Template

Use this procedure to assign park monitoring settings to a universal line template. If you have an LDAP
directory sync configured, you can use the universal line template configuration to provision directory number
settings with park monitoring configured for multiple users.
Before you begin

Procedure
Step 1 From Cisco Unified CM Administration, choose User Management > User Phone/Add > Universal Line
Template.
• Click Find and select an existing template.
• Click Add New to create a new template.
Step 3 Expand the Park Monitoring Settings section and complete the fields. For field descriptions, see Park
Monitoring Settings for Universal Line Templates, on page 384.
Step 4 Click Save.
What to do next
To apply the universal line template to individual directory numbers, you must assign the template to a user
profile, feature group template, and LDAP directory sync. When the sync occurs, the template settings get
applied to the phone lines that are a part of the sync. For LDAP setup, see the "Configure End Users" chapters
in the System Configuration Guide for Cisco Unified Communications Manager.
Park Monitoring Settings for Universal Line Templates

The following table contains the Park Monitoring fields in the Universal Line Template Configuration
window of Cisco Unified Communications Manager.
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Park Monitoring Settings for Universal Line Templates
Table 29: Park Monitoring Settings for Universal Line Templates
Field Description
Forward Destination for External When the person whose call is parked is an external party and the Park
Calls When Not Retrieved Monitoring Forward No Retrieve Timer expires, the system sends
the call to one of these destinations:
• Voicemail—Uses the configuration in Voice Mail Profile to
determine where to send the call.
• Revert to Originator—Returns the call to the call parker.
• To forward calls to another number, input the other number in the
text box.
If no option is selected, the call returns to the call parker.
Calling Search Space for If you have configured parked calls to be redirected to a configured
Forwarding External Calls When number, select the calling search space for the forward destination.
Not Retrieved
Forward Destination for Internal When the person whose call is parked is an internal party and the Park
Calls When Not Retrieved Monitoring Forward No Retrieve Timer expires, the system sends
the call to one of these destinations:
• Voicemail—Uses the configuration in Voice Mail Profile to
determine where to send the call.
• Revert to Originator—Returns the call to the call parker.
• To forward calls to another number, input the other number in the
text box.
If no option is selected, the call returns to the call parker.
Calling Search Space for If you have configured parked calls to be redirected to a configured
Forwarding Internal Calls When number, select the calling search space for the forward destination.
Not Retrieved
Park Monitor Reversion Timer This timer determines the number of seconds that Unified
(seconds) Communications Manager waits before prompting the user to retrieve
a call that the user parked. This timer starts when the user presses the
Park softkey on the phone, and a reminder is issued when the timer
expires. The default value is 60 seconds.
Note If you select 0 for the timer then phone lines that use this
template will use the value of the Park Monitor Reversion
Timer cluster-wide service parameter.
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Call Park Interactions
Call Park Interactions

Feature Interaction
CTI Applications CTI applications access call park functionality, including monitoring activity on
call park DNs. To monitor a call park DN, add an application or end user that is
associated with the CTI application to the Standard CTI Allow Call Park
Monitoring user group.
Music On Hold Music On Hold allows users to place calls on hold with music that a streaming
source provides. The Music On Hold audio source for Call Park is selected by
the setting of the Network Hold MOH Audio Source setting within the Phone
Configuration window. If you do not choose an audio source within the device
configuration, Cisco Unified CM uses the audio source that is defined in the
device pool or the system default if the device pool does not specify an audio
source ID.
Route Plan Report The route plan report displays the patterns and directory numbers that are
configured in Unified Communications Manager. Use the route plan report to
look for overlapping patterns and directory numbers before assigning a directory
number to Call Park.
Calling Search Space and Assign the call park directory number or range to a partition to limit call park
Partitions access to users on the basis of the device calling search space.
Immediate Divert Call Park supports Immediate Divert (iDivert or Divert softkey). For example,
user A calls user B, and user B parks the call. User B retrieves the call and then
decides to send the call to a voice-messaging mailbox by pressing the iDivert or
Divert softkey. User A receives the voice mail greeting of user B.
Barge • Barge with Call Park–The target phone (the phone that is being barged upon)
controls the call. The barge initiator “piggybacks” on the target phone. The
target phone includes most of the common features, even when the target
is being barged; therefore, the barge initiator has no feature access. When
the target parks a call, the barge initiator then must release its call (the barge).
• cBarge with Call Park–The target and barge initiator act as peers. The cBarge
feature uses a conference bridge, which causes it to function like a MeetMe
conference. Both phones (target and barge initiator) have full access to their
features.
Directed Call Park We recommend that you do not configure both Directed Call Park and the Park
softkey for Call Park, but the possibility exists to configure both. If you configure
both, ensure that the call park and directed call park numbers do not overlap.
QSIG Intercluster Trunks When a user parks a call across a QSIG intercluster trunk or a QSIG gateway
trunk, the caller who has been parked (the parkee) does not see the To parked
number message. The phone continues to display the original connected number.
The call has been parked, and the user who parked the call can retrieve it. When
the call is retrieved from the parked state, the call continues, but the caller who
was parked does not see the newly connected number.
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Call Park Restrictions
Call Park Restrictions

Feature Restriction
Call Park Unified Communications Manager can park only one call at each call park
extension number.
Shared Line For shared line devices across nodes, the line registers to the node on which the
device registers first. For example, if a device from subscriber2 registers first and
the line is created in subscriber2 and the publisher node, the line belongs to
subscriber2. Each node must be configured with the call park number.
Backup To achieve failover or fallback, configure call park numbers on the publisher
node and subscriber nodes. With this configuration, when the primary node is
down, the line device association gets changed to the secondary node, and the
secondary node call park number gets used.
Directed Call Park If a directed call park (or call park) is initiated from a shared line and the call is
not retrieved from any device, the parked call does not always get reverted to the
recipient in the shared line (parker).
Conference When a conference call is set up between both the shared line and the caller on
park reversion or park reversion fails causing a two-party call (between the other
shared line and caller). The reason is that, on park reversion, Unified
Communications Manager extends the call to both devices sharing the line and
tries to add either party in conference (party already in conference or party that
hit the park). If the party attempts to add the party who is already in the conference
first, then the park reversion fails. When park reversion fails, the shared line can
still barge into the call as usual.
Delete Server If any call park numbers are configured for Unified Communications Manager
on a node that is being deleted in the Server Configuration window (System >
Server), the node deletion fails. Before you can delete the node, you must delete
the call park numbers in Cisco Unified Communications Manager Administration.
Troubleshooting Call Park

User Cannot Park Calls
Problem
User cannot park calls. When the user presses the Park softkey or feature button, the call does not get parked.
Solution
Ensure that a unique call park number is assigned to each Unified Communications Manager in the cluster.
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Call Park Number is Not Displayed Long Enough
The partition that is assigned to the call park number does not match the partition that is assigned to the phone
directory number. For more information on partition, see the System Configuration Guide for Cisco Unified
Call Park Number is Not Displayed Long Enough

Problem
The call park number is not displayed long enough for the user.
Solution
Set the Call Park Display Timer to a longer duration. See Configure Clusterwide Call Park, on page 373 for
more information about the Timer.
Directed Call Park Overview

Directed Call Park allows a user to transfer a call to an available user-selected directed call park number.
Configured Directed Call Park numbers exist cluster-wide. You can configure phones that support the directed
call park Busy Lamp Field (BLF) to monitor the busy or idle status of specific directed call park numbers.
Users can also use the BLF to speed dial a directed call park number.
Unified Communications Manager can park only one call at each directed call park number. To retrieve a
parked call, a user must dial a configured retrieval prefix followed by the directed call park number at which
the call is parked.
Directed Call Park Prerequisites

Make sure that the phones in your deployment support Directed Call Park. For a list of supported phones, run
the Phone Feature List report from Cisco Unified Reporting, selecting Assisted Directed Call Park as the
feature. For details, see Generate a Phone Feature List, on page 1.
Directed Call Park Configuration Task Flow

Before you begin
• Review Directed Call Park Prerequisites, on page 388
Procedure

Step 1 Configure ClusterWide Directed Call Park, on To configure clusterwide parameter for directed
page 389 call park.
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Configure ClusterWide Directed Call Park

Step 2 Configure a Directed Call Park Number, on To add, copy, and update a single Directed Call
page 389 Park extension number or range of extension
numbers.
Step 3 Configure BLF/Directed Call Park Buttons, on Configure a phone button template for
page 391 BLF/Directed Call Park.
Step 4 Synchronize Directed Call Park with Affected Synchronize Directed Call Park with Affected
Devices, on page 392 Devices
Configure ClusterWide Directed Call Park

Procedure
Step 2 To set the timer, update the Call Park Reversion Timer fields in the Clusterwide Parameter(Feature- General)
section.
The default is 60 seconds. This parameter determines the time that a call remains parked. When this timer
expires, the parked call returns to the device that parked the call or to another specified number, depending
on what you configure in the Directed Call Park Configuration window.
Configure a Directed Call Park Number

Before you begin
Ensure that each directed call park directory number, partition, and range is unique within the Unified
Communications Manager. Before you begin, generate a route plan report. If the Park softkey is also activated
(not recommended), ensure that no overlap exists between call park numbers and directed call park numbers.
If reversion number is not configured, the call reverts to the parker (parking party) after the Call Park Reversion
Timer expires.
Configure ClusterWide Directed Call Park, on page 389
Procedure
Step 1 Choose Call Routing > Directed Call Park.

• To add a new directed call park number, click Add New.
• To copy a directed call park number, find the directed call park number or range of numbers and then
click the Copy icon.
• To update a directed call park number, find the directed call park number or range of numbers.
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Directed Call Park Configuration Settings
The directed call park number configuration window is displayed.
Step 3 Configure the fields in the Directed Call Park settings area. See Directed Call Park Configuration Settings,
on page 390 for more information about the fields and their configuration options.
Step 4 To save the new or changed call park numbers in the database, click Save.
If you update a directed call park number, Unified Communications Manager reverts any call that is parked
on that number only after the Call Park Reversion Timer expires.

The Apply Configuration Information dialog is displayed.
Step 6 Click OK.

Step 7 If you are using BLF to monitor directed Call Park numbers, click Restart Devices on the Directed Call
Park Configuration window. This step is optional if you are using change notification.
Directed Call Park Configuration Settings

Field Description
Number Enter the directed call park number. You can enter digits (0-9) or the wildcard
character ([], - , *, ^, #) and X (the system allows one or two Xs). For example,
enter 5555 to define a single call park number of 5555 or enter 55XX to define
a range of directed call park extension numbers from 5500 to 5599. Make sure
that the directed call park numbers are unique and that they do not overlap with
call park numbers.
Description Provide a brief description of this directed call park number or range. The
description can include up to 50 characters in any language, but it cannot include
double quotation marks (“), percentage sign (%), ampersand (&), or angle brackets
(<>) and tabs.
Partition If you want to use a partition to restrict access to the directed call park numbers,
choose the desired partition from the drop-down list. If you do not want to restrict
access to the directed call park numbers, leave the partition as the default of
<None>.
Note Make sure that the combination of directed call park number and
partition is unique within Unified Communications Manager.
Reversion Number Enter the number to which you want the parked call to return if not retrieved, or
leave the field blank.
Note A reversion number can comprise digits only; you cannot use
wildcards.
Reversion Calling Search Using the drop-down list, choose the calling search space or leave the calling
Space search space as the default of <None>.
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Configure BLF/Directed Call Park Buttons
Field Description
Retrieval Prefix For this required field, enter the prefix for retrieving a parked call. The system
needs the retrieval prefix to distinguish between an attempt to retrieve a parked
call and an attempt to initiate a directed park.
Configure BLF/Directed Call Park Buttons

Before you begin
Configure ClusterWide Directed Call Park, on page 389
Procedure
Step 2 After the configuration window displays, click the Add a new BLF Directed Call Park link in the Association
Information pane.
Note The link does not display in the Association Information pane if the phone button template that
you applied to the phone or device profile does not support BLF/Directed Call Park.
Step 3 Configure the fields in the BLF/Directed Call Park fields area. See BLF/Directed Call Park Configuration
Fields, on page 391 for more information about the fields and their configuration options.
Step 4 After you complete the configuration, click Save and close the window.
The directory numbers are displayed in the Association Information pane of the Phone Configuration Window.
BLF/Directed Call Park Configuration Fields

Table 30: BLF/Directed Call Park Button Configuration Fields
Field Description
Directory Number The Directory Number drop-down list displays a list of Directed Call Park number
that exist in the Unified Communications Manager database.
For phones that are running SCCP or phones that are running SIP, choose the
number (and corresponding partition, if it is displayed) that you want the system
to dial when the user presses the speed-dial button; for example, 6002 in 3.
Directory numbers that display without specific partitions belong to the default
partition.
Label Enter the text that you want to display for the BLF/Directed Call Park button.
This field supports internationalization. If your phone does not support
internationalization, the system uses the text that displays in the Label ASCII
field.
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Synchronize Directed Call Park with Affected Devices
Field Description
Label ASCII Enter the text that you want to display for the BLF/Directed Call Park button.
The ASCII label represents the noninternationalized version of the text that you
enter in the Label field. If the phone does not support internationalization, the
system uses the text that displays in this field.
Note If you enter text in the Label ASCII field that differs from the text
in the Label field, Cisco Unified Communications Manager
Administration accepts the configuration for both fields, even though
the text differs.
Synchronize Directed Call Park with Affected Devices

Procedure
Step 1 Choose Call Routing > Directed Call Park.

The Find and List Directed Call Parks window is displayed.
Step 2 Choose the search criteria to use.

Step 3 Click Find.
The window displays a list of directed call parks that match the search criteria.
Step 4 Click the directed call park to which you want to synchronize applicable devices. The Directed Call Park
Configuration window is displayed.
Step 5 Make any additional configuration changes.
Step 6 Click Save.
The Apply Configuration Information dialog is displayed.
Step 8 Click OK.
Directed Call Park Interactions

The following table describes feature interactions with the Directed Call Park feature.
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Directed Call Park Interactions
Feature Interaction
Music On Hold The Music On Hold Audio Source for directed call park is assigned via the Default
Network Hold MOH Audio Source service parameter. To assign the parameter:
1. From Cisco Unified CM Administration, choose System > Service
Parameters.
2. From the Server drop-down list, choose a Unified Communications Manger
cluster node.
3. From the Service drop-down list, select Cisco CallManager.
4. Under Clusterwide Paramters (Service), assign a MOH audio source to the
Default Network Hold MOH Audio Source ID parameter. The default is
1.
5. Click Save.
Note For detailed information on adding MOH audio sources to the

system, refer to the "Configure Music On Hold" section of this
guide.
Calling Search Space and Assign the Directed Call Park Directory number or range to a partition to limit
Partitions Directed Call Park access to users on the basis of the device calling search space.
Immediate Divert Directed call park supports Immediate Divert (iDivert or Divert softkey). For
example, user A calls user B, and user B parks the call. User B retrieves the call
and then decides to send the call to a voice-messaging mailbox by pressing the
iDivert or Divert softkey. User A receives the voicemail greeting of user B.
Barge • Barge with Directed Call Park–The target phone (the phone that is being
barged upon) controls the call. The barge initiator “piggybacks” on the target
phone. The target phone includes most of the common features, even when
the target is being barged; therefore, the barge initiator has no feature access.
When the target parks a call by using directed call park, the barge initiator
then must release its call (the barge).
• cBarge with Directed Call Park–The target and barge initiator act as peers.
The cBarge feature uses a conference bridge that makes it behave like to a
meet-me conference. Both phones (target and barge initiator) retain full
access to their features.
Call Park We recommend that you do not configure both directed call park and the Park
softkey for call park, but the possibility exists to configure both. If you configure
both, ensure that the call park and directed call park numbers do not overlap.
A caller who has been parked (the parkee) by using the directed call park feature
cannot, while parked, use the standard call park feature.
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Directed Call Park Restrictions
Directed Call Park Restrictions

Feature Restriction
Directed Call Park Unified Communications Manager one party can park only one call at each
number Directed Call Park number.
You cannot delete a Directed Call Park number that a device is configured to
monitor (using the BLF button). A message indicates that the Directed Call Park
number or range cannot be deleted because it is in use. To determine which
devices are using the number, click the Dependency Records link on the Directed
Call Park Configuration window.
Standard Call Park A caller who has been parked (the parkee) by using the Directed Call Park feature
Feature cannot, while parked, use the standard call park feature.
Directed Call Park Feature We recommend that you do not press both the Transfer and Directed Call Park
buttons simultaneously, as this may result in both DPark and Transfer failures.
Directed Call Park BLF The Directed Call Park BLF cannot monitor a range of Directed Call Park
numbers. A user can monitor only individual Directed Call Park numbers by
using the Directed Call Park BLF. For example, if you configure a Directed Call
Park number range 8X, you cannot use the Directed Call Park BLF to monitor
that whole range of 80 to 89.
Directed Call Park for The following limitations apply to Directed Call Park for phones that are running
phones that are running SIP:
SIP
• Directed Call Park gets invoked by using the Transfer softkey on Cisco
Unified IP Phones 7940 and 7960 that are running SIP.
• The system does not support directed call park when the Blind Transfer
softkey is used on Cisco Unified IP Phones 7940 and 7960 that are running
SIP.
• The system does not support directed call park BLF on Cisco Unified IP
Phones 7940 and 7960 that are running SIP, and third-party phones that are
running SIP.
Troubleshooting Directed Call Park

User Cannot Retrieve Parked Calls
User cannot retrieve parked calls. After dialing the directed call park number to retrieve a parked call, the
user receives a busy tone, and the IP phone displays the message, “Park Slot Unavailable”.
Ensure that the user dials the retrieval prefix followed by the directed call park number.
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User cannot park calls. After the Transfer softkey (or Transfer button if available) is pressed and the directed
call park number is dialed, the call does not get parked.
Ensure that the partition that is assigned to the call park number matches the partition that is assigned to the
phone directory number. Ensure that the partition and calling search space are configured correctly for the
device. For more information about the partition, see the System Configuration Guide for Cisco Unified
User Receives a Reorder Tone After the Reversion Timer Expires

User cannot park calls. The user receives a reorder tone after the reversion timer expires.
Ensure that the user presses the Transfer softkey (or Transfer button if available) before dialing the directed
call park number, and then presses the Transfer softkey (or Transfer button) again or goes on hook after dialing
the directed call park number. Because directed call park is a transfer function, the directed call park number
cannot be dialed alone.
Note You can complete the transfer only by going on hook rather than pressing the Transfer softkey (or Transfer
button) a second time if the Transfer On-hook Enabled service parameter is set to True.
User Receives a Reorder Tone or Announcement

User cannot park calls. After pressing the Transfer softkey (or Transfer button if available) and dialing the
directed call park number, the user receives a reorder tone or announcement.
Ensure that the dialed number is configured as a directed call park number.
User Cannot Park a Call at a Number Within The Range

After configuring a range of directed call park numbers, the user cannot park a call at a number within the
range.
Review the syntax for entering a range of directed call park numbers. If incorrect syntax is used, the system
may appear to configure the range when it actually does not.
Parked Calls Revert Too Quickly

Parked calls revert too quickly.
Set the Call Park Reversion Timer to a longer duration.
Park Slot Unavailable

User cannot park calls. After pressing the Transfer softkey (or Transfer button if available) and dialing the
directed call park number, the user receives a busy tone, and the IP phone displays the message, “Park Slot
Unavailable".
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Parked Calls Do Not Revert to the Parked Call Number
Ensure that the dialed directed call park number is not already occupied by a parked call or park the call on
a different directed call park number.
Parked Calls Do Not Revert to the Parked Call Number

Parked calls do not revert to the number that parked the call.
Check the configuration of the directed call park number to ensure that it is configured to revert to the number
that parked the call rather than to a different directory number.
Number or Range Cannot Be Deleted Because It Is in Use

When an attempt is made to delete a directed call park number or range, a message displays that indicates that
the number or range cannot be deleted because it is in use.
You cannot delete a directed call park number that a device is configured to monitor (by using the BLF button).
To determine which devices are using the number, click the Dependency Records link in the Directed Call
Park Configuration window.
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CHAPTER 31
Extension Mobility
• Extension Mobility Overview, on page 397
• Extension Mobility Prerequisites, on page 397
• Extension Mobility Configuration Task Flow, on page 398
• Cisco Extension Mobility Interactions, on page 406
• Cisco Extension Mobility Restrictions, on page 407
• Extension Mobility Troubleshooting, on page 408
Extension Mobility Overview

Cisco Extension Mobility allows users to temporarily access their phone settings, such as line appearances,
services, and speed dials, from other phones within your system. If you have a single phone that will be used
by multiple workers, for example, you can configure extension mobility so that individual users can log in to
the phone and access their settings without affecting settings on other user accounts.
After a user logs in using extension mobility and if the extension mobility profile is already associated to the
application user, then CTI application sends device-related information.
CTI application can control a device the user is logged into (using that extension mobility profile) without
having to have direct control of the device. Therefore, the recording with the device profile association to the
application user should work though they have not associated the device directly.
Extension Mobility Prerequisites

• A TFTP server that is reachable.
• Extension mobility functionality extends to most Cisco Unified IP Phones. Check the phone documentation
to verify that Cisco Extension Mobility is supported.
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Extension Mobility Configuration Task Flow
Extension Mobility Configuration Task Flow

Before you begin
Procedure

support the extension mobility feature.
Step 2 Activate Extension Mobility Services, on page

398
Step 3 Configure the Cisco Extension Mobility Phone Configure the extension mobility IP phone
Service, on page 399 service to which users can later subscribe to
access extension mobility.
Step 4 Create an Extension Mobility Device Profile Configure an extension mobility device profile.
for Users, on page 400 This profile acts as a virtual device that maps
onto a physical device when a user logs in to
extension mobility. The physical device takes
on the characteristics in this profile.
Step 5 Associate a Device Profile to a User, on page Associate a device profile to users so that they
400 can access their settings from a different phone.
You associate a user device profile to a user in
the same way that you associate a physical
device.
Step 6 Subscribe to Extension Mobility, on page 401 Subscribe IP phones and device profiles to the
extension mobility service so that users can log
in, use, and log out of extension mobility.
Step 7 Configure the Change Credential IP Phone To allow users to change their PINs on their
Service, on page 401 phones, you must configure the change
credential Cisco Unified IP Phone service and
associate the user, the device profile, or the IP
phone with the change credential phone service.
Step 8 (Optional) Configure Service Parameters for If you want to modify the behavior of extension
Extension Mobility, on page 402 mobility, configure the service parameters.
Activate Extension Mobility Services

Procedure
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Configure the Cisco Extension Mobility Phone Service
Step 2 From the Server drop-down list, choose the requried node.
Step 3 Activate the following services:
a) Cisco CallManager
b) Cisco Tftp
c) Cisco Extension Mobility
d) ILS Service
Note You must choose publisher node to activate the ILS services.
Step 4 Click Save.

Step 5 Click OK.

Configure the extension mobility IP phone service to which users can later subscribe to access extension
mobility.
Procedure
Step 3 In the Service Name field, enter a name for the service.
Step 4 In the Service URL field, enter the Service URL.
The format is http://<IP Address>:8080/emapp/EMAppServlet?device=#DEVICENAME#.
IP Address is the IP address of the Unified Communications Manager where Cisco Extension Mobility
is activated and running.
It can either be a IPv4 or a IPv6 address.
Example:
http://123.45.67.89:8080/emapp/EMAppServlet?device=#DEVICENAME#
Example:
http://[2001:0001:0001:0067:0000:0000:0000:0134]:8080/emapp/EMAppServlet?device=#DEVICENAME#
This format allows a user to sign-in using User ID and PIN. You can configure more sign-in options for IP
phone users who have subscribed to the extension mobility service. To configure more sign-in options, append
the loginType parameter to the Service URL, in the following formats:
• loginType=DN enables users to sign in using Primary Extension and PIN.
The Service URL format is:http://<IP
Address>:8080/emapp/EMAppServlet?device=#DEVICENAME#&loginType=DN.
• loginType=SP enables users to sign in using Self Service User ID and PIN.
Address>:8080/emapp/EMAppServlet?device=#DEVICENAME#&loginType=SP.
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Create an Extension Mobility Device Profile for Users
• loginType=UID enables users to sign in using User ID and PIN.

Address>:8080/emapp/EMAppServlet?device=#DEVICENAME#&loginType=UID.
If you do not append loginType to the end of the URL, the default sign-in option displayed is User ID and
PIN.
Step 5 In the Service Type field, choose whether the service is provisioned to the Services, Directories, or Messages
button.
Step 6 Click Save.

Configure an extension mobility device profile. This profile acts as a virtual device that maps onto a physical
device when a user logs in to extension mobility. The physical device takes on the characteristics in this
profile.
Procedure
Step 1 From Cisco Unified CM Administration, choose Device > Device Settings > Device Profile.
• Click Find to modify the settings and choose an existing device profile from the resulting list.
• Click Add New to add a new device profile and choose an option from the Device Profile Type. Click
Next.
• Choose a device protocol from the Device Protocol drop-down list and click Next.
Step 3 Configure the fields. For more information on the fields and their configuration options, see Online Help.
Step 4 Click Save.
Step 5 From the Association Information section, click Add a new DN.
Step 6 In the Directory Number field, enter the directory number and click Save.
Step 7 Click Reset and follow the prompts.
Associate a Device Profile to a User

Associate a device profile to users so that they can access their settings from a different phone. You associate
a user device profile to a user in the same way that you associate a physical device.
Tip You can use the Bulk Administration Tool (BAT) to add and delete several user device profiles for Cisco
Extension Mobility at one time. See the Bulk Administration Guide for Cisco Unified Communications
Manager.
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Subscribe to Extension Mobility
Procedure
• Click Find to modify the settings for an existing user, enter search criteria, and choosing an existing user
from the resulting list.
• Click Add New to add a new user.
Step 3 Under Extension Mobility, locate the device profile that you created and move it from Available Profiles
to Controlled Profiles.
Step 4 Check the Home Cluster check box.
Step 5 Click Save.

Subscribe IP phones and device profiles to the extension mobility service so that users can log in, use, and
log out of extension mobility.
Procedure
Step 1 Perform one of the following tasks from Cisco Unified CM Administration:
• Choose Device > Phone, specify search criteria, click Find, and choose a phone which users will use
for extension mobility.
• Choose Device > Device Settings > Device Profile, specify search criteria, click Find, and choose the
device profile that you created.
Step 2 From the Related Links drop-down list, choose Subscribe/Unsubscribe Services, and then click Go.
Step 3 From the Select a Service drop-down list, choose the Extension Mobility service.
Step 4 Click Next.
Step 5 Click Subscribe.
Step 6 Click Save and close the popup window.
Configure the Change Credential IP Phone Service

To allow users to change their PINs on their phones, you must configure the change credential Cisco Unified
IP Phone service and associate the user, the device profile, or the IP phone with the change credential phone
service.
Procedure
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Configure Service Parameters for Extension Mobility

Step 3 In the Service Name field, enter Change Credential.
Step 4 In the Service URL field, enter the following value, where server designates the server where the Change
Credential IP phone service runs:
http://server:8080/changecredential/ChangeCredentialServlet?device=#DEVICENAME#
Step 5 (Optional) In the Secure-Service URL field, enter the following value, where server is the server where
the Change Credential IP phone service runs:
https://server:8443/changecredential/ChangeCredentialServlet?device=#DEVICENAME#
Step 6 Configure the remaining fields in the IP Phone Services Configuration window, and choose Save.
Step 7 To subscribe the Cisco Unified IP Phone to the Change Credential IP phone service, choose Device > Phone.
Step 8 In the Phone Configuration window, go to the Related Links drop-down list and choose
Subscribe/Unsubscribe Services.
Step 9 Click Go.
Step 10 From the Select a Service drop-down list, choose the Change Credential IP phone service.
Step 11 Click Next.
Step 13 Click Save.
Configure Service Parameters for Extension Mobility

(Optional)
If you want to modify the behavior of extension mobility, configure the service parameters.
Procedure
Step 2 From the Server field, choose the node that is running the Cisco Extension Mobility service.
Step 3 From the Service field, choose Cisco Extension Mobility.
Step 4 Click Advanced to show all service parameters.
See Extension Mobility Service Parameters, on page 403 for more information about these service parameters
and their configuration options.
Step 5 Click Save.
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Extension Mobility Service Parameters

Table 31: Extension Mobility Service Parameters
Service Parameter Description
Enforce Intra-cluster Select True to specify a maximum time for local logins. After this time, the
Maximum Login Time system automatically logs out the device. False, which is the default setting,
means that no maximum time for logins exists.
To set an automatic logout, you must choose True for this service parameter and
also specify a system maximum login time for the Intra-cluster Maximum
Login Time service parameter. Cisco Unified Communications Manager then
uses the automatic logout service for all logins.
If the value of Enforce Intra-cluster Maximum Login Time is set to False and
you specify a valid maximum login time for the Intra-cluster Maximum Login
Time service parameter, then the value of Enforce Intra-cluster Maximum
Login Time automatically changes to True.
Intra-cluster Maximum This parameter sets the maximum time that a user can be locally logged in to a
Login Time device, such as 8:00 (8 hours) or:30 (30 minutes).
The system ignores this parameter and set the maximum login time to 0:00, if
the Enforce Intra-cluster Maximum Login Time parameter is set to False.
Valid values are between 0:00 and 168:00 in the format HHH:MM, where HHH
represents the number of hours and MM represents the number of minutes.
Note If you grant a user access to set their Extension Mobility maximum
login time (configured via the Allow End User to set their
Extension Mobility maximum login time check box in the User
Profile Configuration) the user's configuration in the Self-Care
Portal overrides the value of the Intra-cluster Maximum Login
Time service parameter.
Maximum Concurrent Specify the maximum number of login or logout operations that can occur
Requests simultaneously. This number prevents the Cisco Extension Mobility service from
consuming excessive system resources. The default value of 5 is acceptable in
most cases.
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Multiple Login Behavior When users are logged in to one phone and then login to a second phone either
in the same cluster or on a different cluster, users can view the login behavior on
the second phone based on the Multiple Login Behavior setting defined on the
Service Parameter Configuration page.
Choose one of the following options from the drop–down list:
• Multiple Logins Allowed—You can login to more than one device at a time.
• Multiple Logins Not Allowed—You can be logged in to only one device.
The login attempts to the second device fails and the phone displays the
error code “25” (Multi-Login Not Allowed). You can login successfully,
only when you have logged out from the first device. This is the default
value.
• Auto Logout—When you try to login to a second device (either Extension
Mobility or Extension Mobility Cross Cluster), the Cisco Unified
Communications Manager automatically logs you out of the first device.
This is a required field.

Note Multiple login behavior is also applicable between two Extension
Mobility Cross Cluster logins.
Alphanumeric User ID Choose True to allow the user ID to contain alphanumeric characters. Choosing
False allows the user ID to contain only numeric characters.
Note The Alphanumeric User ID parameter applies systemwide. You can
have a mix of alphanumeric and numeric user IDs. The system
supports only user IDs that can be entered by using the alphanumeric
keypad. The case-sensitive userid field requires the characters to be
lowercase.
Remember the Last User When you choose False, the system does not remember the last user who logged
Logged In in to the phone. Use this option when the user access the phone on a temporary
basis only. Choose True to remember the last user that logged into the phone.
Use this option when a phone has only one user.
For example, Cisco Extension Mobility is used to enable the types of calls that
are allowed from a phone. Individuals who are not logged in and who are using
their office phone can make only internal or emergency calls. But after logging
in using Cisco Extension Mobility, the user can make local, long-distance, and
international calls. In this scenario, only this user regularly logs in to the phone.
It makes sense to set the Cisco Extension Mobility to remember the last user ID
that logged in.
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Clear Call Logs on Choose True to specify that the call logs are cleared during the Cisco Extension
Intra-cluster EM Mobility manual login and logout process.
While a user is using the Cisco Extension Mobility service on an IP phone, all
calls (placed, received, or missed) appear in a call log and can be retrieved and
seen on the IP phone display. To ensure privacy, set the Clear Call Log service
parameter to True. This ensures that the call logs are cleared when a user logs
out and another user logs in.
For extension mobility cross cluster (EMCC), the call log is always cleared when
the user logs in or out of a phone.
Note Call logs are cleared only during manual login/logout. If a Cisco
Extension Mobility logout occurs automatically or any occurrence
other than a manual logout, the call logs are not cleared.
Validate IP Address This parameter sets whether validation occurs on the IP address of the source
that is requesting login or logout.
If the parameter is set to True, the IP address from which a Cisco Extension
Mobility log in or log out request occurs and is validated to ensure that it is trusted.
Validation is first performed against the cache for the device that will log in or
log out.
If the IP address is found in the cache or in the list of trusted IP addresses or is
a registered device, the device can log in or log out. If the IP address is not found,
the log in or log out attempt is blocked.
If the parameter is set to False, the Cisco Extension Mobility log in or log out
request is not validated.
Validation of IP addresses can affect the time that is required to log in or log out
a device, but it offers additional security that prevents unauthorized log in or log
out attempts. This function is recommended, especially when used with logins
from separate trusted proxy servers for remote devices.
Trusted List of IPs This parameter appears as a text box (the maximum length is 1024 characters).
You can enter strings of trusted IP addresses or hostnames which are separated
by semicolons, in the text box. IP address ranges and regular expressions are not
supported.
Allow Proxy If the parameter is True, the Cisco Extension Mobility log in and log out
operations that use a web proxy are allowed.
If the parameter is False, the Cisco Extension Mobility log in and log out requests
coming from behind a proxy get rejected.
The setting that you select takes effect only if the Validate IP Address parameter
specifies true.
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Cisco Extension Mobility Interactions
Extension Mobility Cache In this field, enter the size of the device cache that is maintained by Cisco
Size Extension Mobility. The minimum value for this field is 1000 and the maximum
is 20000. The default value is 10000.
The value that you enter takes effect only if the Validate IP Address parameter
is True.
Cisco Extension Mobility Interactions

Table 32: Cisco Extension Mobility Interactions
Feature Interaction
Assistant A manager who uses Cisco Extension Mobility can simultaneously use Cisco Unified
Communications Manager Assistant. The manager logs in to the Cisco Unified IP
Phone by using Cisco Extension Mobility and then chooses the Cisco IP Manager
Assistant service. When the Cisco IP Manager Assistant service starts, the manager
can access assistants and all Cisco Unified Communications Manager Assistant
features (such as call filtering and Do Not Disturb).
BLF Presence When you configure BLF/speed dial buttons in a user device profile, a phone that
supports Cisco Extension Mobility displays BLF presence status on the BLF/SpeedDial
buttons after you log in to the device.
When the extension mobility user logs out, a phone that supports Cisco Extension
Mobility displays BLF presence status on the BLF/SpeedDial buttons for the logout
profile that is configured.
Call Display When you enable call display restrictions, Cisco Extension Mobility functions as
Restrictions usual: when a user is logged in to the device, the presentation or restriction of the call
information depends on the user device profile that is associated with that user. When
the user logs out, the presentation or restriction of the call information depends on
the configuration that is defined for that phone type in the Phone Configuration
window.
To use call display restrictions with Cisco Extension Mobility, check the Ignore
Presentation Indicators (internal calls only) check obx in both the Device Profile
Configuration window and the Phone Configuration window.
Call Forward All An enhancement to call forward all calling search space (CSS) lets you upgrade to
Calling Search Space later releases of Cisco Unified Communications Manager without loss of functionality.
The CFA CSS Activation Policy service parameter supports this enhancement. In
the Service Parameter Configuration window, this parameter displays in the
Clusterwide Parameters (Feature - Forward) section with two options:
• With Configured CSS (default)
• With Activating Device/Line CSS
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Cisco Extension Mobility Restrictions
Feature Interaction
Do Not Disturb For extension mobility, the device profile settings include do not disturb (DND)
incoming call alert and DND status. When a user logs in and enables DND, the DND
incoming call alert and DND status settings are saved, and these settings are used
when the user logs in again.
Note When a user who is logged in to extension mobility modifies the DND
incoming call alert or DND status settings, this action does not affect
the actual device settings.
Intercom Cisco Extension Mobility supports the intercom feature. To support intercom, Cisco
Extension Mobility uses a default device that is configured for an intercom line. An
intercom line is presented on only the default device.
You can assign an intercom line to a device profile. When a user logs in to a device
that is not the default device, the intercome line is not presented.
The following additional considerations apply to intercom for Cisco Extension
Mobility:
• When Unified Communications Manager assigns an intercom line to a device
and the default device value is empty, the current device is selected as the default
device.
• When AXL programatically assigns an intercom DN, you must update the
intercom DN separately by using Cisco Unified Communications Manager
Administration to set the default device.
• When you delete a device that is set as the intercom default device for an intercom
line, the intercom default device is no longer set to the deleted device.
Internet Protocol Cisco Extension Mobility Supports IPv6. You can use phones with an IP addressing
Version 6 (IPv6) mode of IPv6 or dual-stack (IPv4 and IPv6).
Prime Line If you select On for the Always Use Prime Line parameter in the Device Profile or
Default Device Profile Configuration window, a Cisco Extension Mobility user can
use this feature after logging in to the device that supports Cisco Extension Mobility.
Cisco Extension Mobility Restrictions

Table 33: Cisco Extension Mobility Restrictions
Feature Restriction
Cache Cisco Extension Mobility maintains a cache of all logged-in user information for 2
minutes. If a request comes to extension mobility regarding a user who is represented
in the cache, the user is validated with information from the cache. For example, if
a user changes the password, logs out, and then logs back in within 2 minutes, both
the old and new passwords are recognized.
Call Back When a Cisco Extension Mobility user logs out of a device, all call back services that
are active for the Cisco Extension Mobility user are automatically cancelled.
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Extension Mobility Troubleshooting
Feature Restriction
Character Display The characters that display when a user logs in depend on the current locale of the
phone. For example, if the phone is currently in the English locale (based on the
Logout profile of the phone), the user can only enter English characters in the UserID.
Hold Reversion Cisco Extension Mobility does not support the hold reversion feature.
IP Phones Cisco Extension Mobility requires a physical Cisco Unified IP Phone for login. Users
of office phones that are configured with Cisco Extension Mobility cannot remotely
log in to their phones.
Locale If the user locale that is associated with the user or profile is not the same as the locale
or device, after a successful login, the phone will restart and then reset. This behavior
occurs because the phone configuration file is rebuilt. Addon-module mismatches
between profile and device can cause the same behavior.
Log Out If Cisco Extension Mobility is stopped or restarted, the system does not automatically
log out users who are already logged in after the logout interval expires. Those phones
automatically log out users only once a day. You can manually log out these users
from either the phones or from Cisco Unified CM Administration.
Secure Tone Cisco Extension Mobility and join across line services are disabled on protected
phones.
User Group Although you can add users to the Standard EM authentication proxy rights user
group, those users are not authorized to authenticate by proxy.
Remember the Last The service parameter Remember the Last User Logged In is applicable only for
User Logged In default Extension Mobility service URL or the Extension Mobility service URL with
loginType as UID.
Extension Mobility Troubleshooting

Troubleshoot Extension Mobility
Procedure
• Configure the Cisco Extension Mobility trace directory and enable debug tracing by performing the
following steps:
a) From Cisco Unified Serviceability, choose Trace > Trace Configuration.
b) From the Servers drop-down list, select a server.
c) From the Configured Services drop-down-list, select Cisco Extension Mobility.
• Make sure that you entered the correct URL for the Cisco Extension Mobility service. Remember that
the URL is case sensitive.
• Check that you have thoroughly and correctly performed all the configuration procedures.
• If a problem occurs with authentication of a Cisco Extension Mobility user, go to the user pages and
verify the PIN.
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Problem “Error 201 Authentication Error” appears on the phone.
Solution The user should check that the correct user ID and PIN were entered; the user should check with the
system administrator that the user ID and PIN are correct.
Blank User ID or PIN

Problem “Error 202 Blank User ID or PIN” appears on the phone.
Solution Enter a valid user ID and PIN.
Busy Please Try Again

Problem “Error 26 Busy Please Try Again” appears on the phone.
Solution Check whether the number of concurrent login and logout requests is greater than the Maximum
Concurrent requests service parameter. If so, lower the number of concurrent requests.
Note To verify the number of concurrent login and logout requests, use the Cisco Unified Real-Time Monitoring
Tool to view the Requests In Progress counter in the Extension Mobility object. For more information, see
the Cisco Unified Real-Time Monitoring Tool Administration Guide at http://www.cisco.com/c/en/us/support/
unified-communications/unified-communications-manager-callmanager/products-maintenance-guides-list.html
Database Error
Problem “Error 6 Database Error” appears on the phone.
Solution Check whether a large number of requests exists. If a large number of requests exists, the Requests
In Progress counter in the Extension Mobility object counter shows a high value. If the requests are rejected
because of a large number of concurrent requests, the Requests Throttled counter also shows a high value.
Collect detailed database logs.
Dev Logon Disabled

Problem “Error 22 Dev Logon Disabled” appears on the phone.
Solution Verify that you checked the Enable Extension Mobility check box in the Phone Configuration
window (Device > Phone).
Device Name Empty

Problem “Error 207 Device Name Empty” appears on the phone.
Solution Check that the URL that is configured for Cisco Extension Mobility is correct. See the Related Topics
section for more information.
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EM Service Connection Error
Related Topics
Configure the Cisco Extension Mobility Phone Service, on page 399

Problem “Error 207 EM Service Connection Error” appears on the phone.
Solution Verify that the Cisco Extension Mobility service is running by selecting Tools > Control
Center—Feature in Cisco Unified Serviceability.
Extension Mobility Performance During Upgrade

Problem Extension Mobility (EM) login performance during Publisher switch version after the upgrade.
Solution If Extension Mobility (EM) users are logged in during the switch version upgrade of Unified
Communications Manager Publisher, and if the Publisher is inactive, EM login data is lost during the switch
version and EM profiles are logged out.
Note If EM login profiles are logged out, users can log in again, or log in only when Unified Communications
Manager is active after the switch version.
Host Not Found

Problem The “Host Not Found” error message appears on the phone.
Solution Check that the Cisco Tomcat service is running by selecting Tools > Control Center—Network
Services in CIsco Unified Serviceability.
HTTP Error
Problem HTTP Error (503) appears on the phone.
Solution
• If you get this error when you press the Services button, check that the Cisco IP Phone Services service
is running by selecting Tools > Control Center—Network Services in Cisco Unified Serviceability.
• If you get this error when you select Extension Mobility service, check that the Cisco Extension Mobility
Application service is running by selecting Tools > Control Center—Network Services in Cisco Unified
Serviceability.
Phone Resets
Problem After users log in or log out, their phones reset instead of restarting.
Possible Cause Locale change is the probable cause of the reset.
Solution No action is required. If the user locale that is associated with the logged-in user or profile is not the
same as the locale or device, after a successful login the phone will restart and then reset. This pattern occurs
because the phone configuration file is rebuilt.
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Phone Services Unavailable After Login

Problem After logging in, the user finds that the phone services are not available.
Possible Cause This problem occurs because the user profile had no services associated with it when it
was loaded on the phone.
Solution
• Ensure that the user profile includes the Cisco Extension Mobility service.
• Change the configuration of the phone where the user is logged in to include Cisco Extension Mobility.
After the phone is updated, the user can access the phone services.
Phone Services Unavailable After Logout

Problem After a user logs out and the phone reverts to the default device profile, the phone services are no
longer available.
Solution
• Verify that the Synchronization Between Auto Device Profile and Phone Configuration enterprise
parameter is set to True.
• Subscribe the phone to the Cisco Extension Mobility service.
User Logged in Elsewhere

Problem “Error 25 User Logged in Elsewhere” appears on the phone.
Solution Check whether the user is logged in to another phone. If multiple logins must be allowed, ensure that
the Multiple Login Behavior service parameter is set to Multiple Logins Allowed.
User Profile Absent

Problem “Error 205 User Profile Absent” appears on the phone.
Solution Associate a device profile to the user.
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User Profile Absent
412
CHAPTER 32
Extension Mobility Cross Cluster
• Extension Mobility Cross Cluster Overview, on page 413
• Extension Mobility Cross Cluster Prerequisites, on page 413
• Extension Mobility Cross Cluster Configuration Task Flow, on page 413
• Extension Mobility Cross Cluster Interactions, on page 434
• Extension Mobility Cross Cluster Restrictions, on page 435
• Extension Mobility Cross Cluster Troubleshooting, on page 439
Extension Mobility Cross Cluster Overview

The extension mobility cross cluster (EMCC) feature provides users with the same functionality as extension
mobility, but also allows them to move from one cluster (the home cluster) and log in to a temporary phone
on another remote cluster (the visiting cluster). From there, they can access their phone settings from any
location as if the they were using an IP phone at the home office.
Extension Mobility Cross Cluster Prerequisites

• Other call-control entities that support and use the extension mobility cross cluster (EMCC) configuration;
for example, other Cisco Unified Communications Manager clusters, EMCC intercluster service profiles,
and EMCC remote cluster services
• Clusters that are set to nonsecure or mixed mode. See Extension Mobility Cross Cluster and Security
Mode for Different Cluster Versions, on page 437 for more information.
• Supported phones in secure or nonsecure mode
Extension Mobility Cross Cluster Configuration Task Flow

Before you begin
• Review Extension Mobility Cross Cluster Prerequisites, on page 413
• Review Extension Mobility Cross Cluster Interaction and Restriction
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Extension Mobility Cross Cluster Configuration Task Flow
Procedure

support the extension mobility cross cluster
feature.
Step 2 To Configure Extension Mobility, on page 415, Configure extension mobility to allow users to
perform the following subtasks: temporarily access their phone settings, such as
line appearances, services, and speed dials, from
• Activate Services for Extension Mobility
other phones in one cluster. Perform this task
Cross Cluster, on page 416
flow on both home and remote clusters, so that
• Configure the Extension Mobility Phone users will be able to access settings from either
Service, on page 416 a home or visiting cluster.
• Configure a Device Profile for Extension
Mobility Cross Cluster, on page 417
• Enable Extension Mobility Cross Cluster
for a User, on page 423
• Subscribe Devices to Extension Mobility,
on page 423
Step 3 To Configure Certificates for Extension To configure the home and remote clusters
Mobility Cross Cluster, on page 423, perform properly, you must export certificates on each
the following subtasks: cluster to the same SFTP server and SFTP
directory and consolidate them on one of the
• Activate the Bulk Provisioning Service,
participating clusters. This procedure ensures
on page 424
that trust is established between the two clusters.
• Configure Bulk Certificate Management
and Export Certificates, on page 424
• Consolidate the Certificates, on page 425
• Import the Certificates into the Clusters,
on page 426
Step 4 To Configure Extension Mobility Cross Cluster

Devices and Templates, on page 427, perform
the following subtasks:
• Create a Common Device Configuration,
on page 427
• Configure an Extension Mobility Cross
Cluster Template, on page 428
• Set the Default Template, on page 428
• Add Extension Mobility Cross Cluster
Devices, on page 429
Step 5 Configure a Geolocation Filter for Extension Configure a geolocation filter to specify criteria
Mobility Cross Cluster, on page 429 for device location matching, such as country,
state, and city values. Geolocations are used to
identify the location of a device, and the filter
indicates what parts of the geolocation are
significant.
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Configure Extension Mobility

Step 6 Configure Feature Parameters for Extension Select values for the feature parameters that you
Mobility Cross Cluster, on page 429 configured, such as the geolocation filter.
Step 7 Configure Intercluster SIP Trunk for Extension Configure trunks to process inbound or
Mobility Cross Cluster, on page 433 outbound traffic for intercluster PSTN access
and RSVP agent services. You can configure
one trunk for both PSTN access and RSVP
agent services or one trunk for each service.
You do not need more than two SIP trunks for
extension mobility cross cluster.
Step 8 Configure an Intercluster Service Profile for Configure the intercluster service profile to
Extension Mobility Cross Cluster, on page 433 activate extension mobility cross cluster. The
profile collects all the configuration that
precedes and provides a results report.
Step 9 Configure Remote Cluster Services, on page Configure the remote cluster for extension
434 mobility cross cluster. This step completes the
link between the home cluster with remote
(visiting) cluster.
Configure Extension Mobility

Configure extension mobility to allow users to temporarily access their phone settings, such as line appearances,
services, and speed dials, from other phones in one cluster. Perform this task flow on both home and remote
clusters, so that users will be able to access settings from either a home or visiting cluster.
Procedure

Step 1 Activate Services for Extension Mobility Cross
Cluster, on page 416
Step 2 Configure the Extension Mobility Phone Create the Extension Mobility phone service to
Service, on page 416 which you can subscribe your users.
Step 3 Configure a Device Profile for Extension Create a device profile to map settings onto a
Mobility Cross Cluster, on page 417 real device when a user logs in to Extension
Mobility cross cluster.
Step 4 Enable Extension Mobility Cross Cluster for a

User, on page 423
Step 5 Subscribe Devices to Extension Mobility, on Enable Extension Mobility on devices and
page 423 subscribe to the service if you have not set up
an enterprise subscription for all devices.
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Activate Services for Extension Mobility Cross Cluster
Activate Services for Extension Mobility Cross Cluster
Procedure
b) Cisco Tftp
d) ILS Service
Step 4 Click Save.

Step 5 Click OK.
Configure the Extension Mobility Phone Service

Create the Extension Mobility phone service to which you can subscribe your users.
Procedure
For example, enter a name such as Extension Mobility or EM. For Java MIDlet services, the service name
must exactly match the name that is defined in the Java Application Descriptor (JAD) file.
Step 4 In the Service URL field, enter the service URL in the following format:
http://<IP Address>:8080/emapp/EMAppServlet?device=#DEVICENAME#&EMCC=#EMCC#.
Step 5 (Optional) If you want to create a secure URL using HTTPS, enter the secure service URL in the following
format:
https://<IP
Address>:8443/emapp/EMAppServlet?device=#DEVICENAME#&EMCC=#EMCC#
Step 6 (Optional) If you want to configure more sign-in options, append the loginType parameter to the Service
URL in the following formats:
• loginType=DN enables users to sign in using Primary Extension and PIN.The Service URL format is:
http://<IP Address>:8080/emapp/EMAppServlet?device=#DEVICENAME#&EMCC=#EMCC#&loginType=DN.
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Configure a Device Profile for Extension Mobility Cross Cluster
The Service URL format is:

http: //<IP
Address>:8080/emapp/EMAppServlet?device=#DEVICENAME#&EMCC=#EMCC#&loginType=SP.

The Service URL format is: http://<IP
Address>:8080/emapp/EMAppServlet?device=#DEVICENAME#&EMCC=#EMCC#&loginType=UID.
The loginType parameter can also be appended to a secure URL. If you do not append loginType to the
end of the URL, the default sign in option displayed is User ID and PIN.
Step 7 Use the default values for the Service Category and Service Type fields.
Step 8 Check the Enable check box.
Step 9 (Optional) Check the Enterprise Subscription check box to subscribe all phones and device profiles to this
phone service.
Note If you check this check box when configuring the service for the first time, you will set up this
IP phone service as an enterprise subscription service. All phones and device profiles in the
enterprise will automatically subscribe to this IP phone service, removing the need for you to
subscribe them individually.
Step 10 Click Save.
Configure a Device Profile for Extension Mobility Cross Cluster

Create a device profile to map settings onto a real device when a user logs in to Extension Mobility cross
cluster.
Procedure
• Click Find to modify an existing device profile, enter search criteria. Click a device profile name in the
resulting list.
• Click Add New to add a new device profile and click Next to choose a device profile type. Click Next
to choose a protocol, then click Next.
Step 3 Configure the fields on the Device Profile Configuration window. See Device Profile Fields for Extension
Mobility Cross Cluster, on page 418 for more information about the fields and their configuration options.
Step 4 Click Save.
Step 5 Add a directory number (DN) to the new device profile.
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Device Profile Fields for Extension Mobility Cross Cluster
Table 34: Device Profile Settings
Field Description
Product Type Displays the product type to which this device profile applies.
Device Protocol Displays the device protocol to which this device profile applies.
Device Profile Name Enter a unique name. This name can comprise up to 50 characters in length.
Description Enter a description of the device profile. For text, use anything that describes this
particular user device profile.
User Hold MOH Audio Specifies the audio source that plays when a user initiates a hold action, choose
Source an audio source from the User Hold MOH Audio Source drop-down list.
If you do not choose an audio source, Unified Communications Manager uses
the audio source that is defined in the device pool or the system default if the
device pool does not specify an audio source ID.
Note You define audio sources in the Music On Hold Audio Source
Configuration window. For access, choose Media Resources >
Music On Hold Audio Source.
User Locale From the drop-down list, choose the locale that is associated with the phone user
interface. The user locale identifies a set of detailed information, including
language and font, to support users.
Unified Communications Manager makes this field available only for phone
models that support localization.
Note If no user locale is specified, Unified Communications Manager
uses the user locale that is associated with the device pool.
If the users require information to display (on the phone) in any
language other than English, verify that the locale installer is
installed before configuring user locale. See the Unified
Communications Manager Locale Installer documentation.
Phone Button Template From the Phone Button Template drop-down list, choose a phone button template.
Tip If you want to configure BLF/SpeedDials for the profile for presence
monitoring, choose a phone button template that you configured
for BLF/SpeedDials. After you save the configuration, the Add a
New BLF SD link displays in the Association Information pane.
For more information on BLF/SpeedDials, see the Feature
Softkey Template From the Softkey Template drop-down list, choose the softkey template from
the list that displays.
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Field Description
Privacy From the Privacy drop-down list, choose On for each phone on which you want
privacy. For more information, see the Feature Configuration Guide for Cisco
Single Button Barge From the drop-down list, choose from the following options:
• Off—This device does not allow users to use the Single Button Barge/cBarge
feature.
• Barge—Choosing this option allows users to press the Single Button Barge
shared-line button on the phone to barge into a call using Barge.
• Default—This device inherits the Single Button Barge/cBarge setting from
the service parameter and device pool settings.
Note If the server parameter and device pool settings are different,
the device will inherit the setting from the service parameter
setting.
For more information, see the Feature Configuration Guide for Cisco Unified
Join Across Lines From the drop-down list, choose from the following options:
• Off—This device does not allow users to use the Join Across Lines feature.
• On—This device allows users to join calls across multiple lines.
• Default—This device inherits the Join Across Lines setting from the service
parameter and device pool settings.
Note If the server parameter and device pool settings are different,
the device will inherit the setting from the service parameter
setting.
For more information, see the System Configuration Guide for Cisco Unified
Always Use Prime Line From the drop-down list, choose one of the following options:
• Off—When the phone is idle and receives a call on any line, the phone user
answers the call from the line on which the call is received.
• On—When the phone is idle (off hook) and receives a call on any line, the
primary line gets chosen for the call. Calls on other lines continue to ring,
and the phone user must select those other lines to answer these calls.
• Default—Unified Communications Manager uses the configuration from
the Always Use Prime Line service parameter, which supports the Cisco
CallManager service.
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Field Description
Always Use Prime Line From the drop-down list, choose one of the following options:
for Voice Message
• On—If the phone is idle, the primary line on the phone becomes the active
line for retrieving voice messages when the phone user presses the Messages
button on the phone.
• Off—If the phone is idle, pressing the Messages button on the phone
automatically dials the voice-messaging system from the line that has a voice
message. Unified Communications Manager always selects the first line
that has a voice message. If no line has a voice message, the primary line
gets used when the phone user presses the Messages button.
• Default—Unified Communications Manager uses the configuration from
the Always Use Prime Line for Voice Message service parameter, which
supports the Cisco CallManager service.
Ignore Presentation To configure call display restrictions and ignore any presentation restriction that
Indicators (internal calls is received for internal calls, check the “Ignore Presentation Indicators (internal
only) calls only)” check box.
Tip Use this configuration in combination with the calling line ID
presentation and connected line ID presentation configuration at
the translation pattern level. Together, these settings allow you to
configure call display restrictions to selectively present or block
calling and/or connected line display information for each call. For
more information about call display restrictions, see the Feature
Do Not Disturb Check this check box to enable Do Not Disturb.
DND Option When you enable DND on the phone, this parameter allows you to specify how
the DND feature handles incoming calls:
• Call Reject—This option specifies that no incoming call information gets
presented to the user. Depending on how you configure the DND Incoming
Call Alert parameter, the phone may play a beep or display a flash
notification of the call.
• Ringer Off—This option turns off the ringer, but incoming call information
gets presented to the device, so that the user can accept the call.
• Use Common Phone Profile Setting—This option specifies that the DND
Option setting from the Common Phone Profile window will get used for
this device.
Note For 7940/7960 phones that are running SCCP, you can only choose
the Ringer Off option. For mobile devices and dual-mode phones,
you can only choose the Call Reject option. When you activate DND
Call Reject on a mobile device or dual-mode phone, no call
information gets presented to the device.
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Field Description
DND Incoming Call Alert When you enable the DND Ringer Off or Call Reject option, this parameter
specifies how a call displays on a phone.
From the drop-down list, choose one of the following options:
• None—This option specifies that the DND Incoming Call Alert setting from
the Common Phone Profile window will get used for this device.
• Disable—This option disables both beep and flash notification of a call but
for the DND Ringer Off option, incoming call information still gets
displayed. For the DND Call Reject option, no call alerts display and no
information gets sent to the device.
• Beep Only—For an incoming call, this option causes the phone to play a
beep tone only.
• Flash Only—For an incoming call, this option causes the phone to display
a flash alert.
Extension Mobility Cross From the drop-down list, choose an existing Calling Search Space (CSS) to use
Cluster CSS for this device profile for the Extension Mobility Cross Cluster feature. (To
configure a new CSS or modify an existing CSS, choose Call Routing > Class
of Control > Calling Search Space in Unified Communications Manager.)
Default value specifies None.
The home administrator specifies this CSS, which gets used as the device CSS
that gets assigned to the phone when the user logs in to this remote phone. For
more information, see the Feature Configuration Guide for Cisco Unified
Module 1 You can configure one or two expansion modules for this device profile by
choosing phone templates from the expansion module drop-down list in the
expansion module fields.
Note You can view a phone button list at any time by choosing the View
button list link next to the phone button template fields. A separate
dialog box pops up and displays the phone buttons for that particular
expansion module.
Choose the appropriate expansion module or None.
Module 2 Choose the appropriate expansion module or None.
MLPP Domain If this user device profile will be used for MLPP precedence calls, choose the
MLLP Domain from the drop-down list.
Note You define MLPP domains in the MLPP Domain Configuration
window. For access, choose System > MLPP Domain.
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Field Description
MLPP Indication If this user device profile will be used for MLPP precedence calls, assign an
MLPP Indication setting to the device profile. This setting specifies whether a
device that can play precedence tones will use the capability when it places an
MLPP precedence call.
From the drop-down list, choose a setting to assign to this device profile from
the following options:
1. Default—This device profile inherits its MLPP indication setting from the
device pool of the associated device.
2. Off—This device does not handle nor process indication of an MLPP
precedence call.
3. On—This device profile does handle and process indication of an MLPP
precedence call.
Note Do not configure a device profile with the following combination

of settings: MLPP Indication is set to Off or Default (when default
is Off) while MLPP Preemption is set to Forceful.
MLPP Preemption If this user device profile will be used for MLPP precedence calls, assign an
MLPP Preemption setting to the device profile. This setting specifies whether a
device that can preempt calls in progress will use the capability when it places
an MLPP precedence call.
From the drop-down list, choose a setting to assign to this device profile from
the following options:
1. Default—This device profile inherits its MLPP preemption setting from the
device pool of the associated device.
2. Disabled—This device does not allow preemption of lower precedence calls
to take place when necessary for completion of higher precedence calls.
3. Forceful—This device allows preemption of lower precedence calls to take
place when necessary for completion of higher precedence calls.
Note Do not configure a device profile with the following combination

of settings: MLPP Indication is set to Off or Default (when default
is Off) while MLPP Preemption is set to Forceful.
Login User Id From the Login User ID drop-down list, choose a valid login user ID.
Note If the device profile is used as a logout profile, specify the login
user ID that will be associated with the phone. After the user logs
out from this user device profile, the phone will automatically log
in to this login user ID.
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Enable Extension Mobility Cross Cluster for a User
Enable Extension Mobility Cross Cluster for a User
Procedure
• Click Find to modify the settings for an existing user and choosing an existing user from the resulting
list.
Step 3 In the Extension Mobility pane, check the Enable Extension Mobility Cross Cluster check box.
Step 4 Choose the device profile from the Available Profiles list pane in the Extension Mobility pane.
Step 5 Move the device profile to the Controlled Profiles list pane.
Step 6 Click Save.
Subscribe Devices to Extension Mobility

Enable Extension Mobility on devices and subscribe to the service if you have not set up an enterprise
subscription for all devices.
Procedure
Step 1 From From Cisco Unified CM Administration, choose Device > Phone.
Step 2 Find the phone on which users can use Extension Mobility Cross Cluster.
Step 3 For this device, check the Enable Extension Mobility check box in the Extension Information pane.
Step 4 In the Phone Configuration window, choose the Subscribe/Unsubscribe Services option in the Related
Links drop-down list.
Step 5 Click Go.
Step 6 In the popup window that opens, choose the Extension Mobility service in the Select a Service drop-down
list.
Step 7 Click Next.
Step 9 From the popup window, click Save, and then close the window.
Step 10 In the Phone Configuration window, click Save.
Step 11 Click OK if prompted.
Configure Certificates for Extension Mobility Cross Cluster

To configure the home and remote clusters properly, you must export certificates on each cluster to the same
SFTP server and SFTP directory and consolidate them on one of the participating clusters. This procedure
ensures that trust is established between the two clusters.
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Activate the Bulk Provisioning Service
Before you begin

Configure Extension Mobility, on page 415
Procedure

Step 1 Activate the Bulk Provisioning Service, on page
424
Step 2 Configure Bulk Certificate Management and Configure bulk certificate management in Cisco
Export Certificates, on page 424 Unified OS Administration to export the
certificates from both the home and remote
clusters.
Step 3 Consolidate the Certificates, on page 425 Consolidate certificates when all participating
clusters have exported their certificates. This
option is available only if two or more clusters
exported their certificates to the SFTP server.
Step 4 Import the Certificates into the Clusters, on page Import the certificates back into the home and
426 remote (visiting) clusters.
Activate the Bulk Provisioning Service
Before you begin

Configure Extension Mobility, on page 415
Procedure
Step 3 Check the Cisco Bulk Provisioning Service check box.
Step 4 Click Save.
Step 5 Click OK.
Configure Bulk Certificate Management and Export Certificates

Configure bulk certificate management in Cisco Unified OS Administration to export the certificates from
both the home and remote clusters.
This procedure creates a PKCS12 file that contains certificates for all nodes in the cluster.
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Consolidate the Certificates
Note • Every participating cluster must export certificates to the same SFTP server and SFTP directory.
• You must export certificates on the cluster whenever the Tomcat, Tomcat-ECDSA, TFTP, or CAPF
certificates are regenerated on any of the cluster nodes.
Procedure
Step 1 From Cisco Unified OS Administration, choose Security > Bulk Certificate Management.
Step 2 Configure the settings for a TFTP server that both the home and remote clusters can reach. See the online
help for information about the fields and their configuration options.
Step 3 Click Save.
Step 4 Click Export.
Step 5 In the Bulk Certificate Export window, choose All for the Certificate Type field.
Step 6 Click Export.
Step 7 Click Close.
Note When the bulk certificate export is performed, the certificates are then uploaded to the remote
cluster as follows:
• CAPF certificate gets uploaded as a CallManager-trust
• Tomcat certificate gets uploaded as a Tomcat-trust
• CallManager certificate gets uploaded as a CallManager-trust
• CallManager certificate gets uploaded as a Phone-SAST-trust
• ITLRecovery certificate gets uploaded as a PhoneSast-trust and CallManager-trust
The above steps are performed when certificates are self-signed and there is no common trust in
another cluster. If there is a common trust or the same signer then the export of ALL certificates
is not needed.
Consolidate the Certificates

Consolidate certificates when all participating clusters have exported their certificates. This option is available
only if two or more clusters exported their certificates to the SFTP server.
This procedure consolidates all PKCS12 files in the SFTP server to form a single file.
Note If you export new certificates after consolidation, you must perform this procedure again to include the newly
exported certificates.
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Import the Certificates into the Clusters
Procedure
Step 1 From From Cisco Unified OS Administration, choose Security > Bulk Certificate Management >
Consolidate > Bulk Certificate Consolidate.
Step 2 In the Certificate Type field, choose All.
Step 3 Click Consolidate.
Note When the bulk certificate consolidate is performed, the certificates are then uploaded to the remote
cluster as follows:
Import the Certificates into the Clusters

Import the certificates back into the home and remote (visiting) clusters.
Note After an upgrade, these certificates are preserved. You do not need to reimport or reconsolidate certificates.
Caution After you import the certificates, the phones on the cluster will automatically restart.
Procedure
Step 1 From From Cisco Unified OS Administration, choose Security > Bulk Certificate Management > Import >
Bulk Certificate Import.
Step 2 From the Certificate Type drop-down list, choose All.
Step 3 Choose Import.
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Configure Extension Mobility Cross Cluster Devices and Templates
Note When the bulk certificate import is performed, the certificates are then uploaded to the remote
cluster as follows:
Note The following types of certificates determines phones that are restarted:
• Callmanager - ALL phones only IF TFTP service is activated on the node the certificate
belongs.
• TVS - SOME phones based on Callmanager group membership.
• CAPF - ALL phones only IF CAPF is activated.
Configure Extension Mobility Cross Cluster Devices and Templates

Procedure

Step 1 Create a Common Device Configuration, on Configure a common device configuration to
page 427 specify the services or features that will be
associated with a particular user.
Step 2 Configure an Extension Mobility Cross Cluster Create an extension mobility cross cluster
Template, on page 428 template to link the common device
configuration with this feature.
Step 3 Set the Default Template, on page 428 Set the extension mobility cross cluster template
that you created as the default template.
Step 4 Add Extension Mobility Cross Cluster Devices, Insert extension mobility cross cluster device
on page 429 entries into your system database. Each device
is identified with a unique name in the format
EMCC1, EMCC2, and so on. The Bulk
Administration Tool assigns device numbers
by obtaining the last one used.
Create a Common Device Configuration

Configure a common device configuration to specify the services or features that will be associated with a
particular user.
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Configure an Extension Mobility Cross Cluster Template
Procedure
• Click Find to modify an existing common device configuration and choose a common device configuration
• Click Add New to add a new common device configuration.
Step 3 Configure the fields on the Common Device Configuration window. For more information on the fields and
Step 4 Click Save.
Configure an Extension Mobility Cross Cluster Template

Create an extension mobility cross cluster template to link the common device configuration with this feature.
Procedure
Step 1 From Cisco Unified CM Administration, choose Bulk Administration > EMCC > EMCC Template.
Step 3 Configure the fields on the EMCC Template Configuration window. For more information on the fields
Step 4 Click Save.
Set the Default Template

Set the extension mobility cross cluster template that you created as the default template.
Procedure
Step 1 From Cisco Unified CM Administration, choose Bulk Administration > EMCC > Insert/Update EMCC.
Step 2 Click Update EMCC Devices.
Step 3 From the Default EMCC Template drop-down list, choose the extension mobility cross cluster device
template that you configured.
Step 4 Click Run Immediately.
Step 6 Verify the success of the job:
a) Choose Bulk Administration > Job Scheduler.
b) Locate the Job ID of your job.
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Add Extension Mobility Cross Cluster Devices
Add Extension Mobility Cross Cluster Devices

Insert extension mobility cross cluster device entries into your system database. Each device is identified with
a unique name in the format EMCC1, EMCC2, and so on. The Bulk Administration Tool assigns device
numbers by obtaining the last one used.
Procedure
Step 1 From From Cisco Unified CM Administration, choose Bulk Administration > EMCC > Insert/Update
EMCC.
Step 2 Click Insert EMCC Devices.
Step 3 Enter the number of devices you are adding in the Number of EMCC Devices to be added field.
Step 4 Click Run Immediately and click Submit.
Step 5 Refresh the window and verify that the Number of EMCC Devices already in database value shows the
number of devices that you added.
Configure a Geolocation Filter for Extension Mobility Cross Cluster

Configure a geolocation filter to specify criteria for device location matching, such as country, state, and city
values. Geolocations are used to identify the location of a device, and the filter indicates what parts of the
geolocation are significant.
Procedure
Step 1 From Cisco Unified CM Administration, choose System > Geolocation Filter.
Step 3 Configure the fields on the Geolocation Filter Configuration window. For more information on the fields
Step 4 Click Save.
Configure Feature Parameters for Extension Mobility Cross Cluster

Select values for the feature parameters that you configured, such as the geolocation filter.
Procedure
Step 1 From Cisco Unified CM Administration, choose Advanced Features > EMCC > EMCC Feature
Configuration.
Step 2 Configure the fields on the EMCC Feature Configuration window. See Feature Parameter Fields for
Extension Mobility Cross Cluster, on page 430 for more information about the fields and their configuration
options.
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Feature Parameter Fields for Extension Mobility Cross Cluster
Step 3 Click Save.

Table 35: Feature Parameter Fields for Extension Mobility Cross Cluster
EMCC Parameter Description
Default TFTP Server for Choose the computer name or IP address of the default TFTP server that devices
EMCC Login Device logging into extension mobility cross cluster (EMCC) from a remote cluster
should use.
Backup TFTP Server for Choose the computer name or IP address of the backup TFTP server that devices
EMCC Login Device logging into EMCC from a remote cluster should use.
Default Interval for Specify the number of minutes that elapse between system checks for expired
Expired EMCC Device EMCC devices.
Maintenance
An expired EMCC device is a device that logged in to EMCC from a remote
cluster, but that, because of a WAN failure or a connectivity issue, the phone
logged out of the visiting cluster. When connectivity was restored, the device
logged back into the visiting cluster.
During this maintenance job, the Cisco Extension Mobility service checks the
Unified Communications Manager database for any expired EMCC devices and
automatically logs them out.
The default value is 1440 minutes. Valid values range from 10 minutes to 1440
minutes.
Enable All Remote Choose whether you want all services on a new remote cluster to be automatically
Cluster Services When enabled when you add a new cluster.
Adding A New Remote
Valid values are True (enable all services on the remote cluster automatically)
Cluster
or False (manually enable the services on the remote cluster via the Remote
Cluster Configuration window in Unified Communications Manager). You can
enable the services manually so that you have time to configure the EMCC feature
completely before enabling the remote services.
The default value is False.
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CSS for PSTN Access SIP Choose the calling search space (CSS) that the PSTN Access SIP trunk for
Trunk processing EMCC calls uses.
The PSTN Access SIP trunk is the SIP trunk that you configured for PSTN access
in the Intercluster Service Profile window. Calls over this trunk are intended
for and are routed to only the local PSTN that is co-located with the EMCC
logged-in phone that initiates the call.
Valid values are the following:
• Use Trunk CSS (PSTN calls use the local route group, which can prove
useful for properly routing emergency service calls)
• Use phone's original device CSS (PSTN calls are routed using the configured
calling search space on the remote phone, that is, the CSS that is used when
the phone is not logged into EMCC).
The default value is Use trunk CSS.
EMCC Geolocation Filter Choose the geolocation filter that you have configured for use EMCC.
Based on the information in the geolocation that associates with a phone that is
logged in through Extension Mobility from another cluster, as well as the selected
EMCC geolocation filter, Cisco Unified Communications Manager places the
phone into a roaming device pool.
Cisco Unified Communications Manager determines which roaming device pool
to use by evaluating which device pool best matches the phone geolocation
information after the EMCC geolocation filter is applied.
EMCC Region Max This parameter specifies the maximum audio bit rate for all EMCC calls,
Audio Bit Rate regardless of the region associated with the other party.
The default value is 8 kbps (G.729).
Note All participating EMCC clusters must specify the same value for
the EMCC region max audio bit rate.
EMCC Region Max This parameter specifies the maximum video call bit rate for all EMCC video
Video Call Bit Rate calls, regardless of the maximum video call bit rate of the region associated with
(Includes Audio) the other party.
The default value is 384. Valid values range from 0 to 8128.
Note All participating EMCC clusters must specify the same value for
the EMCC region max video call bit rate.
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EMCC Region Link Loss This parameter specifies the link loss type between any EMCC phone and devices
Type in any remote cluster.
Note To allow two-way audio on EMCC calls, all participating EMCC
clusters must use the same EMCC region link loss type.
Based on the option that you choose, Cisco Unified Communications Manager
attempts to use the optimal audio codec for the EMCC call while observing the
configured EMCC region max audio bit rate.
Valid values are the following:
• Lossy—A link where some packet loss can or may occur, for example, DSL.
• Low Loss—A link where low packet loss occurs, for example, T1.
When you set this parameter to Lossy, Cisco Unified Communications Manager
chooses the optimal codec within the limit that is set by the EMCC Region Max
Audio Bit Rate, based on audio quality. Some packet loss will occur.
When this parameter is set to Low Loss, Cisco Unified Communications Manager
chooses the optimal codec within the limit that is set by the EMCC Region Max
Audio Bit Rate, based on audio quality. Little or no packet loss will occur.
The only difference in the audio codec preference ordering between the low loss
and lossy options is that G.722 is preferred over Internet Speech Audio Codec
(iSAC) when the link loss type is set as low loss, whereas iSAC is preferred over
G.722 when the link loss type is set as lossy.
The default value is Low Loss.
RSVP SIP Trunk Specify the number of seconds that Unified Communications Manager waits
KeepAlive Timer between sending or receiving KeepAlive messages or acknowledgments between
two clusters over EMCC RSVP SIP trunks.
An EMCC RSVP SIP trunk is a SIP trunk that has Cisco Extension Mobility
Cross Cluster configured as the Trunk Service Type and that has been selected
as the SIP Trunk for RSVP Agent in the Intercluster Service Profile window.
When two of these intervals elapse without receipt of a KeepAlive message or
an acknowledgment, Unified Communications Manager releases the RSVP
resources with the remote cluster.
The default value is 15 seconds. Valid values range from 1second to 600 seconds.
Default Server For Choose the default server name or IP address of the primary node in this local
Remote Cluster Update cluster that has the Cisco Extension Mobility service activated. The remote cluster
accesses this node to get information about this local cluster.
Backup Server for Remote Choose the default server name or IP address of the secondary node in this local
Cluster Update cluster that has the Cisco Extension Mobility service activated. The remote cluster
accesses this node when the primary node is down to retrieve information about
this local cluster.
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Configure Intercluster SIP Trunk for Extension Mobility Cross Cluster
Remote Cluster Update Specify an interval, in minutes, during which the Cisco Extension Mobility service
Interval on the local node collects information about the remote EMCC cluster. Collected
information includes such details as the remote cluster Unified Communications
Manager version and service information.
The default value is 30. Valid values range from 15 minutes to 10,080 minutes.
Configure Intercluster SIP Trunk for Extension Mobility Cross Cluster

Configure trunks to process inbound or outbound traffic for intercluster PSTN access and RSVP agent services.
You can configure one trunk for both PSTN access and RSVP agent services or one trunk for each service.
You do not need more than two SIP trunks for extension mobility cross cluster.
Procedure

Step 4 From the Trunk Service Type drop-down list, choose Extension Mobility Cross Clusters.
Step 5 Click Next.
Step 6 Configure the fields in the Trunk Configuration window. For more information on the fields and their
configuration options, see Online Help.
Step 7 Click Save.
Configure an Intercluster Service Profile for Extension Mobility Cross Cluster

Configure the intercluster service profile to activate extension mobility cross cluster. The profile collects all
the configuration that precedes and provides a results report.
Procedure
Step 1 From Cisco Unified CM Administration, choose Advance Features > EMCC > EMCC Intercluster Service
Profile.
Step 2 Configure the fields on the EMCC Intercluster Service Profile Configuration window. For more information
Step 3 If no failure messages appear in the popup window, click Save.
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Configure Remote Cluster Services
Configure Remote Cluster Services

Configure the remote cluster for extension mobility cross cluster. This step completes the link between the
home cluster with remote (visiting) cluster.
Procedure
Step 1 From From Cisco Unified CM Administration, choose Advanced Features > Cluster View.
Step 2 Click Find to show a list of known remote clusters.
• Click the remote cluster name and verify the fields if the remote cluster that you want to configure appears.
• Click Add New if the remote cluster that you want to configure does not appear and configure the
following fields:
a. For the Cluster Id field, ensure that the ID matches the enterprise parameter value of the cluster ID
of the other clusters.
b. n the Fully Qualified Name field, enter the IP address of the remote cluster or a domain name that
can resolve to any node on the remote cluster.
c. Click Save.
Note For extension mobility cross cluster, the TFTP check box should always be disabled.
Extension Mobility Cross Cluster Interactions

Table 36: Extension Mobility Cross Cluster Interactions
Feature Interaction
Audio The default maximum audio bit-rate for EMCC login device is set to 8 kbps (G.729).
Call Admission • The home cluster is unaware of visiting cluster locations and regions.
Control (CAC)
• The system cannot apply Cisco Unified Communications Manager locations and
regions across the cluster boundaries.
• RSVP agent-based CAC uses RSVP agents in the visiting cluster.
Call Forwarding EMCC supports call forwarding.

Cisco Extension User authentication takes place across clusters.
Mobility login and
logout
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Extension Mobility Cross Cluster Restrictions
Feature Interaction
Media resources for Examples include RSVP agent, TRP, music on mold (MOH), MTP, transcoder, and
the visiting phone conference bridge.
Media resources are local to the visiting phone (other than RSVP agents).
PSTN access for the • E911 calls are routed to the local gateways of the PSTN.
visiting phone
• Local calls are routed to the local gateways of the PSTN.
• Calls terminating to local route groups route to local gateways in the visiting
cluster.
Other call features Example restriction: Intercom configuration specifies configuration to a static device,
and services so EMCC does not support the intercom feature.
Security • Cross-cluster security is provided by default.

• Cisco Unified IP Phones with secure and nonsecure phone security profiles are
supported.
Internet Protocol Cisco Extension Mobility Cross Cluster Supports IPv6. You can use phones with an
Version 6 (IPv6) IP addressing mode of IPv6 or dual-stack (IPv4 and IPv6).

Table 37: Extension Mobility Cross Cluster Restrictions
Unsupported Features • EMCC does not support the intercom feature, because intercom configuration
requires a static device.
• Location CAC is not supported, but RSVP-based CAC is supported.
EMCC Device For EMCC to function properly, you cannot configure the same phone (device name)
Cannot Be in two clusters. Otherwise, login will fail due to the duplicate device error (37). For
Provisioned in More this reason, for cluster deployed with EMCC you should disable Autoregistration on
Than One Cluster all Unified Communication Manager nodes to prevent a new device being created in
the home cluster after EMCC logout.
Number of EMCC Cisco Unified Communications Manager can support a MaxPhones value of 60,000.
Devices
Include EMCC in the total number of devices that are supported in the cluster by
using the following calculation:
Phones + (2 x EMCC devices) = MaxPhones
Note EMCC login does not affect the number of licenses that are used in the
home cluster.
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Visiting Device • If the home cluster administrator disables EMCC for a user while the user is
Logout Limitations logged in with EMCC, the system does not automatically log this user out.
Instead, the system does not allow future EMCC attempts by this user. The
current EMCC session continues until the user logs out.
• In the visiting cluster, the Phone Configuration window has a Log Out button
for extension mobility. This button is also used by the visiting cluster
administrator to log out an EMCC phone. Because the EMCC phone is not
currently registered with the visiting Cisco Unified Communications Manager,
this operation is like a database cleanup in the visiting cluster. The EMCC phone
remains registered with the home Cisco Unified Communications Manager until
the phone returns to the visiting cluster because of a reset or a logout from the
home cluster.
Visiting Device The extension mobility service in participating clusters performs a periodic remote
Login Limitations cluster update. The Remote Cluster Update Interval feature parameter controls the
update interval. The default interval is 30 minutes.
If the extension mobility service on clusterA does not receive a reply from a remote
cluster (such as clusterB) for this update, the Remote Cluster window for clusterA
shows that the Remote Activated service is set to False for clusterB.
In this case, the visiting cluster does not receive any response from the home cluster
and sets the Remote Activated values for the home cluster to False.
During this interval, a visiting phone may not be able to log in by using EMCC. The
visiting phone receives the “Login is unavailable” error message.
At this point, a login attempt to EMCC from a visiting phone can fail; the phone
displays the “Login is unavailable” error message. This error occurs because the
visiting cluster has not yet detected the change of the home cluster from out-of-service
to in-service.
Remote cluster status change is based on the value of the Remote Cluster Update
Interval EMCC feature parameter and on when the visiting extension mobility service
performed the last query or update.
You can select Update Remote Cluster Now in the Remote Cluster Service
Configuration window (Advanced Features > EMCC > EMCC Remote Cluster)
to change Remote Activate values to True, which also allows EMCC logins. Otherwise,
after the next periodic update cycle, EMCC logins by visiting phones will return to
normal.
EMCC Login Result for Different Cluster Versions with loginType

The following table shows the login result of the Extension Mobility Cross Cluster feature for different cluster
versions when the loginType parameter is used in the service URL.
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Extension Mobility Cross Cluster and Security Mode for Different Cluster Versions
Table 38: EMCC Login Result for Different Cluster Versions with loginType
Visiting Cluster Version Home Cluster Version loginType in Visiting EMCC Login Result
Cluster EM URL*
12.0 12.0 Not mentioned (Default Success

URL)
12.0 12.0 UID, SP, or DN Success
12.0 11.5 and below Not mentioned (Default Success

URL)
12.0 11.5 and below UID, SP, or DN Fail

Fail with error code - 1 **
11.5 and below 12.0 Not mentioned (Default Success

URL)
11.5 and below 12.0 UID, SP, or DN *** Success
Note • * Following are the loginType parameter options:

• UID—Users login using User ID and PIN
• SP—Users login using Self Service User ID and PIN
• DN—Users login using Primary Extension and PIN
• ** Fail with error code - 1 — (When EMService could not parse the XML request from
EMApp/EMService)
• *** loginType will be ignored and User ID / PIN login prompt gets populated on the phone
Extension Mobility Cross Cluster and Security Mode for Different Cluster
Versions
Note Phone configuration files can be encrypted only if both the home cluster and visiting cluster versions are 9.x
or later, and when the TFTP encryption configuration flag is enabled.
During EMCC login, if both the visiting cluster and home cluster versions are in 9.x or later, the phone will
behave in various modes as shown in the following table.
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Extension Mobility Cross Cluster and Security Mode for Different Cluster Versions
Table 39: Supported Security Modes When Both Visiting Cluster and Home Cluster Are In 9.x or later Versions
Home Home Cluster Visiting Cluster Visiting Cluster Visiting Phone EMCC Status
Cluster Mode Version Mode Mode
Version
9.x or Mixed 9.x or later Mixed Secure Secure EMCC

later
9.x or Mixed 9.x or later Mixed Non-secure Non-secure

later EMCC
9.x or Mixed 9.x or later Non-secure Non-secure Non-secure

later EMCC
9.x or Non-secure 9.x or later Mixed Secure Login fails

later
9.x or Non-secure 9.x or later Non-secure Non-secure Non-secure

later EMCC
During EMCC login, if the visiting cluster version is 8.x and the home cluster version is 9.x or later, the phone
will behave in various modes as shown in the following table.
Table 40: Supported Security Modes When Visiting Cluster Is In 8.x and Home Cluster Is In 9.x or later Version
Version
9.x or Mixed 8.x Mixed Secure Not supported

later
9.x or Mixed 8.x Mixed Non-secure Non-secure

later EMCC
9.x or Mixed 8.x Non-secure Non-secure Non-secure

later EMCC
9.x or Non-secure 8.x Mixed Secure Not supported

later
9.x or Non-secure 8.x Non-secure Non-secure Non-secure

later EMCC
During EMCC login, if the visiting cluster version is 9.x or later and the home cluster version is 8.x, the phone
will behave in various modes as shown in the following table.
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Extension Mobility Cross Cluster Troubleshooting
Table 41: Supported Security Modes When Visiting Cluster Is In 9.x or later and Home Cluster Is In 8.x Version
Version
8.x Mixed 9.x or later Mixed Secure Login fails
8.x Mixed 9.x or later Mixed Non-secure Non-secure

EMCC
8.x Mixed 9.x or later Non-secure Non-secure Non-secure

EMCC
8.x Non-secure 9.x or later Mixed Secure Login fails
8.x Non-secure 9.x or later Non-secure Secure Non-secure

EMCC
Extension Mobility Cross Cluster Troubleshooting

Extension Mobility Application Error Codes
Table 42: Extension Mobility Application Error Codes
Error Code Phone Display Quick Description Reason
201 Please try to login again (201) Authentication Error If the user is an EMCC user, this e
occur if “EMCC” is not activated
Intercluster Service Profile wind
202 Please try to login again (202) Blank userid or pin The user enters a blank user ID or
204 Login is unavailable (204) Directory server error The EMApp sends this error to the p
IMS could not authenticate the us
given PIN.
205 Login is unavailable (205) User Profile Absent Occurs when the user profile infor
cannot be retrieved from the cache
Logout is unavailable (205)
database.
207 Login is unavailable(207) Device Name Empty Occurs when the device or name ta
in the request URI. This cannot ha
Logout is unavailable(207)
real devices and can occur only if
is sent from third-party applicatio
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Extension Mobility Service Error Codes
208 Login is unavailable(208) EMService Connection Error The visiting EMApp cannot connect t
Visiting EMService. (The service is d
not activated.)
The visiting EMService cannot conne
Home EMService (the WAN is down
certificates are not trusted.)
210 Login is unavailable(210) Init Fail-Contact Admin An error (such as a database connecti
failure) occurred while initializing EM
The error can occur because of a failu
connect to the database during startup
211 Login is unavailable(211) EMCC Not Activated Occurs when the PSTN is not activate
Intercluster Service Profile window
visiting cluster.
212 Login is unavailable(212) Cluster ID is invalid Occurs when a remote cluster update
sending an incorrect cluster ID to the
cluster.
213 Login is unavailable(213) Device does not support EMCC Occurs when a device does not support
215 LoginType invalid(215) Login Type is invalid Occurs when loginType is invalid. T
allowed values are:
• SP for Self-service User ID
• DN for Primary Extension
• UID for User ID
216 DN has multiple users(216) DN has multiple users Occurs when the Extension used for E
is assigned for multiple users as Prim

Table 43: Extension Mobility Service Error Codes
0 Login is unavailable(0) Unknown Error The EMService failed for an unknown re

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1 Login is unavailable(1) Error on parsing When the EMService cannot parse th

request from the EMApp or EMService
occurs when third-party applications
incorrect query to login XML (EM AP
can also occur because of a version m
between home and visiting clusters.
2 Login is unavailable(2) EMCC Authentication Error The EMCC user credentials cannot be
authenticated because the user entered
PIN.
3 Login is unavailable(3) Invalid App User Invalid application user. This error co
occurs because of the EM API.
4 Login is unavailable(4) Policy Validation error The EM Service sends this error when
validate the login or logout request be
unknown reason, an error while query
database or an error while retrieving i
from the cache.
5 Login is unavailable(5) Dev. logon disabled A user logs into a device that has Ena
Extension Mobility unchecked in the
6 Login is unavailable(6) Database Error Whenever the database returns an exce

executing the query or stored procedu
EM Service requests (login/logout or
query), the EM Service sends this erro
EMApp.
8 Login is unavailable(8) Query type undetermined No valid query was sent to the EMSe
(DeviceUserQuery and UserDeviceQue
ones). This error occurs because of th
or incorrect XML input.
9 Login is unavailable(9) Dir. User Info Error This error appears in two cases:
Logout is unavailable(9) 1. IMS returns an exception when it
authenticate a user.
2. When information about a user ca
retrieved either from the cache or
10 Login is unavailable(10) User lacks app proxy rights The user tries to log in on behalf of an
By default, a CCMSysUser has admin
rights.
11 Login is unavailable(11) Device Does not exist The phone record entry is absent in th
table.
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12 Phone record entry is absent Dev. Profile not found No device profile is associated with the r
in the device table user.
18 Login is unavailable(18) Another user logged in Another user is already logged in on the p
19 Logout is unavailable(19) No user logged in The system attempted to log out a user w
not logged in. This error occurs when sen
logout requests from third-party applicatio
API).
20 Login is unavailable(20) Hoteling flag error Enable Extension Mobility is unchecke

Phone Configuration window.
21 Login is unavailable(21) Hoteling Status error The current user status was not retrieved
either the local cache or database.
22 Login is unavailable(22) Dev. logon disabled Occurs when EM is not enabled on device
request is sent via EM API or when the s
button is pressed on phone.
23 Login is Unavailable (23) User does not exist Occurs when the given user ID is not fou
any of the remote clusters).
Logout is Unavailable (23)
25 Multi-Login Not Allowed User logged in elsewhere The user is currently logged in to some o
(25) phone either in the local cluster or remote
26 Login is unavailable(26) Busy, please try again Occurs when the EMService has currently
the threshold level of Maximum Concurr
Requests service parameter.
28 Login is unavailable(28) Untrusted IP Error Occurs when the Validate IP Address ser
parameter is set to True and the user tries
in or log out from a machine whose IP ad
not trusted. For example, a third-party app
or EM API from a machine is not listed i
Trusted List of Ips service parameter.
29 Login is unavailable(29) ris down-contact admin The Real-Time Information Server Data C
(RISDC) cache is not created or initialize
the EMService is unable to connect to RI
30 Login is unavailable(30) Proxy not allowed When login and logout occur through prox
is set in HTTP header) and the Allow Pro
service parameter is set to False.
31 Login is unavailable(31) EMCC Not Activated for the user Occurs when the Enable Extension Mob
Cross Cluster check box is not checked
End User Configuration window of the
cluster.
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32 Login is unavailable(32) Device does not support EMCC Occurs when a device model does not h
capability.
33 Login is unavailable(33) No free EMCC dummy device Occurs when all the EMCC dummy d
in use by other EMCC logins.
35 Login is unavailable(35) Visiting Cluster Information is not Occurs when the home cluster does n
present in Home Cluster entry for this visiting cluster.
36 Login is unavailable(36) No Remote Cluster Occurs when the administrator has no

remote cluster.
37 Login is Unavailable (37) Duplicate Device Name Occurs when the same device name ex
the home cluster and visiting cluster.
Logout is Unavailable (37)
38 Login is unavailable(38) EMCC Not Allowed Occurs when the home cluster does n
allow EMCC login (The Enable Exte
Mobility Cross Cluster check box is n
in the home cluster).
39 Please try to login again (201) Configuration Issue Occurs when the Default TFTP Serv
Backup TFTP Server for EMCC log
are not set properly in EMCC Feature
Configuration Page.
Note This is internal error cod
40 Please try to login again (23) No Response from Remote Host Occurs when response not getting fro
Host.
41 PIN change is required PIN change is required Occurs when admin enables User Mu
at Next Login for PIN. In that case u
redirected to Change credentials page
42 Login is unavailable(42) Invalid ClusterID Occurs when the remote cluster ID is

This error can occur during a remote
update.
43 Login is unavailable(43) Device Security mode error The Device Security Profile that is as
the EMCC device should be set to No
its Device Security Mode.
44 Please try to login again (201) Configuration Issue Occurs when the cluster ID is not vali
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45 Login is unsuccessful(45) Remote Cluster version not Occurs during EMCC login when the vis
supported cluster version is 9.x and is in mixed mod
phone is in secure mode, and the home cl
version is 8.x.
46 Login is unsuccessful(46) Remote Cluster security mode not Occurs during EMCC login when the vis
supported cluster security mode is in mixed mode, th
is in secure mode, and the home cluster is
nonsecure mode.
47 DN has multiple users(47) DN has multiple users Occurs during EMCC login when the Ex
used for login is assigned for multiple use
Primary.
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CHAPTER 33
Extension Mobility Roaming Across Clusters
Note To deploy Extension Mobility Roaming Across Clusters, you must be running a minimum release of Cisco
Unified Communications Manager 12.0(1)SU1.
• Extension Mobility Roaming Across Clusters Overview, on page 445

• System Requirements for Extension Mobility Roaming Across Clusters, on page 446
• Extension Mobility Roaming Across Clusters Login, on page 446
• ILS Interaction, on page 449
• Extension Mobility Roaming Across Clusters Task Flow, on page 449
• Extension Mobility Roaming Across Clusters Interactions and Restrictions, on page 454
• Different Types of Extension Mobility, on page 454
• Extension Mobility Roaming Across Clusters Troubleshooting, on page 455
Extension Mobility Roaming Across Clusters Overview

Extension Mobility Roaming Across Clusters gives users the ability to roam across multiple cluster and make
or receive calls even when the user's home cluster is down. This feature leverages the Intercluster Lookup
Service (ILS) to replicate Extension Mobility users' directory numbers across all the clusters.
When a user logs in to a roaming cluster, their phone registers to the roaming cluster using the directory
number. Unlike Extension Mobility Cross Cluster (EMCC), where the phone from the visiting cluster registers
to the home cluster, the roaming feature allows the user to maintain their registration in whichever cluster
they are visiting.
Configuration Overview
To deploy this feature, you must do the following:
• Set up an ILS network—ILS is used to synchronize directory number across the clusters.
For details on configuring ILS, see the Configure Intercluster Lookup Service chapter in the System
• Set up a uniform dial plan—you require a uniform dial plan across the ILS network.
To set up a dial plan, see Configure the Dial Plan chapter in the System Configuration Guide for Cisco
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System Requirements for Extension Mobility Roaming Across Clusters
• Device profile and user information must be synced in all the clusters.
• Configure Extension Mobility.
• Configure Roaming access to your Extension Mobility users.
System Requirements for Extension Mobility Roaming Across

Clusters
The following system requirements exist for Cisco Unified Communications Manager:
• Cisco Unified Communications Manager, Release 12.0(1)SU1 or higher.
• Cisco Extension Mobility service must be running.
• Intercluster Lookup Service must be running.
Extension Mobility Roaming Across Clusters Login

Login Terminology
The following figure depicts the home cluster versus a roaming cluster in Extension Mobility Roaming across
Cluster.
Figure 7: Home Cluster vs. Roaming Cluster
Home Cluster
Home cluster is a cluster, where the user configuration such as User Device profile, Dial Plans reside
here.
Roaming Cluster
Roaming cluster is a cluster, where users can do the Extension Mobility login to any Extension Mobility
capable phone just like in their home cluster.
Superuser
A superuser is a user, who is associated to the Standard EM Roaming Across Clusters Super Users
access control group. This user has a privilege to do the Extension Mobility login from a roaming cluster
and can make or receive calls.
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Note Superuser information must be shared across all the clusters irrespective of the cluster in which the user
is logging-in.
Login Process
Cisco Unified Communications Manager supports the Extension Mobility login for a superuser created across
multiple clusters. Extension Mobility login, in the roaming cluster allows superuser to access their phone
settings, such as line appearances, services, dial plans. A superuser can make or receive calls from the roaming
cluster, in the same way as they do in the home cluster.
Figure 8: Call Flow When a User Is in Home Cluster
In the preceding figure, let us assume Bob's DN as 1000-001, Alice's DN as 2000-001 and Kally's DN as
3000-001 registered with Cluster 1, 2 and 3 respectively. When Kally dials Bob's DN 1000-001, Cluster 3
routes the call to Cluster 1 and Bob and Kally are connected.
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Figure 9: Call Flow When User Is in Roaming Cluster
Let us assume Bob's home cluster is down and Bob is configured as superuser who can roam across the
clusters. When Bob moves to Cluster 2 and does Extension Mobility Login, hosting phone gets re-registered
with Bob's settings. Once the login is successful, all other clusters are updated with Bob's new location. Now
when Kally dials Bob's DN 1000-01, Cluster 3 routes the call to Cluster 2 and Bob and Kally are connected.
Similarly, Bob can call Kally by dialing DN 3000-001.
Note • If a superuser did the Extension Mobility login to another cluster, user will automatically log out from
the home cluster. If the cluster is down, it waits until the cluster is up to log out from the user's previous
login.
• Extension Mobility Roaming Across Clusters supports the multi login behavior. Hence, superuser can
login from multiple devices within the same cluster but not across the clusters.
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ILS Interaction
ILS Interaction
In Cisco Unified CM Administration, you can configure ILS on a pair of clusters and then join those clusters
to form an ILS network. Once you have established the ILS network, you can join additional clusters to the
network without having to configure the connections between each cluster.
Whenever Extension Mobility login or logout occurs, ILS sync starts to update the available information to
other clusters.
Note Configuring user as superuser automatically initiates the ILS sync irrespective of Directory Number
configuration for ILS.
For more information, see the Configure Intercluster Lookup Service chapter in the System Configuration
Guide for Cisco Unified Communications Manager at https://www.cisco.com/c/en/us/support/
unified-communications/unified-communications-manager-callmanager/
products-installation-and-configuration-guides-list.html.
Extension Mobility Roaming Across Clusters Task Flow

Procedure

support the Extension Mobility feature.
Step 2 Configure Extension Mobility by completing, Configure Extension Mobility to allow users to
the following subtasks in order: temporarily access their phone settings, such as
line appearances, services, and speed dials, from
• Activate Extension Mobility Services, on
other phones when they login from remote
page 398
cluster. Perform this task flow on both home
• Configure the Cisco Extension Mobility and remote clusters, so that users will be able
Phone Service, on page 399 to access settings from either a home or remote
• Create an Extension Mobility Device cluster.
Profile for Users, on page 400
• Associate a Device Profile to a User, on
page 400
• Subscribe to Extension Mobility, on page
401
Step 3 Configure Roaming for Extension Mobility Use this procedure to give Extension Mobility
Users, on page 453 users the ability to roam between different
clusters in an ILS network, while using the same
login credentials.
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Procedure


Activate Extension Mobility Services

Procedure
b) Cisco Tftp
d) ILS Service
Step 4 Click Save.

Step 5 Click OK.

Configure the extension mobility IP phone service to which users can later subscribe to access extension
mobility.
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Procedure
Step 4 In the Service URL field, enter the Service URL.
The format is http://<IP Address>:8080/emapp/EMAppServlet?device=#DEVICENAME#.
IP Address is the IP address of the Unified Communications Manager where Cisco Extension Mobility
is activated and running.
It can either be a IPv4 or a IPv6 address.
Example:
http://123.45.67.89:8080/emapp/EMAppServlet?device=#DEVICENAME#
Example:
http://[2001:0001:0001:0067:0000:0000:0000:0134]:8080/emapp/EMAppServlet?device=#DEVICENAME#
This format allows a user to sign-in using User ID and PIN. You can configure more sign-in options for IP
phone users who have subscribed to the extension mobility service. To configure more sign-in options, append
the loginType parameter to the Service URL, in the following formats:
• loginType=DN enables users to sign in using Primary Extension and PIN.
Address>:8080/emapp/EMAppServlet?device=#DEVICENAME#&loginType=DN.
Address>:8080/emapp/EMAppServlet?device=#DEVICENAME#&loginType=SP.
Address>:8080/emapp/EMAppServlet?device=#DEVICENAME#&loginType=UID.
If you do not append loginType to the end of the URL, the default sign-in option displayed is User ID and
PIN.
Step 5 In the Service Type field, choose whether the service is provisioned to the Services, Directories, or Messages
button.
Step 6 Click Save.

Configure an extension mobility device profile. This profile acts as a virtual device that maps onto a physical
device when a user logs in to extension mobility. The physical device takes on the characteristics in this
profile.
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Procedure
• Click Find to modify the settings and choose an existing device profile from the resulting list.
• Click Add New to add a new device profile and choose an option from the Device Profile Type. Click
Next.
• Choose a device protocol from the Device Protocol drop-down list and click Next.
Step 3 Configure the fields. For more information on the fields and their configuration options, see Online Help.
Step 4 Click Save.
Step 5 From the Association Information section, click Add a new DN.
Step 6 In the Directory Number field, enter the directory number and click Save.
Step 7 Click Reset and follow the prompts.

Associate a device profile to users so that they can access their settings from a different phone. You associate
a user device profile to a user in the same way that you associate a physical device.
Tip You can use the Bulk Administration Tool (BAT) to add and delete several user device profiles for Cisco
Extension Mobility at one time. See the Bulk Administration Guide for Cisco Unified Communications
Manager.
Procedure
• Click Find to modify the settings for an existing user, enter search criteria, and choosing an existing user
Step 3 Under Extension Mobility, locate the device profile that you created and move it from Available Profiles
to Controlled Profiles.
Step 4 Check the Home Cluster check box.
Step 5 Click Save.
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Subscribe IP phones and device profiles to the extension mobility service so that users can log in, use, and
log out of extension mobility.
Procedure
Step 1 Perform one of the following tasks from Cisco Unified CM Administration:
• Choose Device > Phone, specify search criteria, click Find, and choose a phone which users will use
for extension mobility.
• Choose Device > Device Settings > Device Profile, specify search criteria, click Find, and choose the
device profile that you created.
Step 2 From the Related Links drop-down list, choose Subscribe/Unsubscribe Services, and then click Go.
Step 3 From the Select a Service drop-down list, choose the Extension Mobility service.
Step 4 Click Next.
Step 6 Click Save and close the popup window.
Configure Roaming for Extension Mobility Users

Use this procedure to give Extension Mobility users the ability to roam between different clusters in an ILS
network, while using the same login credentials. To do this, you must assign the selected users to the Standard
EM Roaming Across Clusters Super Users access control group.
Before you begin

An ILS network must have been set up as ILS is used to replicate user and login information across the clusters.
Procedure
Step 1 From Cisco Unified CM Administration, choose User Management > User Settings > Access Control
Group.
Step 2 Click Find and select the Standard EM Roaming Across Clusters Super Users group.
Step 3 Click Add End Users to Group button. The Find and List Users pop-up window appears.
Step 4 Click Find and select all the users to whom you want to provide roaming ability.
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Extension Mobility Roaming Across Clusters Interactions and Restrictions
Extension Mobility Roaming Across Clusters Interactions and

Restrictions
Extension Mobility Roaming Across Clusters Interactions
This section lists the interactions of the Extension Mobility Roaming across Cluster with other Cisco Unified
Communications Manager Administration components.
• Extension Mobility
• Inter-Cluster Lookup Service (ILS)
Extension Mobility Roaming Across Clusters Restrictions

This section lists the restrictions of the Extension Mobility Roaming across Cluster with other Cisco Unified
Communications Manager Administration components.
• If the hub ILS is down, the spokes connected to it will not synchronize until the hub is back.
Different Types of Extension Mobility

The following table lists the different types of Extension Mobility features available in Cisco Unified
Communications Manager and their differences.
Table 44: Differences Between EM, EMCC, and Extension Mobility Roaming Across Cluster
Extension Mobility (EM) Extension Mobility Cross Extension Mobility

Cluster (EMCC) Roaming Across Cluster
Description Allows users to Allows users to access Allows user to roam

temporarily access their their phone settings from across other clusters using
phone settings from other a phone in another cluster. own login credentials.
phones in the same
cluster.
When the user logs in to N/A The remote cluster phone The roaming cluster
a phone in another cluster registers to the user’s phone registers in the
home cluster, accessing roaming cluster only.
the settings in the home
cluster.
Intercluster Single cluster only Multiple clusters Multiple clusters
Configuration Single cluster only EMCC must be Extension Mobility

configured in the home Roaming must be
cluster and each cluster configured in all the
that the user visit. cluster.
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Extension Mobility Roaming Across Clusters Troubleshooting
Extension Mobility (EM) Extension Mobility Cross Extension Mobility

Cluster (EMCC) Roaming Across Cluster
User Information Single cluster only Must be maintained in all Superuser information
clusters. maintained in all the
cluster.
Extension Mobility Roaming Across Clusters Troubleshooting

This section provides information about error codes for EMApp and EMService.
Problem “Error 201 Authentication Error” appears on the phone.
Solution The user should check that the correct user ID and PIN were entered; the user should check with the
system administrator that the user ID and PIN are correct.
Blank User ID or PIN

Problem “Error 202 Blank User ID or PIN” appears on the phone.
Solution Enter a valid user ID and PIN.
Busy Please Try Again

Problem “Error 26 Busy Please Try Again” appears on the phone.
Solution Check whether the number of concurrent login and logout requests is greater than the Maximum
Concurrent requests service parameter. If so, lower the number of concurrent requests.
Note To verify the number of concurrent login and logout requests, use the Cisco Unified Real-Time Monitoring
Tool to view the Requests In Progress counter in the Extension Mobility object. For more information, see
the Cisco Unified Real-Time Monitoring Tool Administration Guide at http://www.cisco.com/c/en/us/support/
unified-communications/unified-communications-manager-callmanager/products-maintenance-guides-list.html
Database Error
Problem “Error 6 Database Error” appears on the phone.
Solution Check whether a large number of requests exists. If a large number of requests exists, the Requests
In Progress counter in the Extension Mobility object counter shows a high value. If the requests are rejected
because of a large number of concurrent requests, the Requests Throttled counter also shows a high value.
Collect detailed database logs.
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Dev Logon Disabled
Dev Logon Disabled

Problem “Error 22 Dev Logon Disabled” appears on the phone.
Solution Verify that you checked the Enable Extension Mobility check box in the Phone Configuration
window (Device > Phone).
Device Name Empty

Problem “Error 207 Device Name Empty” appears on the phone.
Solution Check that the URL that is configured for Cisco Extension Mobility is correct. See the Related Topics
section for more information.
Related Topics
Configure the Cisco Extension Mobility Phone Service, on page 399

Problem “Error 207 EM Service Connection Error” appears on the phone.
Solution Verify that the Cisco Extension Mobility service is running by selecting Tools > Control
Center—Feature in Cisco Unified Serviceability.
Host Not Found

Problem The “Host Not Found” error message appears on the phone.
Solution Check that the Cisco Tomcat service is running by selecting Tools > Control Center—Network
Services in CIsco Unified Serviceability.
HTTP Error
Problem HTTP Error (503) appears on the phone.
Solution
• If you get this error when you press the Services button, check that the Cisco IP Phone Services service
is running by selecting Tools > Control Center—Network Services in Cisco Unified Serviceability.
• If you get this error when you select Extension Mobility service, check that the Cisco Extension Mobility
Application service is running by selecting Tools > Control Center—Network Services in Cisco Unified
Serviceability.
Phone Resets
Problem After users log in or log out, their phones reset instead of restarting.
Possible Cause Locale change is the probable cause of the reset.

Solution No action is required. If the user locale that is associated with the logged-in user or profile is not the
same as the locale or device, after a successful login the phone will restart and then reset. This pattern occurs
because the phone configuration file is rebuilt.
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Problem After logging in, the user finds that the phone services are not available.
Possible Cause This problem occurs because the user profile had no services associated with it when it
was loaded on the phone.
Solution
• Ensure that the user profile includes the Cisco Extension Mobility service.
• Change the configuration of the phone where the user is logged in to include Cisco Extension Mobility.
After the phone is updated, the user can access the phone services.
Phone Services Unavailable After Logout

Problem After a user logs out and the phone reverts to the default device profile, the phone services are no
longer available.
Solution
• Verify that the Synchronization Between Auto Device Profile and Phone Configuration enterprise
parameter is set to True.
• Subscribe the phone to the Cisco Extension Mobility service.
User Logged in Elsewhere

Problem “Error 25 User Logged in Elsewhere” appears on the phone.
Solution Check whether the user is logged in to another phone. If multiple logins must be allowed, ensure that
the Multiple Login Behavior service parameter is set to Multiple Logins Allowed.
User Profile Absent

Problem “Error 205 User Profile Absent” appears on the phone.
Solution Associate a device profile to the user.
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User Profile Absent
458
CHAPTER 34
Hold Reversion
• Hold Reversion Overview, on page 459
• Hold Reversion Prerequisites, on page 459
• Hold Reversion Configuration Task Flow, on page 460
• Hold Reversion Interactions, on page 463
• Hold Reversion Restrictions, on page 464
Hold Reversion Overview

When you place a call on hold, the Hold Reversion feature alerts you when the held call exceeds a configured
time limit. When the configured time limit expires, an alert is generated on your phone to remind you to handle
the call.
The following alerts are available:
• The Phone rings or beeps once
• The status line displays “Hold Reversion”
• The LED next to the line button flashes continuously
• A vibrating handset icon displays
Note The type of alert that you receive depends on the capabilities of your phone.
To retrieve a reverted call, you can:

• Pick up the handset
• Press the speaker button on the phone
• Press the headset button
• Select the line that is associated with the reverted call
• Press the Resume softkey
For details, see the user guide for your particular phone model.
Hold Reversion Prerequisites

• Cisco CallManager service must be running on at least one node in the cluster
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Hold Reversion Configuration Task Flow
• Cisco CTIManager service must be running on at least one node in the cluster
• Cisco Database Layer Monitor service must be running on the same node as the Cisco CallManager
service
• Cisco RIS Data Collector service must be running on the same node as the Cisco CallManager service
• Cisco Tftp service must be running on at least one node in the cluster
• Cisco Unified Communications Manager Locale Installer must be installed, if you want to use non-English
phone locales or country-specific tones
Hold Reversion Configuration Task Flow

Perform the following steps to configure Hold Reversion for your phones. This procedure assumes that you
have configured directory numbers for phones, or that you are using auto-registration.
Before you begin

• If phone users want the hold reversion messages to display in a language other than English, or if you
want the user to receive country-specific tones for calls, verify that you have installed the locale installer.
• Review Hold Reversion Prerequisites, on page 459
Procedure

Step 1 Generate a Phone Feature List, on page 1 Run a phone feature list report to determine
which phones support the Hold Reversion
feature.
Step 2 Configure Call Focus Priority for Hold Configure the call focus priority setting against
Reversion, on page 460 the device pool for your phones.
Step 3 Perform one of the following procedures: Configure the Hold Reversion timer settings.
You can configure the timer using a clusterwide
• Configure Hold Reversion Timer Defaults
service parameter, or configure the settings on
for Cluster, on page 461
an individual phone line.
• Configure Hold Reversion Timer Settings
for Phone, on page 462 Note The settings on an individual
phone line override the
clusterwide service parameter
settings.
Configure Call Focus Priority for Hold Reversion

As an administrator, you can prioritize incoming calls and reverted calls. By default, all incoming calls are
handled before reverted calls, however you can change the call focus priority so that reverted calls take
precedence.
Before you begin

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Configure Hold Reversion Timer Defaults for Cluster
Procedure
Step 1 From Cisco Unified CM Administration, choose System > Device Pool and open the device pool that applies
to your phones.
Step 2 In the Reverted Call Focus Priority field, choose one of the following options and click Save:
• Default—Incoming calls have priority over reverted calls.
• Highest—Reverted calls have priority over incoming calls.
Step 3 Click Save.

Step 4 Reset any devices in the Device Pool by performing the following steps:
a) Click Reset. The Device Reset window displays.
b) In the Device Reset window, click Reset.
What to do next
Perform one of the following procedures to configure Hold Reversion Timer Settings:
• Configure Hold Reversion Timer Defaults for Cluster, on page 461
• Configure Hold Reversion Timer Settings for Phone, on page 462
Configure Hold Reversion Timer Defaults for Cluster

Perform this procedure to configure clusterwide service parameters that apply hold reversion timer default
settings for all phones in the cluster.
Note When you configure the clusterwide service parameters, the configuration is applied as the default hold
reversion setting for all phones in the cluster. However, the settings on an individual phone line can override
the clusterwide defaults.
Before you begin

Configure Call Focus Priority for Hold Reversion, on page 460
Procedure
Step 2 From the Server drop-down list, choose the server that is running the CallManager service.
Step 4 Configure values for the following clusterwide service parameters:
• Hold Reversion Duration—Enter a number from 0 to 1200 (inclusive) to specify the wait time in seconds
before Cisco Unified Communications Manager issues a reverted call alert to the holding party phone.
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Configure Hold Reversion Timer Settings for Phone
If you enter 0, Cisco Unified Communications Manager does not issue reverted call alerts, unless it is
configured on a phone line.
• Hold Reversion Interval Notification—Enter a number from 0 to 1200 (inclusive) to specify the wait
time in seconds before Cisco Unified Communications Manager sends periodic reminder alerts to the
holding party phone. If you enter 0, Cisco Unified Communications Manager does not send periodic
reminder alerts unless the timer is configured on a phone line.
Step 5 Click Save.
Configure Hold Reversion Timer Settings for Phone

Perform this procedure to configure Hold Reversion timer settings for a phone and phone line.
Note You can also configure Hold Reversion timer settings using a clusterwide service parameter. However, the
settings on an individual phone line override the clusterwide service parameter setting.
Before you begin

Perform Configure Hold Reversion Timer Defaults for Cluster, on page 461 to configure Hold Reversion
clusterwide defaults.
Procedure
Step 1 In Cisco Unified CM Administration, choose Device > Phone.

Step 2 Click Find and select the phone on which you want to configure Hold Reversion.
Step 3 In the Association pane on the left, click the phone line on which you want to configure Hold Reversion.
Step 4 Configure values for the following fields:
• Hold Reversion Ring Duration—Enter a number from 0 to 1200 (inclusive) to specify the wait time
in seconds before Cisco Unified Communications Manager issues a reverted call alert. If you enter 0,
Cisco Unified Communications Manager does not issue reverted call alerts to this DN. If you leave the
field empty (the default setting), Cisco Unified Communications Manager applies the setting from the
Hold Reversion Duration service parameter.
• Hold Reversion Ring Interval Notification—Enter a number from 0 to 1200 (inclusive) to specify the
wait time in seconds before Cisco Unified Communications Manager sends periodic reminder alerts. If
you enter 0, Cisco Unified Communications Manager does not send periodic reminder alerts to this DN.
If you leave the field empty (the default setting), Cisco Unified Communications Manager applies the
setting from the Hold Reversion Interval Notification service parameter.
Step 5 Click Save.

Step 6 Reset the phone by performing the following steps:
a) Click Reset. The Reset Device window displays.
b) Click Reset.
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Hold Reversion Interactions
Hold Reversion Interactions

Table 45: Hold Reversion Feature Interactions
Music on Hold MOH is supported on a reverted call if MOH is configured for a normal held call.
Call Park If hold reversion is invoked and the held party presses the Park softkey, the holding
party still receives hold reversion alerts and can retrieve the call. When the holding
party retrieves the call, the holding party receives MOH, if configured.
If the held party parks before the hold duration exceeds the configured time limit, the
system suppresses all hold reversion alerts until the call is picked up or redirected.
MLPP When a multilevel precedence and preemption (MLPP) call is put on hold and reverts,
the MLPP call loses its preemption status, and the reverted call gets treated as a routine
call.
After the call reverts, the system does not play a preemption ring. If a high precedence
call becomes a reverted call, the system does not play a precedence tone.
CTI Applications CTI applications can access hold reversion functionality when the feature is enabled
for a line or the system. Cisco-provided applications such as Cisco Unified
Communications Manager Assistant and attendant console provide hold reversion
functionality using the CTI interface.
When hold reversion gets invoked, the CTI port receives event notification instead of
the audible tone presented on Cisco Unified IP Phones. CTI ports and route points
receive the event notification once only, whereas Cisco Unified IP Phones receive
alerts at regular intervals.
See the following API documents for information about CTI requirements and
interactions with hold reversion:
• Cisco Unified Communications JTAPI Developer Guide
• Cisco Unified Communications TAPI Developer Guide
Hold Reversion SCCP phones support a minimum Hold Reversion Notification Interval (HRNI) of 5
Interval for SCCP seconds, whereas SIP phones support a minimum of 10 seconds. SCCP phones set
phones when for the minimum HRNI of 5 seconds may experience a Hold Reversion Notification
interacting with SIP ring delay of 10 seconds when handling calls involving SIP phones.
Phones
Shared Lines If a Cisco Unified IP Phone that supports hold reversion shares a line with a phone
device that does not support hold reversion, the hold reversion configuration settings
display only for the line on the supporting device.
If a shared line device disables the feature, hold reversion gets disabled on all other
devices that share the line.
Ring Settings If the ring settings that are configured for the phone specify Disabled, the phone does
not ring, flash, or beep for the hold reversion feature.
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Hold Reversion Restrictions
Hold Reversion Restrictions

Feature Restriction
Cisco Extension Mobility Cisco Extension Mobility and Cisco Web Dialer features do not support the hold
and Cisco Web Dialer reversion feature.
Directory numbers If a directory number is associated to a phone that does not support hold reversion,
the feature settings do not display for that directory number in the Directory
Number Configuration window.
Shared lines If a Cisco Unified IP Phone that supports hold reversion shares a line with a
phone device that does not support hold reversion, the hold reversion configuration
settings display only for the line on the supporting device.
If a shared-line device disables this feature, hold reversion gets disabled on all
other devices that share this line.
Ring settings Hold reversion ring uses the ring settings that Cisco Unified Communications
Manager Administration defines for that user (disable, flash only, ring once, ring,
beep only) except that flash gets converted to flash once, and ring gets converted
to ring once.
Note When an IP Phone call is on normal hold, the ring settings (Phone
Idle) from the Call Manager is applied.
Maximum number of The maximum number of reverted calls on a line equals the maximum number
reverted calls of calls on your system.
Cisco Unified IP Phones You cannot configure hold reversion settings for DNs that are associated with
phones that do not support this feature. Only Cisco Unified IP Phones that support
the hold reversion feature display the hold reversion timer settings in the Directory
Number Configuration window.
When Hold Reversion is configured for the system, the phone must support the
feature or the feature does not activate.
See Cisco Unified IP Phone administration guides for Cisco Unified IP Phone
models that support hold reversion and this version of Unified Communications
Manager for any phone restrictions with hold reversion.
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CHAPTER 35
Accessing Hunt Groups
• Hunt Group Overview, on page 465
• Hunt Group Prerequisites, on page 466
• Hunt Group Configuration Task Flow, on page 466
• Hunt Group Interactions, on page 471
• Hunt Group Restrictions, on page 472
Hunt Group Overview

A Hunt Group is a group of lines that are organized hierarchically, so that if the first number in the hunt group
list is busy, the system dials the second number. If the second number is busy, the system dials the next number,
and so on.
The phone users can log in to or log out of the hunt groups by using the HLog softkey or the Hunt Group line
button on the IP phone. The phone provides a visual status of the login state, so that the user can determine
whether they are logged in to one or more of their line groups.
The Hunt Group feature provides the following functions:
• The HLog softkey on the IP phone allows the user to toggle between login and logout of phone.
• A hunt group allows a caller to automatically find an available line from amongst a group of extensions.
• The Hunt Group Log Off feature allows phone users to prevent their phones from receiving incoming
calls that get routed to directory numbers. Regardless of the phone status, the phone rings normally for
incoming calls that are not calls to one or more line groups associated with the phone.
Note The directory numbers (DNs) belong to line groups that are associated with the
phone.
• System administrators can log in or log out the users from the phones that are automatically logged into
hunt groups.
• The HLog softkey allows a phone user to log a phone out of all line groups to which the phone directory
numbers belong.
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Hunt Group Prerequisites
• From Cisco Unified Communications Manager Release 9.0 onward, the Hunt Group Log Off feature
enables the use of mobile device as a desk phone. When you use the Hlog softkey through your mobile
client, you no longer receive calls that are placed to the hunt pilot.
Hunt Group Prerequisites

• The phones must be running Skinny Client Control Protocol (SCCP) or Session Initiation Protocol (SIP).
• The phone ringtone file must be located in the TFTP directory (/usr/local/cm/tftp).
Hunt Group Configuration Task Flow

Before you begin
• Review Hunt Group Prerequisites, on page 466
Procedure

Step 1 Configure a Softkey Template for Hunt Group, Configure a softkey template for the HLog
on page 467 softkey.
Configuration to apply configuration options to
phones. This is the most commonly used
Configuration if you need to assign a softkey
Step 4 Configure Phones for Hunt Group, on page 470 Configure phones to automatically log in to or
log out of hunt groups and hunt lists.
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Configure a Softkey Template for Hunt Group
Configure a Softkey Template for Hunt Group

The HLog softkey appears on the phone when the phone is in the following call states:
• Connected
• On Hook
• Off Hook
Note You must create a new softkey template to configure the HLog softkey. You cannot configure the HLog
softkey in a standard softkey template.
Use this procedure to make the HLog softkey available:
Procedure
a) Click Add New.
d) Click Save.
template.
click Go.
to display.
Step 9 Click Save.
devices.
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What to do next
• Add a Softkey Template to a Common Device Configuration, on page 468
• Associate a Softkey Template with a Phone, on page 469

Before you begin

Configure a Softkey Template for Hunt Group, on page 467
Procedure

Procedure
a) Click Add New.
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c) Click Save.
to make available.
Step 5 Click Save.
restart them.
Before you begin

Procedure

Step 4 Click Save.

This procedure is optional. You can use this procedure as an alternative to associating the softkey template
with the Common Device Configuration. This procedure also works in conjunction with the Common Device
Configuration: use it when you need to assign a softkey template that overrides the assignment in the Common
Device Configuration or any other default softkey assignment.
Before you begin

Configure a Softkey Template for Hunt Group, on page 467
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Configure Phones for Hunt Group
Procedure

The Find and List Phones window appears.
Step 2 Choose the phone to which you want to add the softkey template.
Step 4 Click Save.
A dialog box appears with a message to press Reset to update the phone settings.
Configure Phones for Hunt Group

Use this procedure to configure phones to automatically log in to or log out of hunt groups and hunt lists.
Before you begin

Ensure the phone directory numbers belong to one or more hunt groups.
See the Administration Guide for Cisco Unified Communications Manager for information on hunt groups
and hunt lists.
Procedure

a) To modify the fields for an existing phone, enter search criteria and choose a phone from the resulting
list. The Phone Configuration window appears.
b) To add a new phone, click Add New.
The Add a New Phone window appears.
Step 3 In the Phone Configuration window, perform one of the following tasks:
a) To log out the phone from the hunt group, uncheck the Logged Into Hunt Group check box.
b) To log in the phone to the hunt group, ensure that the Logged Into Hunt Group check box is checked.
Note The Logged Into Hunt Group check box remains checked by default for all phones.
Step 4 Click Save.
Configure Hunt Group Service Parameter

The Hunt Group Logoff Notification service parameter provides the option to turn audible ringtones on or
off when calls that come in to a line group arrive at a phone that is currently logged out. This ringtone alerts
a logged-out user that there is an incoming call to a hunt list to which the line is a member, but the call will
not ring at the phone of that line group member because of the logged-out status.
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Hunt Group Interactions
To configure the Hunt Group Logoff Notification service parameter, perform the following steps.
Procedure
The Service Parameter Configuration window appears.
Step 4 In the Clusterwide Parameters (Device - Phone) section, configure values for the following Hunt Group
Logoff Notification service parameter:
Enter a name for the ringtone file that Cisco IP Phones play when a member of a line group (hunt group) has
logged out. The default value for this service parameter is None, which indicates no ringtone. You can enter
a maximum of 255 characters.
Step 5 Click Save.
The window refreshes, and Cisco Unified Communications Manager updates the service parameter with your
changes.
Hunt Group Interactions

Feature Interaction
Non-shared-line Directory If a phone is logged out of a line group and an extension on the phone is not
Number shared, the line group does not ring that directory number (DN) in the line group.
When the line group would normally offer the call to the DN, call processing
skips the DN and acts as if the DN does not belong to the line group.
Shared-line Directory Because the Log Out of Hunt Group feature is device-based, when a user logs a
Number phone out, the feature affects only the logged-out phone. Calls to a line group
that contains a shared-line directory number behave as follows:
• The DN does not ring if all phones that share that DN are logged out.
• The DN does ring if one or more phones that share the DN are logged in.
• The audible ring on a phone that is logged out is turned off by default. Cisco
Unified Communications Manager provides a system parameter that can be
set, so that a different ring tone plays when a call comes in to a logged-out
hunt group member.
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Hunt Group Restrictions
Hunt Group Restrictions

Multiple Line Groups When the user enables the Hunt Group Log Off feature by pressing the HLog
softkey, the phone gets logged out from all associated line groups. This is because
Hunt Group Log Off is a device-based feature. If a phone has DNs that belong
to multiple line groups, pressing the HLog softkey logs the phone out of all
associated line groups.
7940, 7960, and • When a phone that is running SIP (7906, 7911, 7941, 7961, ) is logged in
third-party SIP phones to hunt groups and Call Forward All is activated, the call gets presented to
the phone that is running SIP.
• When 7940 and 7960 phones that are running SIP are logged in to hunt
groups and Call Forward All is activated, the phones get skipped and the
next phone in the line group rings.
• 7940 and 7960 phones that are running SIP and third-party phones that are
running SIP can be logged in to or logged out of hunt groups by using the
Phone Configuration window, but no softkey support exists.
running SIP do not show “Logged out of hunt groups” on the status line.
running SIP do not play the Hunt Group Logoff Notification tone regardless
of whether the tone is configured.
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CHAPTER 36
Malicious Call Identification
• Malicious Call Identification Overview, on page 473
• Malicious Call Identification Prerequisites, on page 473
• Malicious Call Identification Configuration Task Flow, on page 474
• Malicious Call Identification Interactions, on page 480
• Malicious Call Identification Restrictions, on page 481
• Malicious Call ID Troubleshooting, on page 482
Malicious Call Identification Overview

You can configure the Malicious Call Identification (MCID) feature to track troublesome or threatening calls.
Users can report these calls by requesting that Cisco Unified Communications Manager identify and register
the source of the incoming call in the network.
When the MCID feature is configured, the following actions take place:
1. The user receives a threatening call and presses Malicious call (or enters the feature code *39 if using a
POTS phone that is connected to an SCCP gateway).
2. Cisco Unified Communications Manager sends the user a confirmation tone and a text message, if the
phone has a display, to acknowledge receiving the MCID notification.
3. Cisco Unified Communications Manager updates the call details record (CDR) for the call with an indication
that the call is registered as a malicious call.
4. Cisco Unified Communications Manager generates the alarm and local syslogs entry that contains the
event information.
5. Cisco Unified Communications Manager sends an MCID invocation through the facility message to the
connected network. The facility information element (IE) encodes the MCID invocation.
6. After receiving this notification, the PSTN or other connected network can take actions, such as providing
legal authorities with the call information.
Malicious Call Identification Prerequisites

• Gateways and connections that support MCID:
• PRI gateways that use the MGCP PRI backhaul interface for T1 (NI2) and E1 (ETSI) connections
• H.323 trunks and gateways
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Malicious Call Identification Configuration Task Flow
• IP Phones that support MCID
Malicious Call Identification Configuration Task Flow

Before you begin
• Review Malicious Call Identification Prerequisites, on page 473
Procedure

support the MCID feature.
Step 2 Set Malicious Call ID Service Parameter, on Enable Cisco Unified Communications Manager
page 475 to flag a call detail record (CDR) with the
MCID indicator.
Step 3 Configure Malicious Call ID Alarms, on page Configure alarms to ensure that alarm
475 information displays in the system logs.
Step 4 Configure a Softkey Template for Malicious Configure a softkey template with MCID.
Call Identification, on page 476
Note The Cisco Unified IP Phones 8900
and 9900 Series support MCID
with feature button only.
to phones. This is the most commonly used
Step 7 To Configure Malicious Call Identification Perform this step to add and configure the
Button, on page 478, complete the following MCID button to a phone.
subtasks:
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Set Malicious Call ID Service Parameter

• Configure Malicious Call ID Phone Button
Template, on page 479
• Associate a Button Template with a Phone
, on page 479
Set Malicious Call ID Service Parameter

To enable Unified Communications Manager to flag a CDR with the MCID indicator, you must enable the
CDR flag.
Before you begin

Configure Malicious Call ID Alarms, on page 475
Procedure
Step 2 From the Server drop-down list, choose the Unified Communications Manager server name.
The Service Parameter Configuration window displays.
Step 4 In the System area, set the CDR Enabled Flag field to True.
Step 5 Click Save.
Configure Malicious Call ID Alarms

In the Local Syslogs, you must set the alarm event level and activate alarms for MCID.
Before you begin

Set Malicious Call ID Service Parameter, on page 475
Procedure
Step 1 From Cisco Unified Serviceability, choose Alarm > Configuration.

The Alarm Configuration window displays.
Step 2 From the Server drop-down list, choose the Unified Communications Manager server and click Go.
Step 3 From the Service Group drop-down list, choose CM Services. The Alarm Configuration window updates
with configuration fields.
Step 5 Under Local Syslogs, in the Alarm Event Level drop-down list, choose Informational.
The Alarm Configuration window updates with configuration fields.
Step 6 Under Local Syslogs, check the Enable Alarm check box.
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Receiving Calls
Configure a Softkey Template for Malicious Call Identification
Step 7 If you want to enable the alarm for all nodes in the cluster, check the Apply to All Nodes check box.
Step 8 To turn on the informational alarm, click Update.
Configure a Softkey Template for Malicious Call Identification
Note Skinny Client Control Protocol (SCCP) IP phones use a softkey to invoke the MCID feature.
Before you begin

Procedure
a) Click Add New.
d) Click Save.
template.
click Go.
Step 6 In the Select a call state to configure field, choose Connected.
The list of Unselected Softkeys changes to display the available softkeys for this call state.
Step 7 In the Unselected Softkeys drop-down list, choose Toggle Malicious Call Trace (MCID).
Step 9 Click Save.

476
Receiving Calls

Before you begin

Configure a Softkey Template for Malicious Call Identification, on page 476
Procedure

Before you begin

Configure a Softkey Template for Malicious Call Identification, on page 476
Procedure
a) Click Add New.
c) Click Save.
to make available.
Step 5 Click Save.
477
Receiving Calls
restart them.
Before you begin

Procedure

Step 4 Click Save.

Procedure

Step 4 Click Save.
Configure Malicious Call Identification Button

The procedures in this section describe how to configure the Malicious Call Identification button.
Before you begin

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Receiving Calls
Configure Malicious Call ID Phone Button Template
Procedure

Step 1 Configure Malicious Call ID Phone Button Perform this step to assign Malicious Call
Template, on page 479. Identification button features to line or speed
dial keys.
Step 2 Associate a Button Template with a Phone , on Perform this step to configure the Malicious
page 479 Call Identification button for a phone.
Configure Malicious Call ID Phone Button Template
Before you begin

Procedure
step.
c) Click Save.
Step 6 Click Save.
devices.
Before you begin

Configure Malicious Call ID Phone Button Template, on page 479
479
Receiving Calls
Malicious Call Identification Interactions
Procedure

button.
Step 5 Click Save.
Malicious Call Identification Interactions

Table 46: Malicious Call Identification Interactions
Feature Interaction
Conference Calls When a user is connected to a conference, the user can use the MCID feature to
flag the call as a malicious call. Cisco Unified Communications Manager sends
the MCID indication to the user, generates the alarm, and updates the CDR.
However, Cisco Unified Communications Manager does not send an MCID
invoke message to the connected network that might be involved in the conference.
Extension Mobility Extension Mobility users can have the MCID softkey as part of their user device
profile and can use this feature when they are logged on to a phone.
Call Detail Records To track malicious calls by using CDR, you must set the CDR Enabled Flag to
True in the Cisco CallManager service parameter. When the MCID feature is
used during a call, the CDR for the call contains CallFlag=MALICIOUS in the
Comment field.
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Receiving Calls
Malicious Call Identification Restrictions
Feature Interaction
Alarms To record alarms for the MCID feature in the Local Syslogs, you must configure
alarms in Cisco Unified Serviceability. Under Local Syslogs, enable alarms for
the Informational alarm event level.
When the MCID feature is used during a call, the system logs an SDL trace and
a Cisco Unified Communications Manager trace in alarms. You can view the
Alarm Event Log by using Cisco Unified Serviceability. The traces provide the
following information:
• Date and time
• Type of event: Information
• Information: The Malicious Call Identification feature is invoked in Cisco
Unified Communications Manager
• Called Party Number
• Called Device Name
• Called Display Name
• Calling Party Number
• Calling Device Name
• Calling Display Name
• Application ID
• Cluster ID
• Node ID
For more information about alarms and traces, see the Cisco Unified Serviceability
Administration Guide at http://www.cisco.com/c/en/us/support/
products-maintenance-guides-list.html.
Cisco ATA 186 analog The Cisco ATA 186 analog phone ports support MCID by using the feature code
phone ports (*39).
Malicious Call Identification Restrictions

Table 47: Malicious Call Identification Restrictions
Feature Restriction
Malicious Call Identification Terminating (MCID-T) Cisco Unified Communications Manager supports
function only the malicious call identification originating
function (MCID-O). Cisco Unified Communications
Manager does not support the malicious call
identification terminating function (MCID-T). If Cisco
Unified Communications Manager receives a
notification from the network of a malicious call
identification, Cisco Unified Communications
Manager ignores the notification.
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Malicious Call ID Troubleshooting
Feature Restriction
Intercluster trunks MCID does not work across intercluster trunks

because Cisco Unified Communications Manager
does not support the MCID-T function.
Cisco MGCP FXS gateways Cisco MGCP FXS gateways do not support MCID.
No mechanism exists for accepting the hookflash and
collecting the feature code in MGCP.
QSIG trunks MCID does not work over QSIG trunks because
MCID is not a QSIG standard.
Cisco VG248 Analog Phone Gateway Cisco VG248 Analog Phone Gateway does not
support MCID.
SIP trunks MCID does not support SIP trunks.
Immediate Divert System does not support using MCID and Immediate
Divert features together.
Malicious Call ID Troubleshooting

To track and troubleshoot Malicious Call ID, you can use Cisco Unified Communications Manager SDL
traces and alarms. For information about setting traps and traces for MCID, see the Cisco Unified Serviceability
Administration Guide. For information about how to generate reports for MCID, see the Cisco Unified CDR
Analysis and Reporting Administration Guide.
482
CHAPTER 37
Call Transfer
• Call Transfer Overview, on page 483
• Call Transfer Configuration Task Flow, on page 484
• Call Transfer Interactions, on page 494
• Call Transfer Restrictions, on page 495
Call Transfer Overview

The transfer feature allows you to redirect a connected call from your phone to another number. After call
transfer, your call is disconnected and the transferred call is established as a new call connection.
Following are the different types of call transfers:
• Consult Transfer and Blind Transfer—In Consult Transfer, a transferring phone user can redirect the
caller to a different target address, after consulting with the target phone user that answers the call. That
is, the transferring phone user will stay on the call until the target phone user answers the call. In Blind
Transfer, the transferring phone user connects the caller to a destination line before the target of the
transfer answers the call.
Most phones use hard keys or softkeys for Transfer. Both Consult Transfer and Blind Transfer do not
require separate configuration. The difference between the two types of transfer depends on when the
transferring party presses the Transfer button a second time. For a consult transfer, the transferring party
presses the Transfer button after the target answers, while for a Blind Transfer, the transferring party
presses the Transfer button before the target answers.
For SCCP-initiated blind transfers, Cisco Unified Communications Manager provides call progress
indications in the form of ring-back to the transferred user.
• Transfer On-Hook—In this type of call transfer, the user presses the Transfer softkey, dials the number
to which the call will be transferred, and then presses the Transfer softkey again, or simply goes on-hook
to complete the transfer operation. You must set the Transfer On-Hook service parameter to True. This
service parameter determines whether a call transfer is completed as a result of the user going on-hook
after initiating a transfer operation.
Both Consult Transfer and Blind Transfer use the Transfer On-Hook option.
• Direct Transfer—This type of transfer allows a user to join two established calls (the two calls can
either be on hold or in the connected state) into one call and then drop the initiator from the transfer.
Direct Transfer does not initiate a consultation call and does not put the active call on hold. The user
uses the DirTrfr softkey to join any two established calls and remove the initiator.
483
Receiving Calls
Call Transfer Configuration Task Flow
Call Transfer Configuration Task Flow

Procedure

Step 1 Configure Consult and Blind Transfer, on page Transfer allows you to redirect a single call to
484 a new number with or without consulting the
transfer recipient. Perform this step to configure
Trnsfer as a softkey and/or button.
Step 2 Configure Transfer On-Hook, on page 489 (Optional) Transfer On-Hook is an option to
complete call transfers. Press Trnsfer, dial the
number to which the call should be transferred
to, and go on-hook to complete the transfer.
Perform this step to configure the service
parameter.
Step 3 Configure Direct Transfer, on page 489 (Optional) Direct Transfer allows you to transfer
two calls to each other (without you remaining
on the line). Perform this step to configure
DirTrfr as a softkey and/or button.
Configure Consult and Blind Transfer

Procedure

Step 1 Configure a Softkey Template for Transfer, on
page 484
Step 2 Configure Transfer Button, on page 487
Configure a Softkey Template for Transfer

Trnsfer softkey is used for consult and blind transfer of a call. The trnsfer sofkey has the following call states:
• connected
• on hold
Use this procedure to make the Trnsfer softkey available:
Procedure
484
Receiving Calls
Associate Transfer Softkey Template with a Common Device Configuration
a) Click Add New.
d) Click Save.
template.
click Go.
to display.
Step 9 Click Save.
devices.
What to do next
• Associate Transfer Softkey Template with a Common Device Configuration, on page 485
• Associate Transfer Softkey Template with a Phone, on page 487
Associate Transfer Softkey Template with a Common Device Configuration

485
Receiving Calls
Add Transfer Softkey Template to the Common Device Configuration
To use the alternative method, see Associate Transfer Softkey Template with a Phone, on page 487.
Before you begin

Configure a Softkey Template for Transfer, on page 484
Procedure

Step 1 Add Transfer Softkey Template to the Common Perform this step to add Trnsfer softkey
Device Configuration, on page 486 template to the Common Device Configuration.
Step 2 Associate a Common Device Configuration Perform this step to link the Trnsfer softkey
with a Phone, on page 487 Common Device Configuration to a phone.
What to do next
Configure Transfer Button, on page 487
Add Transfer Softkey Template to the Common Device Configuration
Before you begin

Procedure
a) Click Add New.
c) Click Save.
to make available.
Step 5 Click Save.
486
Receiving Calls
restart them.
Before you begin

Add Transfer Softkey Template to the Common Device Configuration, on page 486
Procedure

Step 4 Click Save.
Associate Transfer Softkey Template with a Phone

Before you begin

Procedure

Step 4 Click Save.
Configure Transfer Button

The procedures in this section describe how to configure the Transfer button.
487
Receiving Calls
Configure a Phone Button Template for Transfer
Procedure

Step 1 Configure a Phone Button Template for Perform this step to assign Transfer button
Transfer, on page 488 features to line or speed dial keys.
Step 2 Associate Transfer Button Template with a Perform this step to configure the Transfer
Phone, on page 488 button for a phone.
Configure a Phone Button Template for Transfer

Optional. Follow this procedure when you want to assign features to line or speed dial keys.
Procedure
step.
c) Click Save.
Step 6 Click Save.
devices.
Associate Transfer Button Template with a Phone
Procedure

button.
Step 5 Click Save.
488
Receiving Calls
Configure Transfer On-Hook
Configure Transfer On-Hook

Before you begin
Configure Consult and Blind Transfer, on page 484
Procedure
The Service Parameter Configuration window is displayed.
Step 2 From the Server drop-down list, choose the server on which you want to configure the parameter.
Step 3 From the Service drop-down list, choose the Cisco CallManager (Active) service.
Step 4 In the Clusterwide Parameters (Device - Phone), choose True for the Transfer On-Hook Enabled service
parameter.
Step 5 Click Save.
Configure Direct Transfer

Procedure

Step 1 Configure a Softkey Template for Direct Perform this step to add Direct Transfer softkey
Transfer, on page 489 to template and configure the softkey using the
Common Device Configuration or phone.
Step 2 Configure Direct Transfer Button, on page 492 Perform this step to add and configure the
Direct Transfer button to a phone.
Configure a Softkey Template for Direct Transfer

Direct Transfer softkey has the following call states:
• Connected
• On hold
Use this procedure to make the Direct Transfer softkey available:
489
Receiving Calls
Associate Direct Transfer Softkey Template with a Common Device Configuration
Procedure
a) Click Add New.
d) Click Save.
template.
click Go.
to display.
Step 9 Click Save.
devices.
What to do next
• Associate Direct Transfer Softkey Template with a Common Device Configuration, on page 490
• Associate Direct Transfer Softkey Template with a Phone, on page 492
Associate Direct Transfer Softkey Template with a Common Device Configuration

490
Receiving Calls
Add Direct Transfer Softkey Template to the Common Device Configuration
To use the alternative method, see Associate Direct Transfer Softkey Template with a Phone, on page 492
Before you begin

Configure a Softkey Template for Direct Transfer, on page 489
Procedure

Step 1 Add Direct Transfer Softkey Template to the Perform this step to add Direct Transfer softkey
Common Device Configuration, on page 491 template to the Common Device Configuration.
Step 2 Associate a Common Device Configuration Perform this step to add Direct Transfer softkey
with a Phone, on page 492 template to the Common Device Configuration.
Add Direct Transfer Softkey Template to the Common Device Configuration
Procedure
a) Click Add New.
c) Click Save.
to make available.
Step 5 Click Save.
restart them.
491
Receiving Calls
Before you begin

Add Direct Transfer Softkey Template to the Common Device Configuration, on page 491
Procedure

Step 4 Click Save.
Associate Direct Transfer Softkey Template with a Phone

Before you begin

Configure a Softkey Template for Direct Transfer, on page 489
Procedure

Step 4 Click Save.
Configure Direct Transfer Button

The procedures in this section describe how to configure the Direct Transfer button.
Procedure

Step 1 Configure Phone Button Template for Direct Perform this step to assign Direct Transfer
Transfer, on page 493 button features to line or speed dial keys.
492
Receiving Calls
Configure Phone Button Template for Direct Transfer

Step 2 Associate Direct Transfer Button Template with Perform this step to configure the Direct
a Phone, on page 493 Transfer button for a phone.
Configure Phone Button Template for Direct Transfer

Optional. Follow this procedure when you want to assign features to line or speed dial keys.
Procedure
step.
c) Click Save.
Step 6 Click Save.
devices.
Associate Direct Transfer Button Template with a Phone
Before you begin

Configure Phone Button Template for Direct Transfer, on page 493
Procedure

button.
Step 5 Click Save.
493
Receiving Calls
Call Transfer Interactions
Call Transfer Interactions

Feature Interaction
Logical Partitioning The logical partitioning policy check is performed between the
geolocation identifier of the device that is acting as a transferred party
and the geolocation identifier of the device that is acting as a transferred
destination.
Logical partitioning handling takes place in the following circumstances:
• When a phone user uses Transfer softkey to transfer the call, the
second press of the softkey invokes and processes the Call Transfer
feature.
• When other transfer mechanisms, such as Direct Transfer, On-Hook
Transfer, Hook Flash Transfer, and CTI-application-initiated
Transfer results in invoking the Call Transfer feature.
• When the transferred and the transferred destination specifies a
PSTN participant.
• When Cisco Unified Communications Manager uses the geolocation
identifier information that associates with the transferred and
transferred destination device to perform logical partitioning policy
checking.
• Before splitting of the primary and secondary calls, and before
joining.
Logical partitioning handles a denied call as follows:

• Sends External Transfer Restricted message to the VoIP phone.
• Normal Transfer—For a phone that is running SCCP, the primary
call remains on hold, and the consultation call remains active. For
a phone that is running SIP, both primary and consultation calls
remain on hold and must be resumed manually after the failure.
• On-Hook, Hook-Flash and Analog-Phone-Initiated Transfer—Both
the primary and secondary calls are cleared by using the cause
code=63 “Service or option not available” with a reorder tone from
Cisco Unified Communications Manager.
• The Number of Transfer Failures perfmon counter is incremented.
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Receiving Calls
Call Transfer Restrictions
Feature Interaction
Multilevel Precedence and When a switch initiates a call transfer between two segments that have
Preemption (MLPP) the same precedence level, the segments maintain the precedence level
upon transfer. When a call transfer is made between call segments that
are at different precedence levels, the switch that initiates the transfer
marks the connection at the segment that has the higher precedence level.
Cisco Unified Communications Manager supports this requirement by
upgrading the precedence level of a call leg that is involved in a Call
Transfer operation. For example, party A calls party B with Priority
precedence level. Party B then initiates a transfer to party C and dials
the Flash precedence digits when dialing. When the transfer is complete,
the precedence level of party A gets upgraded from Priority to Flash.
The Call Transfer feature is enabled automatically when MLPP is
enabled, and the phones support the Transfer softkey.
Note The precedence level upgrade does not work over a trunk
device such as an intercluster trunk (ICT) or a PRI trunk.

Feature Restriction
Logical Partitioning Logical partitioning handling does not take place when both the transferred
and the transferred destination devices are VoIP phones.
Logical partitioning handling does not take place when geolocation or a
geolocation filter is not associated with any device.
External Call Transfer Restrictions To restrict transfer for external call scenarios, see the “External Call
Transfer Restrictions” chapter.
Hunt Pilot If a call transfer to a hunt pilot is initiated when an announcement is in

progress, the call is redirected only after the announcement is complete.
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Receiving Calls
496
CHAPTER 38
External Call Transfer Restrictions
• External Call Transfer Restrictions Overview, on page 497
• Configure External Call Transfer Restrictions Task Flow, on page 498
• External Call Transfer Restrictions Interactions, on page 502
• External Call Transfer Restrictions Restrictions, on page 502
External Call Transfer Restrictions Overview

External Call Transfer Restrictions is a feature that you can use to configure gateways, trunks, and route
patterns as OnNet (internal) or OffNet (external) devices at the system level. By setting the devices as OffNet,
you can restrict the transferring of an external call to an external device and thus help prevent toll fraud.
If you try to transfer a call on an OffNet gateway or trunk when the service parameter Block OffNet to OffNet
Transfer is set to True, a message displays on the user phone to indicate that the call cannot be transferred.
This chapter uses the following terms:
Term Description
OnNet Device A device that is configured as OnNet and considered
to be internal to the network.
OffNet Device A device that is considered as OffNet and, when

routed, is considered to be external to the network.
Network Location The location of the device, which is considered as

OnNet or OffNet, with respect to the network.
Originating End The device that gets transferred. The system considers
this device as OnNet or OffNet.
Terminating End The device that receives the transferred call. The
system considers this device as OnNet or OffNet.
Incoming Call A call for which only gateways and trunks call
classification settings get used to classify it as OnNet
or OffNet. Route Pattern call classification settings
do not apply.
497
Receiving Calls
Configure External Call Transfer Restrictions Task Flow
Term Description
Outgoing Call A call for which the call classification setting of the
trunk, gateway, and route pattern gets considered. The
Allow Device Override setting on the route pattern
determines whether the trunk or gateway call
classification setting gets used instead of the route
pattern call classification setting.
Configure External Call Transfer Restrictions Task Flow

Procedure

Step 1 Configure the Service Parameter for Call Block external calls from being transferred to
Transfer Restrictions, on page 498 another external device or number.
Step 2 To configure incoming calls perform the Configure gateways and trunks as OnNet
following procedures: (internal) or OffNet (external) by using Gateway
Configuration or Trunk Configuration or by
• Configure the Clusterwide Service
setting a clusterwide service parameter.
Parameter, on page 499
• Configure Gateways for Call Transfer
Restrictions, on page 500
• Configure Trunks for Call Transfer
Restrictions, on page 500
Step 3 Configure Outgoing Calls, on page 501 Configure transfer capabilities with route pattern
configuration.
Configure the Service Parameter for Call Transfer Restrictions

To block external calls from being transferred to another external device or number:
Procedure
Step 1 From the Cisco Unified CM Administration user interface choose System > Service Parameters.
Step 2 On the Service Parameter Configuration window choose the Cisco Unified CM server you want to configure
from the Server drop-down list.
Step 3 Choose Cisco CallManager (Active) from the Service drop-down list.
Step 4 Choose True from the Block OffNet to OffNet Transfer drop-down list. The default value specifies False.
Step 5 Click Save.
498
Receiving Calls
Configure Incoming Calls Task Flow
Configure Incoming Calls Task Flow

Procedure

Step 1 (Optional) Configure the Clusterwide Service Configure all gateways or trunks in the Cisco
Parameter, on page 499 Unified Communications Manager cluster to
be OffNet (external) or OnNet (internal).
Step 2 Configure Gateways for Call Transfer Configure gateways as OnNet (internal) or
Restrictions, on page 500 OffNet (external) by using Gateway
Configuration. When the feature is used in
conjunction with the clusterwide service
parameter Block OffNet to OffNet Transfer, the
configuration determines whether calls can
transfer over a gateway.
You can configure the following devices as
internal and external to Cisco Unified
Communications Manager:
• H.323 gateway
• MGCP FXO trunk
• MGCP T1/E1 trunk
Step 3 Configure Trunks for Call Transfer Restrictions, Configure trunks as OnNet (internal) or OffNet
on page 500 (external) by using Trunk Configuration. When
the feature is used in conjunction with the
clusterwide service parameter Block OffNet to
OffNet Transfer, the configuration determines
whether calls can transfer over a trunk.
You can configure the following devices as
internal and external to Cisco Unified
Communications Manager:
• Intercluster trunk
• SIP trunk
Configure the Clusterwide Service Parameter

To configure all gateways or trunks in the Cisco Unified Communications Manager cluster to be OffNet
(external) or OnNet (internal), perform the following steps:
Before you begin

Configure the Service Parameter for Call Transfer Restrictions, on page 498
499
Receiving Calls
Configure Gateways for Call Transfer Restrictions
Procedure
Step 1 From the Cisco Unified CM Administration user interface choose System > Service Parameters.
Step 2 On the Service Parameter Configuration window choose the Cisco Unified CM server you want to configure
from the Server drop-down list.
Step 3 Choose Cisco CallManager (Active) from the Service drop-down list.
Step 4 Choose either OffNet or OnNet (the default specifies OffNet) from the Call Classification drop-down list.
Configure Gateways for Call Transfer Restrictions

To configure the gateway as OffNet, OnNet, or Use System Default, perform the following procedure. The
system considers calls that come to the network through that gateway as OffNet or OnNet, respectively.
Before you begin

Configure the Clusterwide Service Parameter, on page 499
Procedure
Step 1 From Cisco Unified CM Administration, choose Device > Gateway.

The Find and List Gateways window displays.
Step 2 To list the configured gateways, click Find.

The gateways that are configured in Unified Communications Manager display.
Step 3 Choose the gateway that you want to configure as OffNet or OnNet.
Step 4 In the Call Classification field choose OffNet or OnNet. If you have enabled clusterwide restrictions an all
gateways, configure each gateway to Use System Default (this reads the setting in the Call Classification
service parameter and uses that setting for the gateway).
Step 5 Click Save.
Configure Trunks for Call Transfer Restrictions

To configure the trunk as OffNet, OnNet, or Use System Default, perform the following procedure. The system
considers calls that come to the network through that trunk as OffNet or OnNet, respectively.
Before you begin

Configure Gateways for Call Transfer Restrictions, on page 500
Procedure
500
Receiving Calls
Configure Outgoing Calls
The Find and List Trunk window displays.
Step 2 To list the configured trunks, click Find.

The trunks that are configured in Unified Communications Manager display.
Step 3 Choose the trunk that you want to configure as OffNet or OnNet.
Step 4 From the Call Classification drop-down list, choose one of the following fields:
• OffNet - When you choose this field, this identifies the gateway as an external gateway. When a call
comes in from a gateway that is configured as OffNet, the system sends the outside ring to the destination
device.
• OnNet - When you choose this field, this identifies the gateway as an internal gateway. When a call
comes in from a gateway that is configured as OnNet, the system sends the inside ring to the destination
device.
• Use System Default - When you choose this field, this uses the Unified Communications Manager
clusterwide service parameter Call Classification.
Note If you have enabled clusterwide restrictions an all trunks, configure each trunk to Use System
Default (this reads the setting in the Call Classification service parameter and uses that setting
for the trunk)
Step 5 Click Save.
Configure Outgoing Calls

To classify a call as OnNet or OffNet, administrators can set the Call Classification field to OnNet or OffNet,
respectively, on the Route Pattern Configuration window. Administrators can override the route pattern
setting and use the trunk or gateway setting by checking the Allow Device Override check box on the Route
Pattern Configuration window.
Before you begin

Configure Trunks for Call Transfer Restrictions, on page 500
Procedure
Step 1 From Cisco Unified CM Administration, choose Call Routing > Route/Hunt > Route Pattern and click
Find to list all route patterns.
Step 2 Choose the route pattern you want to configure, or click Add New.
Step 3 In the Route Pattern Configuration window, use the following fields to configure transfer capabilities with
route pattern configuration:
a) Call Classification—Use this drop-down list to classify the call that uses this route Pattern as OffNet or
OnNet.
b) Provide Outside Dial Tone—If Call Classification is set to OffNet, this check box gets checked.
501
Receiving Calls
External Call Transfer Restrictions Interactions
c) Allow Device Override—When this check box is checked, the system uses the Call Classification setting
of the trunk or gateway that is associated with the route pattern instead of the Call Classification setting
on the Route Pattern Configuration window.
Step 4 Click Save.
External Call Transfer Restrictions Interactions

Feature Interaction
Drop Conference The Drop Conference feature determines whether an

existing ad hoc conference should be dropped by
checking whether the conference parties are
configured as OffNet or OnNet. You use the service
parameter Drop Ad Hoc Conference and choose the
option When No OnNet Parties Remain in the
Conference to configure the feature. You determine
OnNet status for each party by checking the device
or route pattern that the party is using. For more
information, see topics related to Ad Hoc Conference
linking in the “Ad Hoc Conferencing” chapter.
Bulk Administration Bulk Administration inserts gateway configuration

(OffNet or OnNet) on the Gateway Template. For
more information, see the Cisco Unified
Communications Manager Bulk Administration Guide.
Dialed Number Analyzer (DNA) When used to perform digit analysis on a gateway,
DNA displays the Call Classification that is
configured for the gateway and the route pattern. For
more information, see the Cisco Unified
Communications Manager Dialed Number Analyzer
Guide.
External Call Transfer Restrictions Restrictions

FXS Gateways FXS gateways such as Cisco Catalyst 6000 24 Port

do not have a Call Classification field on the Gateway
Configuration window; therefore, the system always
considers them as OnNet.
Cisco VG248 Gateway The system does not support the Cisco VG248
Gateway which does not have a Call Classification
field.
502
Receiving Calls
FXS Ports Cisco Unified Communications Manager considers

all Cisco Unified IP Phones and FXS ports as OnNet
(internal) that cannot be configured as OffNet
(external).
503
Receiving Calls
504
PA R T XI
Presence and Privacy Features
• Barge , on page 507
• BLF Presence , on page 519
• Call Display Restrictions , on page 533
• Do Not Disturb , on page 545
• Privacy , on page 557
• Private Line Automatic Ringdown , on page 563
• Secure Tone , on page 569
CHAPTER 39
Barge
• Barge Overview, on page 507
• Barge Configuration Task Flow, on page 509
• Barge Interactions, on page 516
• Barge Restrictions, on page 516
• Barge Troubleshooting, on page 517
Barge Overview
Barge allows a user to be added to a remotely active call that is on a shared line. Remotely active calls for a
line are the active (connected) calls that are made to or from another device that shares a directory number
with the line.
If you configure party entrance tone, a tone plays on the phone when a basic call changes to a barged call or
cbarged call. In addition, a different tone plays when a party leaves the multiparty call.
Phones support Barge in the following conference modes:
• Built-in conference bridge at the phone that is barged—This mode uses the Barge softkey. Most Cisco
Unified IP Phones include the built-in conference bridge capability.
• Shared conference bridge—This mode uses the cBarge softkey.
By pressing the Barge or cBarge softkey in the remote-in-use call state, the user is added to the call with all
parties, and all parties receive a barge beep tone (if configured). If Barge fails, the original call remains active.
If no conference bridge is available (built-in or shared), the barge request gets rejected, and a message displays
on the Barge initiator device. When network or Unified Communications Manager failure occurs, the Barge
call is preserved.
Note To display the softkey option for both Barge and cBarge, disable the Privacy option in Unified Communications
Manager user interface for those devices that have shared line appearances.
For a list of Cisco Unified IP Phones that support Barge, log in to Cisco Unified Reporting and run the Unified
CM Phone Feature List report. Make sure to select Built In Bridge as the feature. For details, see Generate a
Phone Feature List, on page 1.
507
Built-In Conference
Single-Button Barge and Single-Button cBarge

The Single-Button Barge and Single-Button cBarge features allow a user to press the shared-line button of
the remotely active call, to be added to the call. All parties receive a barge beep tone (if configured). If barge
fails, the original call remains active.
Phones support Single-Button Barge and Single-Button cBarge in two conference modes:
• Built-in conference bridge at the phone that is barged—This mode uses the Single-Button Barge feature.
• Shared conference bridge—This mode uses the Single-Button cBarge feature.
By pressing the shared-line button of the remote-in-use call, the user is added to the call with all parties, and
all parties receive a barge beep tone (if configured). If barge fails, the original call remains active. If no
conference bridge is available (built-in or shared), the barge request gets rejected, and a message is displayed
at the Barge initiator device.
Built-In Conference
When the user presses the Barge softkey or a shared-line button, a Barge call is set up by using the built-in
conference bridge, if available. A built-in conference bridge is advantageous because neither a media
interruption nor display changes to the original call occur when the Barge is being set up.
Shared Conference
When the user presses the cBarge softkey, or a shared-line button, a barge call is set up by using the shared
conference bridge, if available. The original call is split and then joined at the conference bridge, which causes
a brief media interruption. The call information for all parties changes to “Barge”. The barged call becomes
a conference call with the barge target device as the conference controller. It can add more parties to the
conference or can drop any party. When any party releases the call, the remaining two parties experience a
brief interruption and then get reconnected as a point-to-point call, which releases the shared conference
resource.
Built-In and Shared Conference Differences

This table describes the differences between barge with built-in conference bridge and shared conference.
Feature Barge with Built-In Conference Barge with Shared Conference
The standard softkey template Yes No

includes the Barge/cBarge softkey.
Note If the single button
Barge/cBarge feature
is enabled, the
softkey is not used.
A media break occurs during barge No Yes

setup.
If configured, a user receives a Yes Yes

barge setup tone.
508
Barge Configuration Task Flow
Feature Barge with Built-In Conference Barge with Shared Conference
Text displays at the barge initiator To barge XXX To Conference

phone.
Text displays at the target phone. To/From Other To Conference
Text displays at the other phones. To/From Target To Conference
Bridge supports a second barge No Yes

setup to an already barged call.
Initiator releases the call. No media interruption occurs for Media break occurs to release the
the two original parties. shared conference bridge when only
two parties remain and to reconnect
the remaining parties as a
point-to-point call.
Target releases the call. Media break occurs to reconnect Media break occurs to release the
initiator with the other party as a shared conference bridge when only
point-to-point call. two parties remain and to reconnect
Other party releases the call. All three parties get released. Media break occurs to release the
shared conference bridge when only
two parties remain and to reconnect
Target puts call on hold and Initiator gets released. Initiator and the other party remain
performs Direct Transfer, Join, or connected.
Call Park.
Barge Configuration Task Flow

Procedure

Step 1 Configure Softkey Template for Built-In Add the Barge softkey to a softkey template.
Conferencing, on page 510 Follow this procedure when you are configuring
barge for built-in conference bridges.
Step 2 Configure Softkey Template for Shared Add the cBarge softkey to a softkey template.
Conferencing, on page 511 Follow this procedure when you are configuring
barge for shared conference bridges.
Step 3 To Associate a Softkey Template with Common Optional. To make the softkey template
Device Configuration, on page 512, complete available to phones, you must complete either
the following subtasks: this step or the following step. Follow this step
509
Configure Softkey Template for Built-In Conferencing

• Add a Softkey Template to Common if your system uses a Common Device
Device Configuration, on page 513 Configuration to apply configuration options to
• Associate Common Device Configuration phones. This is the most commonly used
with Phone, on page 514 method for making a softkey template available
to phones.
Step 4 Associate Softkey Template with Phone, on Optional. Use this procedure either as an
default softkey.
Step 5 Configure Barge for Built-In Conferencing, on Configure barge for built-in conference bridges.
page 514
Step 6 Configure Barge for Shared Conferencing, on Configure barge for shared conference bridges.
page 515
Step 7 Associate User with Device, on page 60 Associate users with devices.
Configure Softkey Template for Built-In Conferencing

Configure a softkey template for Barge and assign the Barge softkey to that template. You can configure the
Barge softkey in the Remote In Use call state.
Procedure
a) Click Add New.
d) Click Save.
template.
510
Configure Softkey Template for Shared Conferencing
click Go.
to display.
Step 9 Click Save.
devices.
What to do next
• Add a Softkey Template to Common Device Configuration, on page 513
• Associate Common Device Configuration with Phone, on page 514
Configure Softkey Template for Shared Conferencing

Configure a softkey template for shared conferencing and assign the cBarge softkey to that template. You can
configure the cBarge softkey in the Remote In Use call state.
Procedure
a) Click Add New.
d) Click Save.
template.
511
Associate Softkey Template with Phone
click Go.
to display.
Step 9 Click Save.
devices.
Associate Softkey Template with Phone

Procedure

The Find and List Phones window is displayed.
Step 2 Find the phone to which you want to add the softkey template.
• From the Common Device Configuration drop-down list, choose the common device configuration
that contains the required softkey template.
• In the Softkey Template drop-down list, choose the softkey template that contains the Barge or cBarge
softkey.
Step 4 Click Save.

Associate a Softkey Template with Common Device Configuration

512
To use the alternative method, see Associate Softkey Template with Phone, on page 512.
Procedure
Step 1 Add a Softkey Template to Common Device Configuration, on page 366

Before you begin

Perform one or both of the following as needed:
• Configure Softkey Template for Built-In Conferencing, on page 510
• Configure Softkey Template for Shared Conferencing, on page 511
Procedure
a) Click Add New.
c) Click Save.
to make available.
Step 5 Click Save.
restart them.
513
Associate Common Device Configuration with Phone
Before you begin

Perform one or both of the following as needed:
• Configure Softkey Template for Built-In Conferencing, on page 510
• Configure Softkey Template for Shared Conferencing, on page 511
Procedure

Step 4 Click Save.
What to do next
Perform one or both of the following:
• Configure Barge for Built-In Conferencing, on page 514
• Configure Barge for Shared Conferencing, on page 515
Configure Barge for Built-In Conferencing

Most Cisco Unified IP Phones include the built-in conference bridge capability; that is, these Cisco IP Phones
have an internal DSP that acts as a small conference bridge to support the barge feature. It can support only
a maximum of three parties that include the phone itself. Starting from firmware version 11.x, Cisco IP Phone
8800 Series have the capability to daisy chain the built-in bridge (BIB) feature.
Procedure
Step 1 From Cisco Unified CM Administration, choose System > Service Parameters and set the Built In Bridge
Enable clusterwide service parameter to On.
Note If this parameter is set to Off, configure barge for each phone by setting the Built in Bridge field
in the Phone Configuration window.
Step 2 Set the Party Entrance Tone clusterwide service parameter to True (required for tones) or configure the
Party Entrance Tone field in the Directory Number Configuration window.
Step 3 Set the Single Button Barge/CBarge Policy to Barge.
Note If this parameter is set to Off, configure single-button barge for each phone by setting the Single
Button Barge field in the Phone Configuration window.
514
Configure Barge for Shared Conferencing
Step 4 Set the Allow Barge When Ringing service parameter to True.
Step 5 Click Save.
Configure Barge for Shared Conferencing

Cisco recommends that you do not configure Barge for shared conferencing (cBarge) for a user who has Barge
configured. Choose only one barge method for each user.
Procedure
Step 1 From Cisco Unified CM Administration, choose System > Service Parameters and set the Built In Bridge
Enable clusterwide service parameter to On.
Note If this parameter is set to Off, configure cBarge for each phone by setting the Built in Bridge
field in the Phone Configuration window.
Step 2 Set the Party Entrance Tone clusterwide service parameter to True (required for tones) or configure the
Party Entrance Tone field in the Directory Number Configuration window.
Step 3 Set the Single Button Barge/CBarge Policy to cBarge.
Note If this parameter is set to Off, configure Single-button cBarge for each phone by setting the Single
Button cBarge field in the Phone Configuration window.
Step 4 Set the Allow Barge When Ringing service parameter to True.
Step 5 Click Save.
Associate User with Device

Before you begin
Perform one or both of the following:
• Configure Barge for Built-In Conferencing, on page 514
• Configure Barge for Shared Conferencing, on page 515
Procedure
Step 2 Specify the appropriate filters in the Find User Where field to and then click Find to retrieve a list of users.
The End User Configuration window appears.
Step 4 Locate the Device Information section.
Step 5 Click Device Association.
515
Barge Interactions
The User Device Association window appears.

Step 6 Find and select the CTI remote device.
Step 7 To complete the association, click Save Selected/Changes.
Step 8 From Related Links drop-down list, choose Back to User, and then click Go.
Barge Interactions
Feature Interaction
cBarge Cisco recommends that you assign either the Barge or cBarge softkey to a softkey
template. By having only one of these softkeys for each device, you can prevent
confusion for users and avoid potential performance issues.
Note You can enable Single-Button Barge or Single-Button cBarge for
a device, but not both.
Call Park When the target parks the call, the barge initiator gets released (if using the built-in
bridge), or the barge initiator and the other party remain connected (if using the
shared conference).
Join When the target joins the call with another call, the barge initiator gets released
(if using the built-in bridge), or the barge initiator and the other party remain
connected (if using the shared conference).
Private Line Automatic A Barge, cBarge, or Single-Button Barge initiator can barge into a call through
Ringdown (PLAR) a shared line that is configured for Barge and Private Line Automatic Ringdown
(PLAR). The initiator can barge into the call if the barge target uses the
preconfigured number that is associated with the PLAR line while on the call.
Cisco Unified Communications Manager does not send the barge invocation to
the PLAR line before connecting the barge call, so the barge occurs regardless
of the state of the PLAR destination.
To make Barge, cBarge, or Single-Button Barge function with PLAR, you must
configure Barge, cBarge, or Single-Button Barge. In addition, you must configure
the PLAR destination, a directory number that is used specifically for PLAR.
Barge Restrictions
Additional callers The Barge initiator cannot conference in additional callers.
Computer Telephony CTI does not support Barge through APIs that TAPI and JTAPI applications
Interface (CTI) invoke. CTI generates events for Barge when it is invoked manually from an IP
phone by using the Barge or cBarge softkey.
516
Barge Troubleshooting
G.711 codec The original call requires G.711 codec. If G.711 is not available, use cBarge
instead.
Cisco Unified IP Phones You can assign a softkey template that contains the Barge softkey to any IP phone
that uses softkeys; however, some IP phones do not support the Barge feature.
Encryption If you configure encryption for Cisco Unified IP Phones 7960 and 7940, those
encrypted devices cannot accept a barge request when they are participating in
an encrypted call. When the call is encrypted, the barge attempt fails. A tone
plays on the phone to indicate that the Barge failed.
Maximum number of calls If the number of shared-line users in the conference is equal to or greater than
the configuration for the Maximum Number of Calls setting for the device from
which you are attempting to barge, the phone displays the message, Error:
Past Limit.
Barge Troubleshooting
No Conference Bridge Available
When the Barge softkey is pressed, the message No Conference Bridge Available is displayed on
the IP phone.
The Built In Bridge field in the Phone Configuration window for the target phone is not set properly.
To resolve the problem, perform the following steps:
1. From Cisco Unified CM Administration, choose Device > Phone and click Find the phone to find the
phone configuration of the phone that is having the problem.
2. Set the Built In Bridge field to On.
3. Click Update.
4. Reset the phone.
Error: Past Limit

The phone displays the message, Error: Past Limit.
The number of shared-line users in the conference is equal to or greater than the configuration for the Maximum
Number of Calls field for the device from which you are attempting to barge.
• Go to Service Parameter Configuration window and locate the Clusterwide Parameters (Feature -
Conference) section. Increase the value of Maximum Ad Hoc Conference parameter as required.
• Check the Maximum Number of Calls value for the shared lines on the device from which you are
attempting to barge and increase the value as required.
517
Error: Past Limit
518
CHAPTER 40
BLF Presence
• BLF Presence Overview, on page 519
• BLF Presence Prerequisites, on page 519
• BLF Presence Configuration Task Flow, on page 520
• BLF Presence Interactions, on page 531
• BLF Presence Restrictions, on page 531
BLF Presence Overview

The Busy Lamp Field (BLF) presence feature allows a user who is a watcher to monitor the real-time status
of another user at a directory number or Session Initiation Protocol (SIP) uniform resource identifier (URI)
from the device of the watcher.
A watcher can monitor the status of the user or BLF presence entity (also called presentity) by using the
following options:
• BLF and SpeedDial buttons
• Missed call, placed call, or received call lists in the directories window
• Shared directories, such as the corporate directory
Call lists and directories display the BLF status for existing entries. When you configure BLF and SpeedDial
buttons, the BLF presence entity appears as a speed dial on the device of the watcher.
To view the status of a BLF presence entity, watchers send BLF presence requests to Cisco Unified
Communications Manager. After administrators configure BLF presence features, real-time status icons appear
on the watcher device to indicate whether the BLF presence entity is on the phone, is not on the phone, the
status is unknown, and so on.
Extension mobility users can use BLF presence features on phones with extension mobility support.
BLF presence group authorization ensures that only authorized watchers can access the BLF presence status
for a destination. Because the administrator ensures that the watcher is authorized to monitor the destination
when a BLF or Speed Dial is configured, BLF presence group authorization does not apply to BLF or Speed
Dials.
BLF Presence Prerequisites

• Configure the phones that you want to use with the BLF presence feature.
519
BLF Presence Configuration Task Flow
• Configure the SIP trunks that you want to use with the BLF presence feature.
BLF Presence Configuration Task Flow

Before you begin
• Review BLF Presence Prerequisites, on page 519
Procedure

Step 1 Configure and synchronize cluster-wide Configure BLF options that apply to all devices
enterprise parameters for Busy Lamp Field and services in the same cluster. You can
(BLF). See Configure/Synchronize synchronize enterprise-parameter configuration
Cluster-Wide Enterprise Parameters for BLF, changes with the configured devices in the
on page 521. least-intrusive manner. For example, a reset or
restart may not be required on some affected
devices.
Step 2 Configure cluster-wide service parameters for Configure presence service parameters to
BLF. See Configure Cluster-Wide Service configure different services on selected servers
Parameters for BLF, on page 522. in Cisco Unified Communications Manager
Administration.
Step 3 Configure BLF presence groups. See Configure Configure BLF presence groups to control the
BLF Presence Groups, on page 522. destinations that watchers can monitor.
Step 4 To associate BLF presence group with devices Apply a BLF presence group to a directory
and users, perform the following subtasks: number, SIP trunk, phone that is running SIP,
phone that is running SCCP, application user
• Associate BLF presence groups with
(for application users that are sending presence
phones. See Associate BLF Presence
requests over the SIP trunk), or end user.
Groups with Phone, on page 524.
• Associate BLF presence groups with SIP
trunks. See Associate BLF Presence
Groups with SIP Trunk, on page 525.
• Associate BLF presence groups with an
end user. See Associate BLF Presence
Groups with End User, on page 526.
• Associate BLF presence groups with an
application user. See Associate BLF
Presence Groups with Application User,
on page 527.
Step 5 Accept BLF presence requests from external To enable application-level authorization for a
trunks and applications. See Accept BLF SIP trunk application in addition to trunk-level
Presence Requests from External Trunks and authorization.
Applications, on page 527.
520
Configure/Synchronize Cluster-Wide Enterprise Parameters for BLF

Step 6 Configure Calling Search Space. See Configure Apply a SUBSCRIBE Calling Search Space to
a Calling Search Space for Presence Requests, the SIP trunk, phone, or end user. The
on page 528. SUBSCRIBE Calling Search Space determines
how Cisco Unified Communications Manager
routes presence requests that come from the
trunk or the phone. Calling search spaces
determine the partitions that calling devices
search when they are attempting to complete a
call. If you do not select a different calling
search space for presence requests, the
SUBSCRIBE Calling Search Space selects the
default option, which is None.
Step 7 Configure a phone button template for BLF and Configure a phone button template for BLF and
SpeedDial buttons. See Configure a Phone SpeedDial buttons for a phone, or user device
Button Template for BLF and SpeedDial profile.
Buttons, on page 529.
Note If the template does not support
BLF and SpeedDials, the Add a
new BLF SD link appears in the
Unassigned Associated Items
pane.
Step 8 Associate button template with a device. See Use a button template with a configured device
Associate Button Template with a Device, on for the BLF presence.
page 530.
Step 9 Configure user device profile. See Configure Configure the user device profiles for BLF
User Device Profile, on page 530. presence.
Configure/Synchronize Cluster-Wide Enterprise Parameters for BLF

Use enterprise parameters for default configuration that apply to all devices and services in the same cluster.
A cluster consists of a set of Cisco Unified Communications Managers that share the same database. When
you install a new Cisco Unified Communications Manager, it uses the enterprise parameters to set the initial
values of its device defaults.
Procedure
Step 2 Configure the fields in the Enterprise Parameters Configuration window. For more information on the
Tip For details about an enterprise parameter, click the parameter name or the question mark that
appears in the Enterprise Parameter Configuration window.
Step 3 Click Save.

Step 4 (Optional) Click Apply Config to synchronize cluster-wide parameters.
521
Configure Cluster-Wide Service Parameters for BLF
The Apply Configuration Information dialog box appears.

Step 5 Click OK.
Configure Cluster-Wide Service Parameters for BLF

You can configure one or multiple services available in the Service Parameter Configuration window for
BLF.
Before you begin

Configure/Synchronize Cluster-Wide Enterprise Parameters for BLF, on page 521
Procedure
Step 2 From the Server drop-down list, choose the server where you want to configure the parameter.
Step 3 Configure the fields in the Service Parameters Configuration window. For more information on the fields
Tip For details about the service parameters, click the parameter name or the question mark that
appears in the Service Parameter Configuration window.
Step 4 Click Save.

Note The Default Inter-Presence Group Subscription parameter does not apply to BLF and SpeedDials.
Configure BLF Presence Groups

You can use BLF presence groups to control the destinations that watchers can monitor. To configure a BLF
presence group, create the group in Cisco Unified Communications Manager Administration and assign one
or more destinations and watchers to the same group.
When you add a new BLF presence group, Unified Communications Manager defines all group relationships
for the new group with the default cluster field as the initial permission fields. To apply different permissions,
configure new permissions between the new group and existing groups for each permission that you want to
change.
Note The system always allows BLF presence requests within the same BLF presence group.
To view the status of a presence entity, watchers send presence requests to Unified Communications Manager.
The system requires watchers to be authorized to initiate status requests for a presence entity with these
requirements:
522
BLF Presence Group Fields for BLF
• The watcher BLF presence group be authorized to obtain the status for the presence entity presence
group, whether inside or outside of the cluster.
• Unified CM must be authorized to accept BLF presence requests from an external presence server or
application.
Before you begin

Configure Cluster-Wide Service Parameters for BLF, on page 522
Procedure
Step 1 From Cisco Unified CM Administration, choose System > BLF Presence Group.
Step 2 Configure the fields in the BLF Presence Group Configuration window. See BLF Presence Group Fields
for BLF, on page 523 for details about the fields and their configuration options.
Note Use the Default Inter-Presence Group Subscription service parameter for the Cisco CallManager
service. It sets the clusterwide permissions parameter for BLF presence groups to allow
subscription or disallow subscription. This field enables administrators to set a system default
and configure BLF presence group relationships by using the default field for the cluster.
Step 3 Click Save.

Note The permissions that you configure for a BLF presence group appear in the BLF Presence Group
Relationship pane. Permissions that use the system default permission field for the group-to-group
relationship do not appear.
What to do next
Associate BLF presence group with devices and users by performing the following subtasks:
• Associate BLF Presence Groups with Phone, on page 524
• Associate BLF Presence Groups with SIP Trunk, on page 525
• Associate BLF Presence Groups with End User, on page 526
• Associate BLF Presence Groups with Application User, on page 527
BLF Presence Group Fields for BLF

Presence authorization works with BLF presence groups. The following table describes the BLF presence
group configuration fields.
Field Description
Name Enter the name of the BLF presence group that you want to configure. For
example, Executive_Group.
Description Enter a description for the BLF presence group that you are configuring.
523
BLF Presence Group Association with Devices and Users
Field Description
Modify Relationship to Select one or more BLF presence groups to configure the permission fields for
Other Presence Groups the named group to the selected groups.
Subscription Permission For the selected BLF presence groups, choose one of the following options
from the drop-down list:
• Use System Default—Set the permissions field to the Default
Inter-Presence Group Subscription clusterwide service parameter field
(Allow Subscription or Disallow Subscription).
• Allow Subscription—Allow members in the named group to view the
real-time status of members in the selected groups.
• Disallow Subscription—Block members in the named group from viewing
the real-time status of members in the selected groups.
The permissions that you configure appear in the BLF Presence Group
relationship pane when you click Save. All groups that use system default
permission field do not appear.
BLF Presence Group Association with Devices and Users

Perform the following procedures to apply a BLF presence group to the phone, SIP trunk, phone that is running
SIP, phone that is running SCCP, directory number, application user (for application users that are sending
presence requests over the SIP trunk), and end user.
Note The system allows presence requests between members in the same BLF presence group.
Associate BLF Presence Groups with Phone

You can use BLF presence for phones and trunks when the phones and trunks have permission to send and
receive presence requests.
Cisco Unified Communications Manager handles the BLF presence requests for Cisco Unified Communications
Manager users, whether inside or outside the cluster. For a Cisco Unified Communications Manager watcher
that sends a BLF presence request through the phone, Cisco Unified Communications Manager responds with
the BLF presence status if the phone and BLF presence entity are colocated
Before you begin

Configure BLF Presence Groups, on page 522
Procedure
Step 1 In the Cisco Unified CM Administration, choose Device > Phone, and click Add New.
Step 2 From the Phone Type drop-down list, select the type of phone that you want to associate BLF presence group
to.
524
Associate BLF Presence Groups with SIP Trunk
Step 3 Click Next.

Step 4 Configure the fields in the Phone Configuration window. See the online help for information about the fields
Note From the SUBSCRIBE Calling Search Space drop-down list, select a SUBSCRIBE calling
search space to use for presence requests for the phone. All calling search spaces that you configure
in Cisco Unified Communications Manager Administration appear in the SUBSCRIBE Calling
Search Space drop-down list. If you do not select a different calling search space for the end
user from the drop-down list, the value of this field applies the default value as None. To configure
a SUBSCRIBE calling search space specifically for this purpose, configure a calling search space
as you configure all calling search spaces.
Step 5 Click Save.
What to do next
Associate BLF Presence Groups with SIP Trunk

If digest authentication is not configured for the SIP trunk, you can configure the trunk to accept incoming
subscriptions, but application-level authorization cannot be initiated, and Unified CM accepts all incoming
requests before performing group authorization. When digest authentication is used with application-level
authorization, Unified CM also authenticates the credentials of the application that is sending the BLF presence
requests.
When there is a BLF presence request for a device that exists outside of the cluster, Unified Communications
Manager queries the external device through the SIP trunk. If the watcher has permission to monitor the
external device, the SIP trunk sends the BLF presence request to the external device, and returns BLF presence
status to the watcher.
Tip To use BLF presence group authorization with incoming presence requests on a SIP trunk, configure a presence
group for the trunk, such as External_Presence_Serv_Group1, and configure the appropriate permissions to
other groups inside the cluster.
If you configure both levels of authorization for SIP trunk presence requests, the BLF presence group for the
SIP trunk gets used only when no BLF presence group is identified in the incoming request for the application.
Before you begin

525
Associate BLF Presence Groups with End User
Procedure
Step 1 From Cisco Unified CM Administration, choose Device > Trunk, and click Add New.
Step 2 From the Trunk Type drop-down list, select the type of phone that you want to associate BLF presence group.
The value in the Device Protocol drop-down list populates automatically.
Step 3 Click Next.
Step 4 Configure the fields in the Trunk Configuration window. See the online help for information about the fields
Note To authorize the Unified CM system to accept incoming BLF presence requests from the SIP
trunk, check the Accept Presence Subscription check box in the SIP Trunk Security Profile
Configuration window. To block incoming presence requests on a SIP trunk, uncheck the check
box. When you allow SIP trunk BLF presence requests, Unified CM accepts requests from the
SIP user agent (SIP proxy server or external BLF presence server) that connects to the trunk.
Consider digest authentication as optional when Unified CM is configured to accept BLF presence
requests from a SIP trunk.
Step 5 Click Save.
What to do next
Associate BLF Presence Groups with End User

An administrator associates BLF presence groups with end user for user directories and call lists and to
configure extension mobility settings.
Before you begin

Procedure
Step 1 In the Cisco Unified CM Administration, choose User Management > End User, and click Add New.
The End User Configuration window appears.
Step 2 Configure the fields in the End User Configuration window. See the online help for information about the
Step 3 Click Save.
526
Associate BLF Presence Groups with Application User
What to do next
Associate BLF Presence Groups with Application User

An administrator associates BLF Presence groups with an application user for external applications. These
external applications send BLF presence requests that is SIP trunk or home on a proxy server which is connected
on SIP trunk. For example, Web Dial, Meeting Place, conference servers, and presence servers.
Before you begin

Procedure
Step 1 In the Cisco Unified CM Administration, choose User Management > Application User, and click Add
New.
The Application User Configuration window appears.
Step 2 Configure the fields in the Application User Configuration window. See the online help for information
Step 3 Click Save.
What to do next
Accept BLF Presence Requests from External Trunks and Applications

To allow BLF presence requests from outside the cluster, configure the system to accept BLF presence requests
from the external trunk or application. You can assign BLF presence groups to trunks and applications outside
the cluster to invoke BLF presence group authorization.
Before you begin

527
Configure a Calling Search Space for Presence Requests

Procedure
Step 1 From Cisco Unified CM Administration, choose Device > Trunk, and click Add New.
The Trunk Configuration window appears.
Step 2 To allow BLF presence requests from a SIP trunk, check the Accept Presence Subscription check box in
the SIP Trunk Security Profile Configuration window.
Step 3 To enable application-level authorization for a SIP trunk application in addition to trunk-level authorization,
check the following check boxes in the SIP Trunk Security Profile Configuration window:
• Enable Digest Authentication
• Enable Application Level Authorization
Note You cannot check Enable Application Level Authorization unless Enable Digest Authentication
is checked.
Step 4 Apply the profile to the trunk. Click Reset so that the changes to the trunk can take effect.
Note If you checked Enable Application Level Authorization, check the Accept Presence
Subscription check box in the Application User Configuration window for the application.
Configure a Calling Search Space for Presence Requests

The SUBSCRIBE Calling Search space option allows you to apply a calling search space separate from the
call-processing Calling Search Space for BLF presence requests. Select a different calling search space for
presence requests, else the SUBSCRIBE Calling Search Space selects the None default option. The
SUBSCRIBE Calling Search Space that is associated with an end user is used for extension mobility calls.
You apply the SUBSCRIBE Calling Search Space to the SIP trunk, phone, or end user. The SUBSCRIBE
Calling Search Space that is associated with an end user is used for extension mobility calls.
Before you begin

Accept BLF Presence Requests from External Trunks and Applications, on page 527
Procedure
Step 2 In the Calling Search Space configuration window, choose the calling search space from the SUBSCRIBE
Calling Search Space drop-down list.
528
Configure a Phone Button Template for BLF and SpeedDial Buttons

Step 5 (Optional) In the Description field, enter a description to identify the calling search space.
Step 6 From the Available Partitions list, select one or multiple partitions, and click the arrow keys.
The selected partitions appear in the Selected Partitions list.
Step 7 (Optional) To add or remove a partition from the Selected Partitions list, click the arrow keys next to the list
box.
Step 8 Click Save.
All calling search spaces that you configure in Cisco Unified Communications Manager Administration appear
in the SUBSCRIBE Calling Search Space drop-down list in the Trunk Configuration or Phone
Configure a Phone Button Template for BLF and SpeedDial Buttons

You can configure BLF and SpeedDial buttons for a phone or user device profile. After you apply the template
to the phone or device profile (and save the phone or device profile configuration), the Add a new BLF SD
link appears in the Association Information pane in Cisco Unified Communications Administration.
Note If the template does not support BLF and SpeedDials, the Add a new BLF SD link appears in the Unassigned
Associated Items pane.
When an administrator decides to add or change a BLF and SpeedDial button for a SIP URI, the administrator
ensures that the watcher is authorized to monitor that destination. If the system uses a SIP trunk to reach a
SIP URI BLF target, the BLF presence group associated with the SIP trunk applies.
Note You do not need to configure BLF presence groups or the Default Inter-Presence Group Subscription parameter
for BLF and SpeedDials.
Before you begin

Configure a Calling Search Space for Presence Requests, on page 528
Procedure
Step 2 Click the Add New button.
The Phone Button Template Configuration window appears.
Step 3 In the Button Template Name field, enter a name for the template.
Step 4 From the Phone Button Template drop-down list, select a template of phone button.
Step 5 Click Copy to create a new button template based on the layout of the selected button template.
529
Associate Button Template with a Device
Step 6 Click Save.
Associate Button Template with a Device

You configure BLF and SpeedDial buttons for a phone or user device profile. The BLF value does not have
to be on the cluster. For information on the Busy Lamp Field (BLF) status icons that display on the phone,
see the Cisco Unified IP Phone documentation that supports your phone. To identify whether your phone
supports BLF presence, see the Cisco Unified IP Phone documentation that supports your phone and this
version of Unified Communications Manager.
Before you begin

Configure a Phone Button Template for BLF and SpeedDial Buttons, on page 529
Procedure
Step 2 Enter the search parameters to find the configured phone button templates, and click Find.
The records matching all the search criteria appear.
Step 3 Click one of the records.
The Device Profile Configuration window appears.
Step 4 From the Phone Button Template list, select a configured phone button template.
Step 5 (Optional) Modify the values of the configured device.
Step 6 Click Save.
Configure User Device Profile

See the “BLF Presence with Extension Mobility” section of BLF Presence Interactions, on page 531 for details.
Before you begin

Associate Button Template with a Device, on page 530
Procedure
Step 1 In the Cisco Unified CM Administration, choose Device > Device Settings > Device Profile.
The Device Profile Configuration window appears.
Step 3 Configure the fields in Device Profile Configuration window. See the online help for information about the
Note If the phone button template that you applied to the phone or device profile does not support BLF
and SpeedDials, the link does not appear in the Association Information pane, but appears in
the Unassigned Associated Items pane.
530
BLF Presence Interactions
Step 4 Click Save.
BLF Presence Interactions

Feature Interaction
Presence BLF with When the H.323 phone is in the RING IN state, the BLF status gets reported as Busy.
DNs on H.323 phones For the presence entities of phones that are running either SCCP or SIP and that are
when the H.323 in the RING IN state, the BLF status gets reported as Idle.
phone device serves
as presence entity
Presence BLF with When the H.323 phone is not connected to Cisco Unified Communications Manager
DNs on H.323 phones for any reason, such as the Ethernet cable is unplugged from the phone, the BLF
when the H.323 status gets reported as Idle all the time. For presence entities of phones that are running
phone device serves either SCCP or SIP and that are not connected to Cisco Unified Communications
as presence entity Manager, the BLF status gets reported as Unknown.
BLF Presence with When you configure BLF and SpeedDial buttons in a user device profile in Cisco
Extension Mobility Unified Communications Manager Administration, a phone that supports Cisco
Extension Mobility displays BLF presence status on the BLF and SpeedDial buttons
after you log in to the device.
When the extension mobility user logs out, a phone that supports Cisco Extension
Mobility displays BLF presence status on the BLF and SpeedDial buttons for the
logout profile that is configured.
BLF Presence Restrictions

SIP Presence Cisco Unified Communications Manager Assistant does not support SIP presence.
BLF Presence Requests Cisco Unified Communications Manager Administration rejects BLF presence
requests to a directory number that is associated with a hunt pilot.
BLF on Call List Feature The BLF on call list feature is not supported on the Cisco Unified IP Phone 7940
and Cisco Unified IP Phone 7960.
531
BLF Presence Restrictions
BLF and SpeedDials The administrator ensures that the watcher is authorized to monitor the destination
when configuring a BLF and SpeedDial. BLF presence group authorization does
not apply to BLF and SpeedDials.
Note BLF presence group authorization does not apply to any directory
number or SIP URI that is configured as a BLF and Speed Dial that
appears in a call list for phones that are running SIP.
If there is an overlapping DN, where there is the same extension in different

partitions, the presence notifications are selected based on the order of the
partitions configured within the SUBSCRIBE CSS assigned to the device.
For example, two BLF speed dials are configured on a phone.
• Extension 1234 in the "internal" partition
• Extension 1234 in the "external" partition
Whichever partition is listed first within the SUBSCRIBE CSS is the one that
will provide BLF presence to the subscribed devices.
BLF Presence For Cisco Unified IP Phones with multiple lines, the phone uses the cached
Authorization information that is associated with the line directory number for missed and
placed calls to determine BLF presence authorization. If this call information is
not present, the phone uses the primary line as the subscriber for BLF presence
authorization. For BLF and SpeedDial buttons on Cisco Unified IP Phones with
multiple lines, the phone uses the first available line as the subscriber.
Cisco Unified IP Phone When a user monitors a directory number that is configured for Cisco Unified
IP Phones 7960 and 7940 that are running SIP, the system displays a status icon
for ‘not on the phone’ on the watcher device when the presence entity is off-hook
(but not in a call connected state). These phones do not detect an off-hook status.
For all other phone types, the system displays the status icon for ‘on the phone’
on the watcher device for an off-hook condition at the presence entity.
SIP Trunks BLF presence requests and responses must route to SIP trunks or routes that are
associated with SIP trunks. The system rejects BLF presence requests routing to
MGCP and H323 trunk devices.
BLF Presence-supported For BLF presence-supported phones that are running SIP, you can configure
Phones that are running directory numbers or SIP URIs as BLF and SpeedDial buttons. For BLF
SIP presence-supported phones that are running SCCP, you can only configure
directory numbers as BLF and SpeedDial buttons.
Phones that are running For phones that are running SIP, BLF presence group authorization also does not
SIP apply to any directory number or SIP URI that is configured as a BLF and Speed
Dial that appears in a call list.
532
CHAPTER 41
Call Display Restrictions
• Call Display Restrictions Overview, on page 533
• Call Display Restrictions Configuration Task Flow, on page 533
• Call Display Restrictions Interactions, on page 542
• Call Display Restrictions Feature Restrictions, on page 544
Call Display Restrictions Overview

Cisco Unified Communications Manager provides flexible configuration options that allow and also restrict
the display of the number and name information for both calling and connected users. You can restrict connected
numbers and names independently of each other.
You can configure connected number and name restrictions on the SIP trunk level or on a call-by-call basis.
The SIP trunk level configuration overrides a call-by-call configuration.
For example, in a hotel environment, you may want to see the display information for calls that are made
between a guest room and the front desk. However, for calls between guest rooms, you can restrict the call
information to display on either phone.
Call Display Restrictions Configuration Task Flow

Before you begin
• Review Call Display Restrictions Interactions, on page 542
Procedure

Step 1 Generate a Phone Feature List, on page 1 Generate a report to identify endpoints that
support the Call Display Restrictions feature.
Step 2 Configure Partitions for Call Display Configure partitions to create a logical grouping
Restrictions, on page 534 of directory numbers (DN) and route patterns
with similar reachability characteristics. For
example, in a hotel environment, you can a
533
Configure Partitions for Call Display Restrictions

configure a partition for dialing between rooms,
and a partition for dialing the public switched
telephone network (PSTN).
Step 3 Configure Calling Search Spaces for Call Configure calling search spaces to identify the
Display Restrictions, on page 535. partitions that calling devices can search when
they attempt to complete a call. Create calling
search spaces for rooms, the front desk, other
hotel extensions, the PSTN, and the room park
range (for call park).
Step 4 Configure the Service Parameter for Connected Configure the service parameter to display the
Number Display Restriction, on page 536. connected line ID as dialed digits only.
Step 5 Configure Translation Patterns, on page 537. Configure translation patterns with different
levels of display restrictions.
Step 6 Configure Phones for Call Display Restrictions, Associate endpoints with the partitions and the
on page 538 calling search spaces that you want to use for
call display restrictions.
Step 7 Configure the PSTN Gateway for Call Display Associate the PSTN gateway with the partitions
Restrictions, on page 540 and the calling search spaces that you want to
use for call display restrictions.
Step 8 Optional. Configure Call Display Restrictions Use this procedure to configure connected
on SIP Trunks, on page 540 number and name restrictions on the SIP trunk
level. The SIP trunk level configuration
overrides call-by-call configuration.
Configure Partitions for Call Display Restrictions

are based on organization, location, and call type. You can configure multiple partition
Procedure
name in this field.
534
Partition Name Guidelines
incoming call.
Step 8 Click Save.
Partition Name Guidelines

2 characters 340
3 characters 256
4 characters 204
5 characters 172
... ...
10 characters 92
15 characters 64
Configure Calling Search Spaces for Call Display Restrictions

Configure calling search spaces to identify the partitions that calling devices can search when they attempt
to complete a call. Create calling search spaces for rooms, the front desk, other hotel extensions, the PSTN,
and the room park range (for call park).
Before you begin

Configure Partitions for Call Display Restrictions, on page 534
535
Configure the Service Parameter for Connected Number Display Restriction
Procedure

Partitions box.
Step 8 Click Save.
Configure the Service Parameter for Connected Number Display Restriction

The connected number display restriction restricts the connected line ID display to dialed digits only. This
option addresses customer privacy issues as well as connected number displays that are meaningless to phone
users.
Before you begin

Configure Calling Search Spaces for Call Display Restrictions, on page 535
Procedure
Step 2 Select the server where the Cisco CallManager service runs, and then select the Cisco CallManager service.
Step 3 Set the Always Display Original Dialed Number service parameter to True to enable this feature.
The default value is False.
Step 4 (Optional) Set the Name Display for Original Dialed Number When Translated service parameter.
The default field shows the alerting name of the original dialed number before translation. You can change
this parameter to show the alerting name of the dialed number after translation. This parameter is not applicable
if the Always Display Original Number service parameter is set to False.
536
Configure Translation Patterns
Step 5 Click Save.
Configure Translation Patterns

Unified Communications Manager uses translation patterns to manipulate dialed digits before it routes a call.
In some cases, the system does not use the dialed number. In other cases, the public switched telephone
network (PSTN) does not recognize the dialed number. For the Call Display Restrictions feature, calls are
routed through different translation patterns before the calls are extended to the actual device.
Before you begin

Configure the Service Parameter for Connected Number Display Restriction, on page 536
Procedure
Step 2 Configure the fields in the Translation Pattern Configuration window. See Translation Pattern Fields for
Call Display Restrictions, on page 537 for more information about the fields and their configuration options.
Step 3 Click Save.
Translation Pattern Fields for Call Display Restrictions

Field Description
Translation Pattern Enter the translation pattern, including numbers and
wildcards. Do not use spaces. For example, for the NANP,
enter 9.@ for typical local access or 8XXX for a typical
private network numbering plan.
Valid characters include the uppercase characters A, B, C,
and D and \+, which represents the international escape
character +.
Description Enter a description for the translation pattern. The

description can include up to 50 characters in any language,
but it cannot include double quotes ("), percentage sign
(%), ampersand (&), or angle brackets (<>).
Partition From the drop-down list, choose the partition to associate

with this translation pattern.
Calling Search Space From the drop-down list, choose the calling search space
to associate with this translation pattern.
537
Configure Phones for Call Display Restrictions
Field Description
Calling Line ID Presentation From the drop-down list, choose one of the following
options:
• Default—Choose this option if you do not want to
change the presentation of the calling line ID.
• Allowed—Choose this option if you want to display
the phone number of the calling party.
• Restricted—Choose this option if you want Cisco
Unified Communications Manager to block the
display of the calling party phone number.
Calling Name Presentation From the drop-down list, choose one of the following
options:
change the presentation of the calling name.
the name of the calling party.
display of the calling name.
Connected Line ID Presentation From the drop-down list, choose one of the following
options:
change the presentation of the connected line ID.
the phone number of the connected party.
display of the connected party phone number.
Connected Name Presentation From the drop-down list, choose one of the following
options:
change the presentation of the connected name.
the name of the connected party.
display of the connected name.

Use this procedure to associate phones with the partitions and the calling search spaces used for call display
restrictions.
538
Before you begin

Configure Translation Patterns, on page 537
Procedure

a) To modify the fields for an existing phone, enter search criteria and choose a phone from the resulting
list. The Phone Configuration window appears.
b) To add a new phone, click Add New.
Step 3 From the Calling Search Space drop-down list, choose the calling search space that you want the system to
use when it determines how to route a dialed number.
Step 4 Check the Ignore presentation indicators (internal calls only) check box to ignore any presentation restriction
on internal calls.
Step 5 Click Save.
The phone is added to the database.
Step 6 To associate the added phone to a directory number, choose Device > Phone, enter search parameters to
search the phone that you added.
Step 7 In the Find and List Phones window, click the phone name.
Step 8 From the Association pane, click the phone name to add or modify the directory number.
Step 9 In the Directory Number Configuration window, add or modify the value of directory number in the
Directory Number text box, and select a value in the Route Partition drop-down list.
Step 10 Click Save.
Phone Configuration Example

Configure phone A (Room-1) with partition P_Room and device/line calling search space
CSS_FromRoom
{ P_Phones, CSS_FromRoom} : 221/Room-1
Configure phone B (Room-2) with partition P_Room and device/line calling search space
CSS_FromRoom
{ P_Phones, CSS_FromRoom} : 222/Room-2
Configure phone C (Front Desk-1) with partition P_FrontDesk and device/line calling search space
CSS_FromFrontDesk and Ignore Presentation Indicators check box enabled
{ P_FrontDesk, CSS_FromFrontDesk, IgnorePresentationIndicators set} : 100/Reception
Configure phone D (Front Desk-2) with partition P_FrontDesk and device/line calling search space
CSS_FromFrontDesk and Ignore Presentation Indicators check box enabled
539
Configure the PSTN Gateway for Call Display Restrictions
{ P_FrontDesk, CSS_FromFrontDesk, IgnorePresentationIndicators set} : 200/Reception

Configure phone E (Club) with partition P_Club and calling search space CSS_FromClub
{ P_Club, CSS_FromClub) : 300/Club
Configure the PSTN Gateway for Call Display Restrictions

Associate the PSTN gateway with the partitions and the calling search spaces that you want to use for call
display restrictions.
Before you begin

Configure Phones for Call Display Restrictions, on page 538
Procedure
Step 1 In Cisco Unified CM Administration, choose Device > Gateway.

Step 2 Enter search criteria and choose the PSTN gateway from the resulting list.
The Gateway Configuration window appears.
Step 3 From the Calling Search Space drop-down list, choose the calling search space that you want the system to
use when it determines how to route an incoming call from the PSTN.
Step 4 Click Save and Reset to apply the configuration changes.
Step 5 (Optional) To associate the available trunk or gateway, in Cisco Unified Communications Manager
Administration, choose SIP Route Pattern, and select a SIP trunk or route list from the SIP Trunk/Route
List drop-down list.
Gateway Configuration Example

Configure PSTN Gateway E with route pattern P_PSTN and calling search space CSS_FromPSTN
{CSS_FromPSTN}, RoutePattern {P_PSTN}
Configure Call Display Restrictions on SIP Trunks

You can configure connected number and name restrictions on the SIP trunk level. The SIP trunk-level
configuration overrides call-by-call configuration.
Before you begin

(Optional) Configure the PSTN Gateway for Call Display Restrictions, on page 540
Procedure

The Find and List Trunks window appears.
540
SIP Trunk Fields for Call Display Restrictions
Step 2 Enter search criteria and click Find.

Step 3 Select the name of the trunk that you want to update.
Step 4 Configure the fields in the SIP Trunk Configuration window. See SIP Trunk Fields for Call Display
Restrictions, on page 541 for more information about the fields and their configuration options.
Step 5 Click Save.
SIP Trunk Fields for Call Display Restrictions

Table 49: Inbound Calls
Field Description
Calling Line ID Presentation From the drop-down list, choose one of the following
options:
change the presentation of the calling line ID.
the phone number of the calling party.
display of the calling party phone number.
Calling Name Presentation From the drop-down list, choose one of the following
options:
change the presentation of the calling name.
the name of the calling party.
display of the calling name.
Calling Search Space From the drop-down list, choose the calling search space
to associate with this translation pattern.
541
Call Display Restrictions Interactions
Table 50: Outbound Calls
Field Description
Connected Line ID Presentation From the drop-down list, choose one of the following
options:
change the presentation of the connected line ID.
the phone number of the connected party.
display of the connected party phone number.
Connected Name Presentation From the drop-down list, choose one of the following
options:
change the presentation of the connected name.
the name of the connected party.
display of the connected name.

This section describes how the Call Display Restrictions feature interacts with Cisco Unified Communications
Manager applications and call processing features.
542
Feature Interaction
Call Park When you use the Call Display Restrictions feature with Call Park, you must
configure an associated translation pattern for each individual call park number
to preserve the Call Display Restrictions feature. You cannot configure a single
translation pattern to cover a range of call park numbers.
Consider the following scenario as an example:
1. The system administrator creates a call park range of 77x and places it in a
partition called P_ParkRange. (The phones in the guest rooms can see that
the P_ParkRange partition is made visible to the phones in the guest rooms
by inclusion of it in the calling search space of the phones [CSS_FromRoom]).
2. The administrator configures a separate translation pattern for each call park
directory number and configures the display fields to Restricted. (In the
current scenario, the administrator creates translations patterns for 770, 771,
772...779.)
Note For the Call Display Restrictions feature to work correctly, the
administrator must configure separate translation patterns and
not a single translation pattern for a range of numbers (such as
77x or 77[0-9]).
3. Room-1 calls Room-2.

4. Room-2 answers the call, and Room-1 parks the call.
5. When Room-1 retrieves the call, Room-2 does not see Room-1 call
information display.
See Call Park Overview

Conference List When you use Call Display Restrictions, you restrict the display information for
the list of participants in a conference.
See Ad Hoc Conferencing Overview
Conference and Voice When you use Call Display Restrictions with features, such as conference and
Mail voice mail, the call information display on the phones reflects that status. For
example, when the conference feature is invoked, the call information display
shows To Conference. When voice mail is accessed by choosing the Messages
button, the call information display shows To Voicemail.
Extension Mobility To use Call Display Restrictions with Extension Mobility, enable the Ignore
Presentation Indicators (internal calls only) parameter in both the Cisco Unified
Communications Manager Administration Phone Configuration window and the
Cisco Unified Communications Manager Administration Device Profile
When you enable Call Display Restrictions with Extension Mobility, the
presentation or restriction of the call information depends on the line profile that
is associated with the user who is logged in to the device. The configuration that
is entered in the user device profile (associated with the user) overrides the
configuration that is entered in the phone configuration (of the phone that is
enabled for Extension Mobility).
543
Call Display Restrictions Feature Restrictions
Feature Interaction
Call Forwarding The Connected Number Display restriction applies to all calls that originate in
the system. When this value is set to True, this field interacts with existing Cisco
Unified Communications Manager applications, features, and call processing.
This value applies to all calls that terminate inside or outside the system. The
Connected Number Display is updated to show the modified number or redirected
number when a call is routed to a Call Forward All or Call Forward Busy
destination, or gets redirected through a call transfer or CTI application.
Call Display Restrictions Feature Restrictions

Translation Patterns—Duplicate entries are not allowed in translation patterns.
544
CHAPTER 42
Do Not Disturb
• Do Not Disturb Overview, on page 545
• Do Not Disturb Configuration Task Flow, on page 546
• Do Not Disturb Interactions and Restrictions, on page 554
• Do Not Disturb Troubleshooting, on page 556
Do Not Disturb Overview

Do Not Disturb (DND) provides the following options:
• Call Reject—This option specifies that the incoming call gets rejected. Depending on how you configure
the DND Incoming Call Alert parameter, the phone may play a beep, or display a flash notification of
the call.
• Ringer Off—This option turns off the ringer, but incoming call information gets presented to the device,
so that the user can accept the call.
When DND is enabled, all new incoming calls with normal priority honor the DND settings for the device.
High-priority calls, such as Cisco Emergency Responder (CER) calls, or calls with Multilevel Precedence
and Preemption (MLPP), ring on the device. Also, when you enable DND, the Auto Answer feature gets
disabled.
Users can activate Do Not Disturb on the phone in the following ways:
• Softkey
• Feature button
• Cisco Unified Communications Self-Care Portal
Note You can also enable or disable the feature on a per-phone basis from within Cisco Unified Communications
Manager.
Phone Behavior
When you enable Do Not Disturb, the Cisco Unified IP Phone displays the message “Do Not Disturb is active”.
Some Cisco Unified IP Phones display DND status icons. For details on how individual phone models use
Do Not Disturb, consult the user guide for that particular phone model.
545
Do Not Disturb Configuration Task Flow
When you activate DND, you can still receive incoming call notifications on the phone as specified by the
Incoming Call Alert settings in Cisco Unified Communications Manager Administration, but the phone will
not ring, except for high-priority calls (such as Cisco Emergency Responder and MLPP calls). Also, if you
enable DND while the phone is ringing, the phone stops ringing.
Status Notifications
Do Not Disturb is supported on both SIP and Cisco Skinny Call Control Protocol (SCCP) devices.
SIP phones use the SIP PUBLISH method to signal a DND status change to Cisco Unified Communications
Manager. Cisco Unified Communications Manager uses a Remote-cc REFER request to signal a DND status
change to the SIP phone.
SCCP phones use SCCP messaging to signal a DND status change to Cisco Unified Communications Manager.
Do Not Disturb Configuration Task Flow

Procedure

Step 1 Generate a Phone Feature List, on page 1 Run a Phone Feature List report from Cisco
Unified Reporting to determine which phones
support Do Not Disturb.
Note Cisco Unified IP Phones 7940 and
7960 that are running SIP use their
own backwards-compatible
implementation of Do Not
Disturb, which you configure on
the SIP Profile.
Step 2 Configure Busy Lamp Field Status, on page 547 Configure the Busy Lamp Field status service
parameter.
Step 3 Configure Do Not Disturb on a Common Phone Optional. Configure Do Not Disturb against a
Profile, on page 547 Common Phone Profile. The profile allows you
to apply Do Not Disturb settings to a group of
phones in your network.
Step 4 Apply Do Not Disturb Settings to the Phone, Apply Do Not Disturb settings to the phone.
on page 548.
Step 5 Depending on whether your phone uses softkeys Add a Do Not Disturb feature button or softkey
or feature buttons, perform either of the to your phone.
following tasks:
• Configure a Do Not Disturb Feature
Button, on page 549
• Configure a Do Not Disturb Softkey, on
page 550
546
Configure Busy Lamp Field Status
Configure Busy Lamp Field Status

You can configure how the BLF status depicts Do Not Disturb by setting the BLF Status Depects DND
service parameter. To set the BLF status, do the following:
Before you begin
Note • Busy Lamp Field (BLF) presence status for DND works only when all the registered devices for that
shared line DN are set to DND.
• If you're using Jabber for iOS or Jabber for Android on the same DN, they are considered registered,
even when they are not registered but just configured.
Procedure
Step 1 In Cisco Unified CM Administration, choose System > Service Parameters.

Step 2 Choose the Cisco CallManager service for the server that you want to configure.
Step 3 In the Clusterwide Parameters (System - Presence) pane, specify one of the following values for the BLF
Status Depicts DND service parameter:
• True—If Do Not Disturb is activated on the device, the BLF status indicator for the device or line
appearance reflects the Do Not Disturb state.
• False—If Do Not Disturb is activated on the device, the BLF status indicator for the device or line
appearance reflects the actual device state.
What to do next
Configure Do Not Disturb on a Common Phone Profile, on page 547
Apply Do Not Disturb Settings to the Phone, on page 548
Configure Do Not Disturb on a Common Phone Profile

Common Phone Profiles allow you to configure Do Not Disturb settings and then apply those settings to a
group of phones in your network that use that profile.
Before you begin

Configure Busy Lamp Field Status, on page 547
547
Apply Do Not Disturb Settings to the Phone
Procedure
Step 2 From the DND Option drop-down list, choose how you want the Do Not Disturb feature to handle incoming
calls.
• Call Reject—No incoming call information gets presented to the user. Depending on how you configure
the DND Incoming Call Alert parameter, the phone may play a beep or display a flash notification of
the call.
• Ringer Off—This option turns off the ringer, but incoming call information gets presented to the device,
so the user can accept the call.
Note For mobile phones and dual-mode phones, you can only choose the Call Reject option.
Step 3 From the Incoming Call Alert drop-down list, choose how you want to alert phone users of incoming calls
while Do Not Disturb is turned on.
• Disable—Both beep and flash notification of a call are for disabled. If you configured the DND Ringer
Off option, incoming call information still gets displayed. However, for the DND Call Reject option, no
call alerts display, and no information gets sent to the device.
• Flash Only—The phone flashes for incoming calls.
• Beep Only—The phone displays a flash alert for incoming calls.
Step 4 Click Save.
Apply Do Not Disturb Settings to the Phone

This procedure describes how to apply Do Not Disturb settings on your Cisco Unified IP Phones. You can
apply DND settings through the Phone Configuration window in Cisco Unified CM Administration, or you
can apply DND settings to a Common Phone Profile and then apply that profile to your phone.
Before you begin

If you are using a Common Phone Profile, complete Configure Do Not Disturb on a Common Phone Profile,
on page 547.
Otherwise, complete Configure Busy Lamp Field Status, on page 547
Procedure
Step 1 From Cisco Unified CM Administration, choose Device > Phone

Step 2 Click Find and select the phone on which you want to configure Do Not Disturb.
Step 3 If you want to apply Do Not Disturb settings from a Common Phone Profile, from the Common Phone Profile
drop-down list, choose the profile on which you have configured Do Not Disturb.
Step 4 Check the Do Not Disturb check box to enable Do Not Disturb on the phone.
Step 5 In the DND Option drop-down list, specify from the following options how you want the DND feature to
handle incoming calls.
548
Configure a Do Not Disturb Feature Button
• Call Reject—No incoming call information gets presented to the user. Depending on the configuration,
the phone either plays a beep or displays a flash notification.
• Ringer Off—Incoming call information gets presented to the device so that the user can accept the call,
but the ringer is turned off.
• Use Common Profile Setting—The Do Not Disturb setting for the Common Phone Profile that is
specified for this device gets used.
Note For 7940/7960 phones that are running SCCP, you can only choose the Ringer Off option. For
mobile devices and dual-mode phones, you can only choose the Call Reject option. When you
activate DND Call Reject on a mobile device or dual-mode phone, no call information gets
presented to the device.
Step 6 In the DND Incoming Call Alert drop-down list, specify from the following options how you want the phone
to display an incoming call when DND is turned on.
• None—The DND Incoming Call Alert setting from the Common Phone Profile gets used for this device.
• Disable—For DND Ringer Off, both beep and flash notifications are disabled, but incoming call
information is still displayed. For Call Reject, beep and flash notifications are disabled, and no incoming
call information gets passed to the device.
• Beep only—For incoming calls, the phone plays a beep tone only.
• Flash only—For incoming calls, the phone displays a flash alert.
Step 7 Click Save.
What to do next
Complete either of the following procedures:
Configure a Do Not Disturb Feature Button, on page 549
Configure a Do Not Disturb Softkey, on page 550
Configure a Do Not Disturb Feature Button

Follow these steps to add a Do Not Disturb feature button to your Cisco Unified IP Phone.
Procedure

Step 1 Configure Phone Button Template for Do Not Create a phone button template that includes
Disturb, on page 549 the Do Not Disturb button.
Step 2 Associate a Button Template with a Phone, on Associate the Do Not Disturb button template
page 249 to a phone.
Configure Phone Button Template for Do Not Disturb

Follow this procedure to configure a phone button template that includes the Do Not Disturb button.
549
Associate Button Template with Phone
Procedure
step.
c) Click Save.
Step 6 Click Save.
devices.
Associate Button Template with Phone
Before you begin

Configure Phone Button Template for Do Not Disturb, on page 549
Procedure

button.
Step 5 Click Save.
Configure a Do Not Disturb Softkey

Optional. If your phone uses softkeys, perform the tasks in the following task flow to add a Do Not Disturb
softkey to the phone.
550
Configure Softkey Template for Do Not Disturb
Procedure

Step 1 Configure Softkey Template for Do Not Create a softkey template that includes the Do
Disturb, on page 551 Not Disturb softkey.
Step 2 Perform either of the following procedures: You can associate the softkey to a Common
Device Configuration and then associate that
configuration to a group of phones, or you can
Common Device Configuration, on page
associate the softkey template directly to a
552
phone.
• Associate Softkey Template with a Phone,
on page 553
Configure Softkey Template for Do Not Disturb

Perform these steps to configure a softkey template that includes the Do Not Disturb softkey.
Procedure
a) Click Add New.
d) Click Save.
template.
click Go.
to display.
Step 9 Click Save.
devices.
551
What to do next
Perform one of the following procedures to add the softkey template to a phone.
Associate a Softkey Template with a Common Device Configuration, on page 552
Associate Softkey Template with a Phone, on page 553

When you associate the Do Not Disturb (DND) softkey template to a Common Device Configuration you
can add the DND softkey to a group of Cisco Unified IP Phones that use that Common Device Configuration.
Before you begin

Configure Softkey Template for Do Not Disturb, on page 551
Procedure

Step 1 Add Softkey Template to Common Device Associate the DND softkey template to a
Configuration, on page 552 Common Device Configuration.
Step 2 Associate Common Device Configuration with Add the DND softkey to a phone by associating
Phone, on page 553 the Common Device Configuration to the
phone.
Add Softkey Template to Common Device Configuration
Procedure
a) Click Add New.
c) Click Save.
to make available.
552
Step 5 Click Save.

restart them.
Before you begin

Associate a Softkey Template with a Common Device Configuration, on page 552
Procedure

Step 4 Click Save.
Associate Softkey Template with a Phone

Perform this procedure if you have configured a softkey template with the Do Not Disturb softkey and you
want to associate that softkey template to a phone.
Before you begin

Configure Softkey Template for Do Not Disturb, on page 551
Procedure

Step 4 Click Save.
553
Do Not Disturb Interactions and Restrictions
Do Not Disturb Interactions and Restrictions

This section provides information about Do Not Disturb interactions and restrictions.
Interactions
The following table describes feature interactions with the Do Not Disturb (DND) feature. Unless otherwise
stated, the interactions apply to both the DND Ringer Off and the DND Call Reject option.
Feature Interaction with Do Not Disturb
Call Forward All On Cisco Unified IP Phones, the message that indicates that the Do Not Disturb
(DND) feature is active takes priority over the message that indicates that the user
has new voice messages. However, the message that indicates that the Call Forward
All feature is active has a higher priority than DND.
Park Reversion For locally parked calls, Park Reversion overrides DND. If Phone A has DND
turned on, and a call is parked, the park reversion to Phone A occurs and Phone
A rings.
For remotely parked calls, DND overrides Park Reversion:
• If Phone A activates DND Ringer Off and shares a line with Phone A-prime,
when Phone A-prime parks the call, park reversion on Phone A honors the
DND settings and does not ring.
• If Phone A activated DND Call Reject, the park reversion is not presented to
Phone A.
Pickup For locally placed Pickup requests, Pickup overrides DND. If Phone A has DND
turned on, and has initiated any type of Pickup, the Pickup call presents normally,
and Phone A rings.
For remotely placed Pickup requests, DND overrides Pickup as follows:
• If Phone A is in DND Ringer Off mode and shares a line with Phone A-prime,
when Phone A-prime initiates Pickup, the Pickup call to Phone A honors the
DND settings and Phone A does not ring.
• If Phone A is in DND Call Reject mode, the Pickup call is not presented to
Phone A.
Hold Reversion and Hold Reversion and Intercom override DND, and the call gets presented normally.
Intercom
MLPP and CER Multilevel Precedence and Preemption (phones that are running SCCP) and Cisco
Emergency Responder calls override DND. Multilevel Precedence and Preemption
and Cisco Emergency Responder calls get presented normally, and the phone ring
is supported on both SCCP and SIP.
554
Restrictions
Feature Interaction with Do Not Disturb
Call Back For the originating side, callback overrides DND. When the activating device is
on DND mode, the callback notification (both audio and visual) is still presented
to the user.
For the terminating side, DND overrides callback as follows:
• When the terminating side is on DND Ringer Off, the Callback Available
screen is sent after the terminating side goes off hook and on hook.
• When the terminating side is on DND Call Reject, and is available, a new
screen is sent to the activating device as "<DirectoryNumber> has become
available but is on DND-R" if the activating device is in same cluster. Callback
available notification is sent only after the terminating side disables DND
Call Reject.
Pickup Notification For the DND Ringer Off option, only visual notification gets presented to the
device.
For the DND Call Reject option, no notification gets presented to the device.
Hunt List If a device in a Hunt List has DND Ringer Off activated, the call is still presented
to the user. However, the DND Incoming Call Alert settings would still apply.
If a device in a Hunt List has DND Call Reject activated, any calls to that Hunt
List will go to the next member and will not get sent to this device.
Extension Mobility For Extension Mobility, the device profile settings include DND incoming call
alert and DND status. When a user logs in and enables DND, the DND incoming
call alert and DND status settings get saved, and these settings get used when the
user logs in again.
Note When a user who is logged in to Extension Mobility modifies the
DND incoming call alert or DND status settings, this action does not
affect the actual device settings.
Restrictions
Some restrictions apply to DND usage, depending on the phone or device type in use.
• The following phone models and devices that are running SCCP support only the DND Ringer Off option:
• Cisco IP Communicator
Note Cisco Unified IP Phones 7940 and 7960 that run SIP use their own implementation
of Do Not Disturb, which is backward compatible.
555
Do Not Disturb Troubleshooting
• The following phone models and devices support only the DND Call Reject option:
• Mobile devices (dual mode)
• Remote Destination Profile
• Cisco Unified Mobile Communicator
Do Not Disturb Troubleshooting

This section provides troubleshooting information for Cisco Unified IP Phones (SCCP and SIP).
For SIP phones, use the following information for troubleshooting:
• debugs: sip-dnd, sip-messages, dnd-settings
• show: config, dnd-settings
• sniffer traces
For SCCP phones, use the following information for troubleshooting:

• debug: jvm all info
• sniffer traces
Troubleshooting Errors
The following table describes how to troubleshoot errors with Do No Disturb.
Symptom Actions
DND softkey does not • Verify that the softkey or button template for this phone includes DND.
display
• Capture a sniffer trace and verify that the phone gets the correct softkey or
or button template.
DND feature button does • Verify that the phone firmware is Version 8.3(1) or later.
not display
BLF speed dial does not • Verify that the BLF DND is set to enabled in Enterprise parameters.
show DND status
• Capture a sniffer trace and verify that the phone gets the correct notification
message.
• Verify that the phone firmware is Version 8.3(1) or later.
DND changes are not • Check if the BOT/TCT devices are shared line devices with the DND state
reflected on the set to OFF. If the status is set to ON, changes to the DND status on other
monitoring device. shared lines will not be reflected.
• Make sure the DND status on the BOT/TCT devices is set to OFF to reflect
changes in the DND status on the line you want to monitor.
556
CHAPTER 43
Privacy
• Privacy Overview, on page 557
• Privacy Configuration Task Flow, on page 558
• Privacy Restrictions, on page 561
Privacy Overview
The Privacy feature allows you to enable or disable the capability of users with phones that share the same
line (DN) to view call status and to barge into the call. You can enable or disable privacy for each phone or
for all phones. By default, the system enables privacy for all phones in the cluster.
When the device that is configured for privacy registers with Cisco Unified Communications Manager, the
feature button on the phone that is configured with privacy gets labeled, and the status is indicated through
an icon. If the button has a lamp, it comes on.
When the phone receives an incoming call, the user makes the call private (so the call information does not
display on the shared line) by pressing the Privacy feature button. The Privacy feature button toggles between
On and Off.
To verify if your Cisco Unified IP Phone supports Privacy, see the user documentation for your phone model.
Privacy on Hold
Privacy on Hold allows you to enable or disable the capability of users with phones that share the same line
(DN) to view call status and retrieve calls on hold.
You can enable or disable Privacy on Hold for specific phones or all the phones. Privacy on Hold activates
automatically on all private calls when Privacy on Hold is enabled. By default, the system disables Privacy
on Hold for all phones in the cluster.
To activate Privacy on Hold, users press the Hold softkey or Hold button while on a private call. To return
to the call, users press the Resume softkey. The phone that puts the call on hold displays the status indicator
for a held call; shared lines display the status indicators for a private and held call.
557
Privacy Configuration Task Flow
Privacy Configuration Task Flow

Procedure

support the Privacy feature.
Step 2 Enable Privacy Cluster-wide, on page 558 Enable Privacy by default for all the phones in
the cluster.
Step 3 Enable Privacy for a Device, on page 558 Enable Privacy for specific devices.
Step 4 Configure Privacy Phone Button Template, on Configure Privacy phone button template for a
page 559 device.
Step 5 Associate Privacy Phone Button Template with Associate the phone button template with a user.
a Phone, on page 559
Step 6 Configure Shared Line Appearance, on page Configure the shared line appearance.
560
Step 7 (Optional) Configure Privacy on Hold, on page Configure Privacy on Hold.
561
Enable Privacy Cluster-wide

Perform these steps to enable Privacy by default for the entire cluster.
Procedure
Step 4 From the Privacy Setting drop-down list, choose True.
Step 5 Click Save.
Enable Privacy for a Device

Before you begin
Ensure that the phone model supports Privacy. For more information, see Generate a Phone Feature List, on
page 1.
558
Configure Privacy Phone Button Template
Procedure

Step 2 Specify search criteria and click Find.
The phone search results appear.
Step 3 Select the phone.
Step 4 From the Privacy drop-down list, select Default.
Step 5 Click Save.
Configure Privacy Phone Button Template

Before you begin
Enable Privacy for a Device, on page 558
Procedure
step.
c) Click Save.
Step 6 Click Save.
devices.
Associate Privacy Phone Button Template with a Phone

Before you begin
Configure Privacy Phone Button Template, on page 559
559
Configure Shared Line Appearance
Procedure

button.
Step 5 Click Save.
Configure Shared Line Appearance

Before you begin
Associate Privacy Phone Button Template with a Phone, on page 559
Procedure

The Find and List Phones window appears.
Step 2 To locate a specific phone, enter search criteria and click Find.
A list of phones that match the search criteria is displayed.
Step 3 Choose the phone for which you want to configure shared line appearance.
Step 4 Click Add a new DN link in the Association Information area on the left side of the Phone Configuration
window.
Step 5 Enter the Directory Number and choose the Route Partition to which the directory number belongs.
Step 6 Configure the remaining fields in the Directory Number Configuration window. For more information on
Step 7 Repeat Step 3, on page 560 to Step 6, on page 560 for all the phones for which you want to create a shared line
appearance.
Note Ensure that you assign the same directory number and route partition to all the phones that are
part of the shared line appearance.
560
Configure Privacy on Hold
Configure Privacy on Hold

Procedure
Step 4 Set the Enforce Privacy Setting on Held Calls service parameter to True.
Step 5 Click Save.
Privacy Restrictions
CTI • CTI does not support Privacy through APIs that

TAPI and JTAPI applications invoke. CTI
generates events when Privacy is enabled or
disabled from an IP phone by using the Privacy
feature button.
• CTI does not support Privacy on Hold through
APIs that TAPI/JTAPI applications invoke. CTI
generates events when a Privacy-enabled call is
put on hold and when Privacy gets enabled or
disabled on held calls from an IP phone by using
the Privacy feature button.
561
Privacy Restrictions
562
CHAPTER 44
Private Line Automatic Ringdown
• Private Line Automatic Ringdown Overview, on page 563
• Private Line Automatic Ringdown Configuration Task Flow for SCCP Phones, on page 563
• Private Line Automatic Ringdown Configuration Task Flow for SIP Phones, on page 566
• Private Line Automatic Ringdown Troubleshooting, on page 567
Private Line Automatic Ringdown Overview

The Private Line Automatic Ringdown (PLAR) feature configures a phone so that when the user goes off
hook (or the NewCall softkey or line key gets pressed), the phone immediately dials a preconfigured number.
The phone user cannot dial any other number from the phone line that gets configured for PLAR.
PLAR works with features such as Barge, cBarge, or single button Barge. If you use PLAR with a feature,
you must configure the feature as described in the feature documentation, and you must configure the PLAR
destination, which is a directory number that is used specifically for PLAR.
Private Line Automatic Ringdown Configuration Task Flow for

SCCP Phones
Perform the following tasks to configure Private Line Automatic Ringdown (PLAR) on SCCP phones.
Procedure

Step 1 Create Partition, on page 564 Create a partition for the PLAR destination. The
only directory number that you can assign to
this partition is the PLAR destination.
Step 2 Assign Partitions to Calling Search Spaces, on Assign the partition to a unique CSS, and a CSS
page 564 that includes the PLAR destination device.
Step 3 Assign Partition to the Private Line Automatic Assign the null partition and a CSS to your
Ringdown Destination, on page 565 PLAR destination directory number.
563
Create Partition

Step 4 Configure Translation Pattern for Private Line Create a null translation pattern and assign it to
Automatic Ringdown on Phones, on page 565 your PLAR destination directory number.
Create Partition
Create a new partition for the Private Line Automatic Ringdown (PLAR) destination. For the feature to work,
only the null translation pattern that you configure for PLAR can be assigned to this partition.
Procedure
Step 3 In the Name field, enter a partition name and a description separated by a comma.
Step 4 Click Save.
Assign Partitions to Calling Search Spaces

For Private Line Automatic Ringdown (PLAR) on SCCP phones, you must configure two calling search
spaces (CSS):
• The first CSS should include the new partition for the null translation pattern as well as a partition that
routes to the destination phone.
• The second CSS should include only the new partition for the null translation pattern.
Before you begin

Create Partition, on page 564
Procedure
Step 1 From Cisco Unified CM Administration, choose Call Control > Class of Control > Calling Search Space.
Step 2 Click Find and select the calling search space for the PLAR destination device.
Step 3 Use the arrows to move both of the following partitions to the Selected Partitions list box: the new partition
that you created for the null translation pattern and a partition that routes to the destination device.
Step 4 Click Save.
Step 6 Enter a name and description for the calling search space.
Step 7 Use the arrows to move the new partition to the Selected Partitions list box.
Step 8 Click Save.
564
Assign Partition to the Private Line Automatic Ringdown Destination
Assign Partition to the Private Line Automatic Ringdown Destination

When configuring Private Line Automatic Ringdown (PLAR) on SCCP phones, assign a null partition to the
directory number that you want to use as the PLAR destination.
Note Each PLAR destination directory number must have its own unique partition. Do not add any other directory
numbers to the null partition that you created for the PLAR destination.
Before you begin

Assign Partitions to Calling Search Spaces, on page 564
Procedure
Step 1 In Cisco Unified CM Administration, choose Call Routing > Directory Number.
Step 2 Click Find and select the directory number that you want to use as the PLAR destination.
Step 3 In the Route Partition field, select a partition that you created for your PLAR destination.
Step 4 In the Calling Search Space drop-down list, select the CSS that includes both the null partition and the
destination device.
Step 5 Click Save.
Configure Translation Pattern for Private Line Automatic Ringdown on Phones

To configure Private Line Automatic Ringdown (PLAR) on phones, configure a null translation pattern and
assign the PLAR destination number to that translation pattern.
Before you begin

Assign Partition to the Private Line Automatic Ringdown Destination, on page 565
Procedure
Step 1 In Cisco Unified CM Administration, choose Call Routing > Translation Pattern.
Step 2 Click Add New to create a new translation pattern.
Step 3 Leave the Translation Pattern field empty.
Step 4 From the Partition drop-down list, select the new partition that you created for the null translation pattern.
Step 5 From the Calling Search Space drop-down list, select a calling search space that includes both the new
partition and the partition for the PLAR destination device.
Step 6 In the Called Party Transformation Mask field, enter the PLAR destination directory number.
Step 7 Click Save.
565
Private Line Automatic Ringdown Configuration Task Flow for SIP Phones
Private Line Automatic Ringdown Configuration Task Flow for

SIP Phones
Perform these tasks to configure Private Line Automatic Ringdown (PLAR) on SIP Phones.
Procedure

Step 1 Create SIP Dial Rule for Private Line Automatic Create a SIP dial rule for PLAR.
Ringdown, on page 566
Step 2 Assign Private Line Automatic Ringdown Dial Assign the PLAR dial rule to the phone.
Rule to SIP Phone, on page 566
Create SIP Dial Rule for Private Line Automatic Ringdown

To configure Private Line Automatic Ringdown (PLAR) on SIP phones, you must configure a SIP dial rule
for your PLAR destination number.
Before you begin

Create Partition, on page 564
Assign Partitions to Calling Search Spaces, on page 564
Assign Partition to the Private Line Automatic Ringdown Destination, on page 565
Configure Translation Pattern for Private Line Automatic Ringdown on Phones, on page 565
Procedure
Step 1 In Cisco Unified CM Administration, choose Call Routing > Class of Control > SIP Dial Rules.
Step 3 From the Dial Pattern drop-down list, choose 7940_7960_OTHER.
Step 4 Click Next.
Step 5 Enter a name and description for the dial rule.
Step 6 Click Next.
Step 7 In the Pattern field, enter a pattern that matches the PLAR destination number and click Add PLAR.
Step 8 Click Save.
Assign Private Line Automatic Ringdown Dial Rule to SIP Phone

You can configure Private Line Automatic Ringdown (PLAR) on SIP phones by assigning a PLAR-enabled
SIP Dial Rule to the phone.
566
Private Line Automatic Ringdown Troubleshooting
Before you begin

Create SIP Dial Rule for Private Line Automatic Ringdown, on page 566
Procedure

Step 2 Click Find and select the phone on which you want to configure PLAR.
Step 3 From the SIP Dial Rules drop-down list, choose the dial rule that you created for PLAR.
Step 4 Click Save.

Troubleshooting Private Line Automatic Ringdown on SCCP Phones
Symptom Solution
The phone goes off hook Make sure that the CSS that is assigned to the PLAR translation pattern contains
and the user hears a fast the partition of the PLAR destination.
busy (reorder) tone.
The phone goes off hook Make sure that the CSS that is assigned to the phone contains the partition of
and receives dial tone. the null PLAR translation pattern.
Troubleshooting Private Line Automatic Ringdown on SIP Phones
Symptom Solution
The phone goes off hook Make sure that the CSS of the SIP phone can reach the PLAR destination.
and the user hears fast busy
(reorder) tone.
The phone goes off hook Make sure that the SIP Dial Rule has been created and is assigned to the phone.
and receives a dial tone.
567
568
CHAPTER 45
Secure Tone
• Secure Tone Overview, on page 569
• Secure Tone Prerequisites, on page 570
• Secure Tone Configuration Task Flow, on page 570
• Secure Tone Interactions, on page 573
• Secure Tone Restrictions, on page 573
Secure Tone Overview

The Secure Tone feature can configure a phone to play a secure indication tone when a call is encrypted. The
tone indicates that the call is protected and that confidential information may be exchanged. The 2-second
tone comprises three long beeps. If the call is protected, the tone begins to play on a protected phone as soon
as the called party answers.
When the call is not protected, the system plays a nonsecure indication tone, which comprises six short beeps,
on a protected phone.
Note Only callers on protected phones can hear secure and nonsecure indication tones. Callers on phones that are
not protected cannot hear these tones.
The secure and nonsecure indication tones are supported on the following types of calls:
• Intracluster to IP-to-IP calls
• Intercluster protected calls
• IP-to-Time-Division-Multiplexing (TDM) calls through a protected MGCP E1 PRI gateway
For video calls, the system plays secure and nonsecure indication tones on protected devices.
Note For video calls, the user may first hear secure indication tone for the audio portion of the call and then nonsecure
indication tone for overall nonsecure media.
A lock icon that is displayed on a Cisco Unified IP Phone indicates that the media are encrypted, but does not
indicate that the phone has been configured as a protected device. However, the lock icon must be present for
a protected call to occur.
569
Protected Device Gateways
Protected Device Gateways

You can configure only supported Cisco Unified IP Phones and MGCP E1 PRI gateways as protected devices
in Cisco Unified Communications Manager.
Cisco Unified Communications Manager can also direct an MGCP Cisco IOS gateway to play secure and
nonsecure indication tones when the system determines the protected status of a call.
Protected devices provide these functions:
• You can configure phones that are running SCCP or SIP as protected devices.
• Protected devices can call nonprotected devices that are either encrypted or nonencrypted. In such cases,
the call specifies nonprotected and the system plays nonsecure indication tone to the phones on the call.
• When a protected phone calls another protected phone, but the media is not encrypted, the system plays
a nonsecure indication tone to the phones on the call.
Secure Tone Prerequisites

• You must configure the MGCP gateway for SRTP encryption. Configure the gateway with this command:
mgcp package-capability srtp-package.
• The MGCP gateway must specify an Advanced IP Services or Advanced Enterprise Services image (for
example, c3745-adventerprisek9-mz.124-6.T.bin).
Secure Tone Configuration Task Flow

Before you begin
• Review Secure Tone Prerequisites, on page 570
Procedure

support the Secure Tone feature.
Step 2 Configure Phone As a Protected Device, on Configure the phone as a protected device.
page 571
Step 3 Configure Directory Number for Secure Tones, Configure multiple calls and call waiting
on page 571 settings for the protected device.
Step 4 Configure Secure Tone Service Parameters, on Configure service parameters.
page 572
Step 5 (Optional) Configure MGCP E1 PRI Gateway, This configuration allows the system to pass
on page 572 protected status of the call between Cisco
Unified IP Phone endpoints and the protected
PBX phones that connect to the MGCP
gateway.
570
Configure Phone As a Protected Device
Configure Phone As a Protected Device

Before you begin
Procedure

Step 2 Click the phone for which you want to set secure tone parameters.
Step 3 From the Softkey Template drop-down list in the Device Information portion of the window, choose Standard
Protected Phone.
Note You must use a new softkey template without supplementary service softkeys for a protected
phone.
Step 4 Set the Join Across Lines option to Off.

Step 5 Check the Protected Device check box.
Step 6 From the Device Security Profile drop-down list (in the Protocol Specific Information portion of the window),
choose a secure phone profile that is already configured in the Phone Security Profile Configuration window
(System > Security Profile > Phone Security Profile.
Step 7 Click Save.
What to do next
• Configure Directory Number for Secure Tones, on page 571
• Configure MGCP E1 PRI Gateway, on page 572
Configure Directory Number for Secure Tones

Before you begin
Configure Phone As a Protected Device, on page 571
Procedure
Step 1 Locate the Association section on the Phone Configuration window.

Step 2 Select Add a new DN.
Step 3 Specify a directory number in the Directory Number field.
571
Configure Secure Tone Service Parameters
Step 4 In the Multiple Call/Call Waiting Settings on Device [device name] area of the Directory Number
Configuration window, set the Maximum Number of Calls and Busy Trigger options to 1.
Step 5 Configure the remaining fields in the Directory Number Configuration window. For more information on
Step 6 Click Save.
Configure Secure Tone Service Parameters

Procedure
Step 1 In Cisco Unified Communications Manager Administration, choose System > Service Parameters.
Step 3 From the Service drop-down list, choose Cisco CallManager .
Step 4 In the Clusterwide Parameters (Feature - Secure Tone) area, set the Play Tone to Indicate
Secure/Non-Secure Call Status option to True.
Step 5 Click Save.
Configure MGCP E1 PRI Gateway

If you want the system to pass the protected status of the call between Cisco Unified IP Phone endpoints and
the protected PBX phones that connect to the MGCP gateway, follow these steps:
Before you begin

Configure Phone As a Protected Device, on page 571
Procedure
Step 1 In Cisco Unified Communications Manager Administration, choose Device > Gateway.
Step 2 Specify the appropriate search criteria and click Find.
Step 3 Choose a MGCP gateway.
The Gateway Configuration window appears.
Step 4 Set Global ISDN Switch Type to Euro.
Step 5 Configure the fields in the Gateway Configuration window. See the online help for more information about
the fields and their configuration options.
Step 6 Click Save.
Step 7 Click the Endpoint icon that appears to the right of subunit 0 in the window. The Enable Protected Facility
IE check box appears. Check this check box.
572
Secure Tone Interactions
Secure Tone Interactions

Feature Interaction
Call Transfer, Conference, and Call Waiting When the user invokes these features on a protected
phone, the system plays a secure or nonsecure
indication tone to indicate the updated status of the
call.
Hold/Resume and Call Forward All These features are supported on protected calls.
Secure Tone Restrictions

Cisco Extension Mobility and Join Across Line Cisco Extension Mobility and Join Across Line
services services are disabled on protected phones.
Shared-line configuration Shared-line configuration is not available on protected

phones.
Non-encrypted media If the media between the Cisco Unified IP Phone and
the MGCP E1 PRI gateway are not encrypted, the call
drops.
573
Secure Tone Restrictions
574
PA R T XII
Custom Features
• Branding Customizations , on page 577
• Client Matter Codes and Forced Authorization Codes, on page 585
• Custom Phone Rings and Backgrounds, on page 591
• Music On Hold , on page 599
• Self Care Portal , on page 617
• Emergency Call Handler , on page 621
• Enterprise Groups , on page 635
CHAPTER 46
Branding Customizations
• Branding Overview, on page 577
• Branding Prerequisites, on page 577
• Branding Task Flow, on page 578
• Branding File Requirements, on page 580
Branding Overview
The Branding feature lets you upload customized branding for Cisco Unified Communications Manager.
Branding gets applied to the Cisco Unified CM Administration login and configuration windows. Among the
items that you can modify include:
• Company logos
• Background colors
• Border colors
• Font colors
Append Logo in Self Care Portal

The Branding feature allows you to append your company logo to the Unified Communications Self Care
Portal login page and to the user interface header. You must include the branding_logo.png file in your
branding.zip file and upload the zip file into Cisco Unified Communications Manager. The logo displays
in the Self Care Portal after you enable branding in Cisco Unified Communications Manager.
There is no option to customize background colors or fonts for the Self-Care portal.
Branding Prerequisites
You must create your branding.zip file that contains the specified folder structure and files. For details,
see Branding File Requirements, on page 580.
577
Custom Features
Branding Task Flow
Branding Task Flow

Complete these tasks to apply branding in Cisco Unified Communications Manager and the Unified
Communications Self-Care Portal.
Before you begin

• Review Branding Prerequisites, on page 577
Procedure

Step 1 Configure your branding settings using one of Apply branding across the Cisco Unified
these procedures: Communications Manager cluster.
• Enable Branding, on page 578
• Disable Branding , on page 579
Step 2 Restart the Tomcat Service, on page 580 You must restart the Cisco Tomcat service for
the new branding setting to get picked up by
the Unified Communications Self-Care Portal.
Enable Branding
Use this procedure to enable branding customization for Unified Communications Manager. Branding updates
appear even if the system is enabled for SAML Single Sign-On.
Note To enable branding, you must use the primary administrator account with privilege level 4 access. This is the
main administrator account that is created during installation.
Note Ensure that you use only one among GUI and CLI to enable branding as well as to disable it. For example, if
you enable branding using the GUI interface, you must use the GUI interface itself to disable branding. Else,
it will not function properly.
Before you begin

Prepare your branding.zip file and save it in a location that Unified Communications Manager can access.
Procedure
Step 1 Log in to Cisco Unified OS Administration.

Step 2 Choose Software Upgrades > Branding.
578
Custom Features
Disable Branding
Step 3 Browse to your remote server and select the branding.zip file.
Step 5 Click Enable Branding.
Note You can also enable branding by running the utils branding enable CLI command.
Step 6 Refresh your browser.

Step 7 Repeat this procedure on all Cisco Unified Communications Manager cluster nodes.
If you want to append your company logo to the Self-Care Portal user interface, see Restart the Tomcat Service,
on page 580.
Disable Branding
Use this procedure to disable branding in your Cisco Unified Communications Manager cluster. You also
need to disable branding if you want to remove your company logo from the Self-Care Portal.
Note To disable branding, you must use the primary administrator account with privilege level 4 access. This is
the main administrator account that is created during installation.
Note Ensure that you use only one among GUI and CLI to enable branding as well as to disable it. For example, if
you enable branding using the GUI interface, you must use the GUI interface itself to disable branding. Else,
it will not function properly.
Procedure
Step 1 Log in to Cisco Unified OS Administration.

Step 2 Choose Software Upgrades > Branding.
Step 3 Click Disable Branding.
Note You can also disable branding by running the utils branding disable CLI command.
Step 4 Refresh your browser.

If you want to remove your company logo from the Self-Care Portal user interface, see Restart the Tomcat
Service, on page 580
579
Custom Features
Restart the Tomcat Service
Restart the Tomcat Service

You must restart the Cisco Tomcat service for the branding updates to reflect in the Self-Care Portal.
Before you begin

Make sure that you have completed the following:
• To append your logo to the Self-Care Portal, you must first enable branding in Cisco Unified
Communications Manager. The branding.zip upload file must include a 44x25 pixel
branding_logo.png file with your company logo. For details, Enable Branding, on page 578.
• To remove your logo from the Self-Care Portal, you must disable branding in Cisco Unified
Communications Manager. For details, Disable Branding , on page 579.
Procedure
Step 1 Log in to the Command Line Interface.

Step 2 Run the utils service restart Cisco Tomcat CLI command.
What to do next
After the service restarts, refresh your browser to see the changes in the Self-Care Portal.
Branding File Requirements

Before you apply customized branding to your system, create your branding.zip file according to the
prescribed specifications. On a remote server, create a Branding folder and fill the folder with the specified
contents. Once you have added all the image files and subfolders, zip the entire folder and save the file as
branding.zip.
There are two options for the folder structure, depending on whether you want to use a single image for the
header or a combination of six images in order to create a graded effect for the header.
Table 51: Folder Structure Options
Branding Option Folder Structure
Single Header Option If you want a single image for the header background (callout item 3), your
branding folder must contain the following subfolders and image files:
Branding (folder)
ccmadmin (folder)
BrandingProperties.properties (properties file)
brandingHeader.gif (2048*1 pixel image)
ciscoLogo12pxMargin.gif (44*44 pixel image)
branding_logo.png (44*25 pixel image)
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Custom Features
Branding Option Folder Structure
Graded Header Option If you want to create a graded image for the header background, you need six
separate image files to create the graded effect. Your branding folder must contain
these subfolders and files
Branding(folder)
ccmadmin (folder)
BrandingProperties.properties (file)
brandingHeaderBegLTR.gif (652*1 pixel image)
brandingHeaderBegRTR.gif (652*1 pixel image)
brandingHeaderEndLTR.gif (652*1 pixel image)
brandingHeaderEndRTR.gif (652*1 pixel image)
brandingHeaderMidLTR.gif (652*1 pixel image)
brandingHeaderMidRTR.gif (652*1 pixel image)
ciscoLogo12pxMargin.gif (44*44 pixel image)
branding_logo.png (44*25 pixel image)
User Interface Branding Options

The following images display the customization options for the Cisco Unified CM Administration user
interface:
Figure 10: Branding Options for Unified CM Administration Login Screen
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Custom Features
Figure 11: Branding Options for Unified CM Administration Logged In Screen
The following table describes the callout options.
Table 52: User Interface Branding Options: Login Screen
Item Description Branding edits
1 Company Logo To add your logo to Cisco Unified Communications

Manager, save your company logo as a 44x44 pixel
image with the following filename:
ciscoLogo12pxMargin.gif (44*44 pixels)
Note If you also want to append your logo to
the header and login screen of the
Self-Care Portal, you should also save
your logo as the 44x25 pixel
branding_logo.png file.
2 Unified CM Administration header font heading.heading.color

color
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Custom Features
3 Header Background You can use a single image or a combination of six

images to create a grading effect.
Single Image option—Save your header background
as a single image:
• brandingHeader.gif (2048*1 pixel)
Graded background option:—Save your header

background as six images for a graded effect:
• brandingHeaderBegLTR.gif (652*1 pixel)
• brandingHeaderBegRTR.gif (652*1 pixel)
• brandingHeaderEndLTR.gif (652*1 pixel)
• brandingHeaderEndRTR.gif (652*1 pixel)
• brandingHeaderMidLTR.gif (652*1 pixel)
• brandingHeaderMidRTR.gif (652*1 pixel)
4 Navigation text header.navigation.color
5 Go button header.go.font.color
header.go.background.color
header.go.border.color
6 Username text splash.username.color
7 Password text splash.password.color
8 Login button splash.login.text.color

splash.login.back.ground.color
9 Reset button splash.reset.text.color

splash.reset.back.ground.color
10 Bottom background color – right splash.hex.code.3
11 Bottom background color – left splash.hex.code.2
12 Banner splash.hex.code.1
Table 53: User Interface Branding Options: Logged In Screen
13 User text (for example, 'admin') header.admin.color
14 Search, About and Login text header.hover.link.color
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Custom Features
15 Unified CM Administration text heading splash.header.color
16 System Version, VMware Installation splash.reset.text.color

text
splash.version.color
Branding Properties Editing Example

Branding properties can be edited by adding the hex code in the properties file
(BrandingProperties.properties). The properties file uses HTML-based hex code. For example,
if you want to change the color of the Navigation text item (callout item #4) to red, add the following code
to your properties file:
header.navigation.color="#FF0000"
In this code, header.navigation.color is the branding property that you want to edit, and "#FF0000" is the
new setting (red).
584
CHAPTER 47
Client Matter Codes and Forced Authorization
Codes
• Client Matter Codes and Forced Authorization Codes Overview, on page 585
• Client Matter Codes and Forced Authorization Codes Prerequisites, on page 585
• Client Matter Codes and Forced Authorization Codes Configuration Task Flow, on page 586
• Client Matter Codes and Forced Authorization Codes Interactions, on page 589
• Client Matter Codes and Forced Authorization Codes Restrictions, on page 590
Client Matter Codes and Forced Authorization Codes Overview

With client matter codes (CMCs) and forced authorization codes (FACs), you can effectively manage call
access and accounting. CMCs assist with call accounting and billing for clients, and FACs regulate the types
of calls that certain users can place.
CMCs force the user to enter a code; this action specifies that the call relates to a specific client matter. You
can assign client matter codes to customers, students, or other populations for call accounting and billing
purposes. FACs force the user to enter a valid authorization code that is assigned at a certain access level
before the call is completed.
Client Matter Codes and Forced Authorization Codes

Prerequisites
• Cisco Unified IP Phones that are running SCCP and SIP support CMC and FAC.
• The CMC and FAC tones play only on Cisco Unified IP Phones that are running SCCP or SIP;
TAPI/JTAPI ports; and MGCP FXS ports.
585
Custom Features
Client Matter Codes and Forced Authorization Codes Configuration Task Flow
Client Matter Codes and Forced Authorization Codes

Configuration Task Flow
You can implement CMCs and FACs separately or together. For example, you may authorize users to place
certain classes of calls, such as long distance calls, and also assign the class of calls to a specific client. CMC
and FAC tones sound the same to the user; if you configure both codes, the feature prompts the user to enter
the FAC after the first tone and enter the CMC after the second tone.
Before you begin

• Review Client Matter Codes and Forced Authorization Codes Prerequisites, on page 585
Procedure

Step 1 To Configure Client Matter Codes, on page 586, After you finalize the list of CMCs that you
complete the following subtasks: plan to use, add those codes to the database and
enable the CMC feature in route patterns.
• Add Client Matter Codes, on page 587
• Enable Client Matter Codes, on page 587
Step 2 To Configure Forced Authorization Codes, on After you finalize the list of FACs and
page 587, complete the following subtasks: authorization levels that you plan to use, add
those codes to the database and enable the FAC
• Add Forced Authorization Codes, on page
feature in route patterns.
588
• Enable Forced Authorization Codes, on
page 588
Configure Client Matter Codes

Procedure

Step 1 Add Client Matter Codes, on page 587 Determine unique client matter codes that you
want to use and add them to your system.
Because the number of CMCs directly affects
the time that is required for your system to start
up, limit the number of CMCs to a maximum
of 60,000. If you configure more CMCs than
the maximum number, expect significant delays.
Step 2 Enable Client Matter Codes, on page 587 Enable client matter codes through a route
pattern.
586
Custom Features
Add Client Matter Codes
Add Client Matter Codes

Determine unique client matter codes that you want to use and add them to your system. Because the number
of CMCs directly affects the time that is required for your system to start up, limit the number of CMCs to a
maximum of 60,000. If you configure more CMCs than the maximum number, expect significant delays.
Procedure
Step 1 From Cisco Unified CM Administration, choose Call Routing > Client Matter Codes.
Step 3 In the Client Matter Code field, enter a unique code of no more than 16 digits that the user will enter when
placing a call.
Step 4 In the Description field, enter a client name if you want to identify the client matter code.
Step 5 Click Save.
Enable Client Matter Codes

Enable client matter codes through a route pattern.
Before you begin

Add Client Matter Codes, on page 587
Procedure
• To update an existing route pattern, enter search criteria, click Find, and choose a route pattern from the
resulting list.
• To create a new route pattern, click Add New.
Step 3 In the Route Pattern Configuration window, check the Require Client Matter Code check box.
Step 4 Click Save.
Configure Forced Authorization Codes

Procedure

Step 1 Add Forced Authorization Codes, on page 588 Determine unique forced authorization codes
that you want to use and add them to your
system.
587
Custom Features
Add Forced Authorization Codes

Step 2 Enable Forced Authorization Codes, on page Enable forced authorization codes through a
588 route pattern.
Add Forced Authorization Codes

Use this procedure to determine unique forced authorization codes that you want to use and add them to your
system. To successfully route a call, the user authorization level must be equal to or greater than the
authorization level that is specified for the route pattern for the call.
Procedure
Step 1 From Cisco Unified CM Administration, choose Call Routing > Forced Authorization Codes.
Step 2 In the Authorization Code Name field, enter a unique name that is no more than 50 characters.
This name ties the authorization code to a specific user or group of users.
Step 3 In the Authorization Code field, enter a unique authorization code that is no more than 16 digits.
Users enter this code when they place a call through an FAC-enabled route pattern.
Step 4 In the Authorization Level field, enter a three-digit authorization level in the range of 0 to 255.
Step 5 Click Save.
Enable Forced Authorization Codes

Use this procedure to enable forced authorization codes through a route pattern.
Procedure
• Click Find, and then choose a route pattern from the resulting list to update an existing route pattern.
• Click Add New to create a new route pattern.
Step 3 In the Route Pattern Configuration window, check the Require Forced Authorization Code check box.
Step 4 In the Authorization Level field, enter the authorization level value between 0 and 255.
The FAC level for the user must be greater than or equal to the configured level for the call to route successfully.
Step 5 Click Save.
588
Custom Features
Client Matter Codes and Forced Authorization Codes Interactions
Client Matter Codes and Forced Authorization Codes Interactions

Table 54: Client Matter Codes and Forced Authorization Codes Interactions
Feature Interaction
CDR Analysis and CDR Analysis and Reporting (CAR) allows you to run reports that provide call details
Reporting (CAR) for client matter codes (CMCs), forced authorization codes (FACs), and authorization
levels.
CTI, JTAPI, and In most cases, your system can alert a CTI, JTAPI, or TAPI application that the user
TAPI applications must enter a code during a call. When a user places a call, creates an ad hoc conference,
or performs a consult transfer through a CMC- or FAC-enabled route pattern, the
user must enter a code after receiving the tone.
When a user redirects or blind transfers a call through a CMC- or FAC-enabled route
pattern, the user receives no tone, so the application must send the codes to Cisco
Unified Communications Manager. If your system receives the appropriate codes,
the call connects to the intended party. If your system does not receive the appropriate
codes, Cisco Unified Communications Manager sends an error to the application that
indicates which code is missing.
Cisco Web Dialer Web Dialer supports CMCs and FACs in the following ways:
• A user can enter the destination number in the dial text box of the WD HTML
page or SOAP request, and then manually enter the CMC or FAC on the phone.
• A user can enter the destination number followed by the FAC or CMC in the
dial text box of the WD HTML page or SOAP request.
For example, if the destination number is 5555, the FAC is 111, and the CMC is 222,
a user can make a call by dialing 5555111# (FAC), 5555222# (CMC), or 5555111222#
(CMC and FAC).
Note • WebDialer does not handle any validation for the destination
number. The phone handles the required validation.
• If a user does not provide a code or provides the wrong code, the
call will fail.
• If a user makes a call from the WebApp with a DN that contains
special characters, the call goes successfully after stripping the
special characters. The same rules do not work in SOAP UI.
Speed Dial and You can use speed dial to reach destinations that require a FAC, CMC, dialing pauses,
Abbreviated Speed or additional digits (such as a user extension, a meeting access code, or a voicemail
Dial password). When the user presses the configured speed dial, the phone establishes
the call to the destination number and sends the specified FAC, CMC, and additional
digits with dialing pauses inserted.
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Custom Features
Client Matter Codes and Forced Authorization Codes Restrictions
Client Matter Codes and Forced Authorization Codes Restrictions

Table 55: Client Matter Codes and Forced Authorization Codes Restrictions
Analog gateways H.323 analog gateways do not support CMCs or FACs because these gateways cannot
play tones.
Call forwarding Calls that are forwarded to a CMC- or FAC-enabled route pattern fail because no
user is present to enter the code. When a user presses the CFwdALL softkey and
enters a number that has CMC or FAC enabled on the route pattern, call forwarding
fails.
To minimize call-processing interruptions, test the number before you configure call
forwarding. To do this, dial the intended forwarding number; if you are prompted for
a code, do not configure call forwarding for that number. Advise users of this practice
to reduce the number of complaints that result from forwarded calls that do not reach
the intended destination.
Cisco Unified Calls that originate from a SIP trunk, H.323 gateway, or MGCP gateway fail if they
Mobility encounter a route pattern that requires CMCs or FACs and the caller is not configured
with Cisco Unified Mobility.
Dial via Office The CMC and FAC feature on Cisco Mobility does not support an alternative number
callback number as its dial via office (DVO) callback number. The DVO callback number must be the
number that is registered on the Mobility Identity window.
Failover calls CMCs and FACs do not work with failover calls.
Hearing-impaired After dialing the phone number, hearing-impaired users should wait one or two
users seconds before entering the authorization or client matter code.
Localization Cisco does not localize CMCs or FACs. The CMC and FAC features use the same
default tone for any locale that is supported with Cisco Unified Communications
Manager.
Note For Cisco Mobility, CMCs and FACs are localized.
Overlap sending The CMC and FAC features do not support overlap sending because Cisco Unified
Communications Manager cannot determine when to prompt the user for the code.
If you check the Require Forced Authorization Code or the Require Client Matter
Code check box in the Route Pattern Configuration window, the Allow Overlap
Sending check box is automatically unchecked and vice-versa.
Speed-dial buttons You cannot configure CMCs or FACs for speed-dial buttons. You must enter the
code when the system prompts you to do so.
590
CHAPTER 48
Custom Phone Rings and Backgrounds
• Custom Phone Rings Overview, on page 591
• Custom Phone Rings Prerequisites, on page 591
• Custom Phone Rings Configuration Task Flow, on page 592
• Custom Backgrounds, on page 594
• Custom Backgrounds Configuration Task Flow, on page 594
Custom Phone Rings Overview

Custom Phone Rings allows you to create customized phone rings and upload the customized files to the
Cisco Unified Communications Manager TFTP server where they can be accessed by Cisco Unified IP Phones.
Cisco Unified IP Phones ship with default ring types that are implemented in hardware: Chirp1 and Chirp2.
In addition, Cisco Unified Communications Manager provides the capability of uploading the following files
to phones:
PCM Files
Cisco Unified Communications Manager provides a default set of phone ring sounds that are implemented
in software as pulse code modulation (PCM) audio files. Each PCM file specifies a single ring type.
Ringlist.xml File
The Ringlist.xml file describes the list of ring options that are available for phones.
You can upload customized PCM audio files, such as custom ring tones and call back tones, as well as the
modified Ringlist.xml file to the TFTP directory in Cisco Unified Communications Manager.
Custom Phone Rings Prerequisites

The following prerequisites apply to Custom Phone Rings:
• In order to upload your custom phone rings, the Cisco TFTP service must be running.
• Any PCM files that you want to upload must meet a set of file requirements in order to be compatible
with Cisco Unified IP Phones. For details, review the topic PCM File Format Requirements, on page
593.
• The Ringlist.xml file must meet a set of formatting guidelines. For details, review the topic Ringlist.xml
File Format Requirements, on page 593.
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Custom Features
Custom Phone Rings Configuration Task Flow
Custom Phone Rings Configuration Task Flow

Before you begin
• Review Custom Phone Rings Prerequisites, on page 591
Procedure

Step 1 Prepare Custom Phone Rings for Upload, on Create your customized PCM and Ringlist.xml
page 592 files.
Step 2 Upload Custom Phone Rings to TFTP Server, Upload customized files to the Cisco Unified
on page 592 Communications Manager TFTP server.
Step 3 Restart TFTP Service, on page 593 After the upload completes, restart the Cisco
TFTP service.
Prepare Custom Phone Rings for Upload

Procedure
Step 1 Use the file get tftp <tftp path> CLI command to download the existing Ringlist.xml file, in addition
to any PCM files that you want to modify.
Step 2 Create a PCM file for each ring type that you want to upload. For guidelines on PCM file compatibility with
Cisco Unified Communications Manager, see PCM File Format Requirements, on page 593.
Step 3 Use an ASCII editor to update the Ringlist.xml file with your new phone rings. For details on Ringlist.xml
file formatting requirements, see Ringlist.xml File Format Requirements, on page 593.
Upload Custom Phone Rings to TFTP Server

Before you begin
Prepare Custom Phone Rings for Upload, on page 592
Procedure
Step 1 From Cisco Unified OS Administration, choose Software Upgrades > TFTP > File Management.
Step 3 Click Browse and select the Ringlist.xml file, as well as any PCM files that you want to upload.
592
Custom Features
Restart TFTP Service
Restart TFTP Service

Before you begin
Upload Custom Phone Rings to TFTP Server, on page 592
Procedure
Step 1 Log in to Cisco Unified Serviceability and choose Tools > Control Center - Feature Services.
Step 2 From the Server drop-down list, choose the server on which the Cisco TFTP service is running.
Step 3 Click the radio button that corresponds to the Cisco TFTP service.
PCM File Format Requirements

PCM files for phone rings must meet a set of requirements for proper playback on Cisco Unified IP Phones.
When creating or modifying your PCM files, you can use any standard audio editing packages that support
the following file format requirements:
• Raw PCM
• 8000 samples per second
• 8 bits per sample
• mu-law compression
• Maximum ring size: 16080 samples
• Number of samples in the ring must be evenly divisible by 240
• Ring starts and ends at the zero crossing
Ringlist.xml File Format Requirements

The Ringlist.xml file defines an XML object that contains a list of phone ring types. Each ring type contains
a pointer to the PCM file that is used for that ring type and the text that will display on the Ring Type menu
on a Cisco Unified IP Phone for that ring.
The CiscoIPPhoneRinglist XML object uses the following simple tag set to describe the information:
<CiscoIPPhoneRinglist> <Ring>
<DisplayName/>
<FileName/>
</Ring>
</CiscoIPPhoneRinglist>
The following characteristics apply to the definition names:

• DisplayName defines the name of the custom ring for the associated PCM file that will display on the
Ring Type menu of the Cisco Unified IP Phone.
593
Custom Features
Custom Backgrounds
• FileName specifies the name of the PCM file for the custom ring to associate with DisplayName.
Tip The DisplayName and FileName fields must not exceed 25 characters.
The following example shows a Ringlist.xml file that defines two phone ring types:
<CiscoIPPhoneRinglist> <Ring>
<DisplayName>Analog Synth 1</DisplayName>
<FileName>Analog1.raw</FileName>
</Ring>
<Ring>
<DisplayName>Analog Synth 2</DisplayName>
<FileName>Analog2.raw</FileName>
</Ring>
</CiscoIPPhoneRinglist>
Tip You must include the required DisplayName and FileName for each phone ring type. The Ringlist.xml file
can include up to 50 ring types.
Custom Backgrounds
You can also use the TFTP server to upload new custom background images to the phones in your network.
Phone users can select the images that you upload as their phone backgrounds. You can configure your system
so that phone users can select from an assortment of images or you can assign a specific background image
for all phone.
If you want your phone users to be able to customize their phone backgrounds, then you must prepare and
upload the following files to the TFTP server whenever you upload new images:
• Full-size background image—Refer to your phone documentation for image specifications, including
the image size (in pixels) and color-type, for your phone model.
• A thumbnail image—This is only required if you want your phone users to be able to choose their own
background image. Refer to your phone documentation for the thumbnail image specifications
• An edited List.xml file—This file contains a listing of the background images from which phone
users can select. You must add your new images to this file.
If you want to assign a specific image for all phones then you need to upload the main background image
only. In addition, you also must update the Common Phone Profile to direct the phones to use the image that
you assign.
Custom Backgrounds Configuration Task Flow

Complete these tasks to configure and upload customized background images for the phones in your deployment.
You can configure the system so that phone users can select from an assortment of images, or you can assign
a specific background image that displays on all phones.
594
Custom Features
Create Phone Background Images
Procedure

Step 1 Create Phone Background Images, on page 595 Create your full-size background image and
corresponding thumbnail image (if required).
Refer to your phone documentation for image
specifications, including file type, image size
(in pixels) and color-type.
Note The thumbnail is not required if
you are assigning a specific
background image.
Step 2 Edit the List.xml file, on page 596 Update the List.xml file from the appropriate
TFTP directory with your new images. This is
required so that phone users see the new images
in their list of phone background options.
Note This procedure is required only if
you are giving your users the
option to choose their own
background. If you are assigning
a specific background image then
there is no need to edit this file.
Step 3 Upload Backgrounds to TFTP Server, on page Upload your files to the TFTP server.
597
Step 4 Restart the TFTP Server, on page 597 Restart the Cisco TFTP service in order to push
the images to your phones.
Step 5 Assign Phone Background for Phone Users, on Optional. By default, Cisco Unified
page 597 Communications Manager gives phone users
the option to select their own phone background
image. However, you can use the Common
Phone Profile to assign a specific background
image for all phones that use this Common
Phone Profile.
Create Phone Background Images

Refer to your phone documentation for background image specifications and thumbnail image specifications.
This includes the image sizes (in pixels), file type, and the appropriate destination TFTP directory for that
phone model (the TFTP directory is based on the image specifications).
• If you want phone users to have the option to use, or not use, the image that you upload, you must prepare
both a full-size image and a thumbnail image according to the specifications for that particular phone
model.
• If you want to assign the image to specific phones, you need the full-size image only.
595
Custom Features
Edit the List.xml file
What to do next
If you want phone users to be able to choose their own background image, Edit the List.xml file, on page 596.
If you want to assign a specific background image, you don't need to update the List.xml file. Proceed to
Upload Backgrounds to TFTP Server, on page 597
Edit the List.xml file

If you want phone users to be able to choose their background images, use this procedure to add any new
background images that you upload to the existing List.xml file. Each TFTP image directory contains a
List.xml file that gets used by the phones that use that TFTP directory. This file points to the specific
background and thumbnail image for each background option and can include up to 50 background images.
The images are listed using the order in which they appear on the phone. For each image, the file contains an
<ImageItem> element that includes these two attributes:
• Image: Uniform resource identifier (URI) that specifies where the phone obtains the thumbnail image
that will appear on the Background Images menu of a phone.
• URL: URI that specifies where the phone obtains the full size image.
Example:
The following example (for a Cisco Unified IP Phone 7971G-GE and 7970G) shows a List.xml file that
defines two images. The required Image and URL attributes must be included for each image. The TFTP URI
that displays in the example is the only supported method for linking to full size and thumbnail images as
HTTP URL support is not provided.
<CiscoIPPhoneImageList>
<ImageItem Image=”TFTP:Desktops/320x212x12/TN-Fountain.png”
URL=”TFTP:Desktops/320x212x12/Fountain.png”/>
<ImageItem Image=”TFTP:Desktops/320x212x12/TN-FullMoon.png”
URL=”TFTP:Desktops/320x212x12/FullMoon.png”/>
</CiscoIPPhoneImageList
Procedure
Step 1 Log in to the Command Line Interface

Step 2 Run the file get tftp <filename> CLI command where <filename> represents the file and filepath of the
List.xml file for the appropriate TFTP directory.
Note Make sure that you download the List.xml file from the appropriate TFTP directory as each
image directory has its own file. Refer to your phone documentation for information on the
appropriate TFTP directory for that phone model as the directory is based on the image
specifications.
Step 3 Edit the xml file with a new <ImageItem> element for each new background option that you want to add.
596
Custom Features
Upload Backgrounds to TFTP Server
Upload Backgrounds to TFTP Server

Use this procedure to upload new phone background files to the TFTP server.
• If you want your phone users to be able to choose their own background image, then you must upload
your full-size background image, a thumbnail image, and the updated List.xml file.
• If you are assigning a specific background image, you need to upload the full-size background image
only.
Procedure
Step 1 From Cisco Unified OS Administration, choose Software Upgrades > TFTP File Management
Step 2 Click Upload File and do the following:
a) Click Choose File and select the background file that you want to upload.
b) In the Directory field, enter the appropriate TFTP directory for that phone model. The TFTP directory
corresponds to the image size and color type. Refer to your phone documentation for images specification.
c) Click Upload File
d) Repeat these steps to upload both the thumbnail image and list.xml files as well. These files should
be loaded to the same TFTP directory as the main background image.
Step 3 Click Close.
Restart the TFTP Server

After you have uploaded your custom files to the TFTP directory, restart the Cisco TFTP server to push the
files to the phones.
Procedure
Step 1 Log in to Cisco Unified Serviceability and choose Tools > Control Center - Feature Services.
Step 2 From the Server drop-down list, choose the server on which the Cisco TFTP service is running.
Step 3 Click the radio button that corresponds to the Cisco TFTP service.
Assign Phone Background for Phone Users

By default, Cisco Unified Communications Manager allows phone users to customize their own phone
background image. However, you can use the Common Phone Profile setting to assign a specific background
image for all phones that use this Common Phone Profile.
597
Custom Features
Assign Phone Background for Phone Users
Procedure
Step 2 Do one of the following:
• Click Find and select the Common Phone Profile that your phones use.
• Click Add New to create a new Common Phone Profile.
Step 3 If you want users to be able to choose their background image, make sure that the Enable End User Access
to Phone Background Image Setting check box is checked (this is the default setting).
Step 4 If you want to assign a specific background image for phones that use this profile:
• Uncheck the Enable End User Access to Phone Background Image Setting check box.
• In the Background Image text box, enter the filename of the image file that you want to assign. Also,
check the Override Enterprise Settings check box that corresponds to this text box.
Step 5 Complete any remaining fields in the Common Phone Profile window. For help with the fields and their
settings, refer to the online help.
Step 6 Click Save.
If you have assigned a specific background image, all phones that use this Common Phone Profile will use
the image that you specify.
What to do next
If you have created a new Common Phone Profile, reconfigure your phones so that they use this profile. For
details on how to configure phones in Cisco Unified Communications Manager, see the "Configure Endpoint
Devices" section of the System Configuration Guide for Cisco Unified Communications Manager.
Tip If you have a large number of phones to assign, use the Bulk Administration Tool to assign a Common Phone
Profile to a large number of phones in a single operation. For details, see the Bulk Administration Guide for
Note After you complete your configuration, reset your phones.
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CHAPTER 49
Music On Hold
• Music On Hold Overview, on page 599
• Interwork External Multicast MOH to Unicast MOH, on page 603
• Music On Hold Prerequisites, on page 604
• Music On Hold Configuration Task Flow, on page 604
• Unicast and Multicast Audio Sources, on page 611
• Music On Hold Interactions, on page 613
• Music On Hold Restrictions, on page 614
• Music On Hold Troubleshooting, on page 616
Music On Hold Overview

Use the integrated Music On Hold (MOH) feature to place on-net and off-net users on hold with music from
a streaming source. This source makes music available to any on-net or off-net device that you place on hold.
On-net devices include station devices and applications that an interactive voice response (IVR) or call
distributor places on hold, consult hold, or park hold. Off-net users include those users who are connected
through Media Gateway Control Protocol (MGCP) or Skinny Call Control Protocol (SCCP) gateways, Cisco
IOS H.323 gateways, and Cisco IOS Media Gateway Control Protocol gateways. The system also makes the
Music On Hold feature available for Cisco IP POTS phones that connect to the Cisco IP network through
Foreign Exchange Station (FXS) ports on Cisco IOS H.323 or MGCP and for Cisco MGCP or SCCP gateways.
Start Cisco Unified Communications Manager to create a media resource manager. Music On Hold server
registers to the media resource manager with its music on hold resources. Music On Hold server is a software
application that provides music on hold audio sources and connects a music on hold audio source to multiple
streams.
When an end device or feature places a call on hold, Cisco Unified Communications Manager connects the
held device to a music resource. When the held device is retrieved, it disconnects from the music on hold
resource and resumes normal activity.
Caller-Specific Music On Hold

For SIP calls that a phone receives over the SIP trunk, Cisco Unified Communications Manager can use a
different MOH audio source.
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Custom Features
Increased Capacity of IP Voice Media Streaming Application and Expanded MOH Audio Source
An external application, such as the Cisco Unified Customer Voice Portal (CVP) contact center solution,
determines the most appropriate MOH audio source based on the caller ID, dialed number, or IVR interaction
when a call is received from the public switched telephone network (PSTN).
For details, see the Cisco Unified Customer Voice Portal documentation at http://www.cisco.com/c/en/us/
support/customer-collaboration/unified-customer-voice-portal/tsd-products-support-series-home.html.
Increased Capacity of IP Voice Media Streaming Application and Expanded

MOH Audio Source
Cisco IP Voice Media Streaming application is installed automatically when you install Cisco Unified
Communications Manager. Activate this application to enable the Music On Hold (MOH) feature.
With this release, the capacity of Cisco Unified Communications Manager to support unique and concurrent
MOH audio sources, while the Music On Hold service is running on the MOH server, is increased from 51
to 501. The MOH audio sources are numbered from 1 to 501 with the fixed MOH audio source remaining at
the number 51.
The fixed MOH device cannot use an audio source that connects through a USB MOH device, because Cisco
Unified Communications Manager does not support USB when running on VMware. Use of the fixed MOH
USB device is not supported on VMware. However, provision the external sound device for use with
deployments that utilize Cisco Unified Survivable Remote Site Telephony (SRST) multicast MOH.
You can configure each MOH audio source to use a custom announcement as an initial greeting and/or an
announcement that is played periodically to callers who are hearing the music. Cisco Unified Communications
Manager provides 500 custom announcements that you can use on one or multiple MOH audio sources. These
announcements are not distributed between the Cisco Unified Communications Manager servers within a
cluster. You have to upload these custom announcement files to each server that provides the MOH and
announcement services. You must also upload each custom music file for MOH audio sources to each server.
Performance Impact of Media Devices with Services

The Cisco IP Voice Media Streaming application runs as a service for four media devices—annunciator
(ANN), software conference bridge, Music On Hold (MOH), and software media termination point. Activate
this service on a Cisco Unified Communications Manager server as coresident with call processing. When
you activate this service, ensure that you configure these media devices for limited capacity to avoid any
impact on the call processing. The default settings for the media devices are defined based on this coresident
operation. You can adjust these settings by reducing the use of one or more media devices to increase other
settings.
For example, if you are not using software media termination point devices, you can choose the Run Flag
setting for the SW MTP to False, select System > Service Parameters > Cisco IP Voice Media Streaming
App service > MTP Parameters , and add the MTP Call Count setting to Media Resource > MOH Server >
Maximum Half Duplex Streams configuration. Depending on the call traffic, you can modify the default
settings. However, monitor the server performance activity for CPU, memory, and IO wait. For higher capacity
clusters, such as the ones using 7500 user OVA configuration, it is possible to increase the default media
device settings for Call Count by 25%.
For installations where you expect high usage of the media devices, such as Music On Hold, or where high
call volumes require higher number of media connections, activate the Cisco IP Voice Media Streaming
application service on one or more of the Cisco Unified Communications Manager servers which do not have
call processing activated. Activating this service limits the impact of media device usage to other services,
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Performance Impact of Media Devices with Services
such as call processing. Then, you can increase the configuration settings for maximum number of calls for
the media devices.
When you activate Cisco IP Voice Media Streaming application as co-resident with Cisco Unified
Communications Manager service, it can impact call processing performance. To increase the capacity settings
for Music On Hold or annunciator from the default settings, it is suggested to activate Cisco IP Voice Media
Streaming application on a server without activating Cisco Unified Communications Manager.
The CPU performance is impacted by MOH when active callers are on hold or when multicast MOH audio
streams are configured.
Table 56: General Performance Results
Configuration Notes CPU Performance
Dedicated MOH server, 1000 held calls, 500 MOH 25–45% (7500 user OVA configuration)
sources with greeting and periodic announcements.
Native call queuing with dedicated MOH server and 25–45% (7500 user OVA configuration)
annunciator server, 1000 queued calls, 500 MOH
An annunciator can play up to 300 simultaneous
greeting announcements.
Dedicated MOH server, 500 held calls, 500 MOH 15–35% (7500 user OVA configuration)
Table 57: Extrapolated Recommendations
Configuration Recommendation Limit
When Cisco IP Voice Media Streaming application MOH: 500 held callers, 100 MOH sources, and 48 to
is co-resident with Cisco Unified Communications 64 annunciator callers.
Manager on 2500 OVA (moderate call processing).
When Cisco IP Voice Media Streaming application MOH: 750 held callers, 250 MOH sources, and 250
is a dedicated server on 2500 OVA. annunciator callers.
When Cisco IP Voice Media Streaming application MOH: 500 held callers, 250 MOH sources, and 128
is co-resident with Cisco Unified Communications annunciator callers.
Manager on 7500/10K OVA (moderate call
processing).
When Cisco IP Voice Media Streaming application MOH: 1000 held callers, 500 MOH sources, and
is a dedicated server on 7500/10K OVA. 300-700 annunciator callers (with 1 MOH codec).
Note Reduce annunciator to 300 for two
MOH codecs.
Note These recommendations are specific to MOH/ANN devices. If you combine these devices with the software
media termination point (MTP) and call forward busy (CFB) devices, reduce the limits to provide streams.
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Custom Features
Configuration Limitations for Capacity Planning
Configuration Limitations for Capacity Planning

The Cisco IP Voice Media Streaming application and Self Provisioning IVR services use a media kernel
driver to create and control Real-time Transfer Protocol (RTP) streams. This media kernel driver has a capacity
of 6000 streams. These streams allow the media devices and IVR to make resource reservations.
These reservations are based on the following capacity calculations:
Media Device Capacity
Annunciator (Call Count service parameter) * 3

Where 3 indicates total of receiving (RX) and transmitting (TX) calls for
endpoint and 1 for .wav file.
Software Conference Bridge (Call Count service parameter) * 2

Where 2 indicates total streams of RX and TX endpoints.
Software Media Termination (Call Count service parameter) * 2

Point
Where 2 indicates total streams of RX and TX endpoints.
Music On Hold ((Maximum Half Duplex Streams) * 3) + (501 * 2 * [number of enabled MOH
codecs])
Where:
• (Maximum Half Duplex Streams) is a configuration setting on the MOH
device configuration administration web page.
• 3 indicates total steams of RX, TX, and greeting announcement .wav
file.
• 501 indicates the maximum number of Music On Hold (MOH) sources.
• 2 indicates music .wav stream and possible multicast TX stream.
• [number of enabled MOH codecs] is based on how many MOH codecs
are enabled in the Cisco IP Voice Media Streaming application service
parameters.
Self Provisioning IVR (500 * 2)

Service
Where 500 indicates callers, and 2 indicates total streams from RX and TX
streams.
Hence, to enable MOH to support a maximum of 1000 callers, use the following equation: 1000 * 3 +
501 * 2 * 1 = 4002 driver streams with one enabled codec and 1000 * 3 +501 * 2 * 2 =
5004 with two enabled codecs. Reduce the remaining devices and deactivate the Self Provisioning IVR
service to limit total reservations to 6000, which allows the MOH device to make these reservations. It may
also require that you do not activate the Self Provisioning IVR service on the same server with Cisco IP Voice
Media Streaming application.
If configuration settings of the media devices exceed the capacity of the media device driver, the media devices
that register with the device driver first will be able to reserve their required stream resources. The media
devices that register later are restricted to fewer than requested stream resources. The later registered media
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Custom Features
Interwork External Multicast MOH to Unicast MOH
devices result in logging some alarm messages and automatically reducing the call count for the restricted
media device.
Note A media kernel driver with a capacity of 6000 streams might not support that many simultaneous media device
connections.
Interwork External Multicast MOH to Unicast MOH

With this release, you can configure a Cisco Unified Survivable Remote Site Telephony (SRST) router as an
audio source. This router provides multicast MOH audio for devices that are capable of multicast reception.
In this approach, devices act as if Cisco Unified Communications Manager is sending the multicast MOH
audio. However, devices that are capable of only the unicast reception cannot hear the MOH audio that an
external MOH source (for example, Cisco Unified SRST router) sends. Examples of devices that are capable
of unicast reception only can be public switched telephone network (PSTN) phones, destination to session
border controllers (SBC), and Session Initiation Protocol (SIP) trunks.
In this release of Cisco Unified Communications Manager, this feature is enhanced to receive multicast MOH
audio from an external audio source and send it as unicast MOH audio. Cisco Unified Communications
Manager uses this feature to play multicast MOH audio as unicast MOH for the devices that are capable of
unicast MOH reception only. Examples of an external MOH audio source can be a Cisco Unified SRST router
or software that can send multicast MOH audio.
An administrator configures the fields for this feature from Cisco Unified CM Administration Music On
Hold Audio Source Configuration window.
Note • This feature has no impact on existing functionality of playing multicast MOH audio using an external
audio source for the devices that are capable of multicast reception.
• For the unicast media connection, Cisco Unified Communications Manager MOH Server plays the initial
announcement and periodic announcement even if you configure the MOH audio source with external
multicast source.
Configuration Tips for the Codec-Specific Inbound Audio Stream

Configure an external multicast audio source, such as Cisco Unified SRST router, to MOH server for streaming
the required audio feed.
To configure an external multicast audio source, such as a Cisco Unified SRST router, configure the Source
IPv4 Multicast Address and Source Port Number fields in the MOH Audio Source Configuration window.
• Cisco Unified Communications Manager listens to multicast G.711 mu-law stream on external multicast
IP address and port that you configured on the MOH Audio Source configuration window. An MOH
server can transcode between the G.711 mu-law or a-law or L16 256K wideband MOH codecs. The
external multicast RTP stream uses G.711 mu-law codec for MOH as a source for G.711 mu-law or a-law
or L16 256K wideband MOH codecs. For G.711 a-law and wideband calls, Cisco Unified Communications
Manager MOH server transcodes the inbound G.711 mu-law stream to outbound G.711 a-law or wideband
stream before sending it to the device.
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Custom Features
Music On Hold Prerequisites
• Cisco Unified Communications Manager listens to multicast G.729 stream on external multicast IP and
port value added with four that is configured on the MOH audio source configuration window. For
example, if you configure an MOH audio Source with 239.1.1.1:16384, Cisco Unified Communications
Manager listens to G.711 mu-law stream on 239.1.1.1:16384 and G.729 stream on 239.1.1.1:16388 (port
value added with four). An MOH server cannot transcode for G.729 codecs. Callers who are using MOH
G.729 codec require an external multicast RTP stream using G.729 or G.729a codec.
Music On Hold Prerequisites

• Before you configure multicast, ensure that you configure MOH server and audio sources. If you want
to use fixed audio source, configure it before you configure multicast.
• Make sure to decide whether you are going to do unicast or multicast Music On Hold
• It is crucial to plan the capacity of the deployed and configured hardware and ensure that it can support
the anticipated call volume of the network. You need to know the hardware capacity for MOH resources
and consider the implications of multicast and unicast MOH in relation to this capacity. Ensure that
network call volumes do not exceed these limits. When MOH sessions reach these limits, an additional
load can result in poor MOH quality, erratic MOH operation, or loss of MOH functionality.
• If you use multicast MOH and the devices that listen to multicast MOH streams are not in the same IP
network, you must enable multicast routing in the IP network. Take care when you enable the multicast
routing to avoid the potential flooding of parts of the network with wrongly sent multicast packets
(specially, across WAN links). Disable multicasts on interfaces on which the multicast MOH packets
are not required and use the Max Hops parameter.
• For detailed information on planning your Music On Hold deployment, including server capacities, refer
to the Music On Hold capacities topics in the Cisco Collaboration System Solution Reference Network
Design.
Music On Hold Configuration Task Flow

Complete these tasks to configure Music On Hold (MOH) for your system.
Procedure

Step 1 Activate Cisco IP Voice Media Streaming, on Activate the Cisco IP Voice Media Streaming
page 605 Service Application service to enable Music
On Hold.
Step 2 Configure Music On Hold Server, on page 606 Configure basic server settings for the MOH
server.
Step 3 Upload Audio File for Music On Hold, on page Optional. Upload your own audio files to make
606 them available as MOH audio streams.
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Custom Features

Step 4 Configure Music On Hold Audio Source, on Configure Music On Hold audio streams. You
page 607 can also associate an uploaded audio files to an
MOH audio stream.
Step 5 Configure Fixed Music On Hold Audio Source, Configure the fixed Music On Hold audio
on page 608 source. The system supports a single fixed
MOH audio source (stream 51).
Step 6 Add MOH to Media Resource Group, on page Assign the Music On Hold service to a Media
608 Resource Group. The group compiles the media
resources that are available to an endpoint in a
call.
Step 7 Configure Media Resource Group List, on page Assign your Media Resouce Groups to a
609 prioritized Media Resource Group List.
Step 8 Add Media Resources to Device Pool, on page Make Music On Hold available to endpoints by
609 assigning the Media Resource Group List to a
device or device pool.
Step 9 Configure MOH Service Parameters, on page Optional. Configure optional Music On Hold
610 parameters such as default codecs and default
audio streams for calls on hold.

The Cisco IP Voice Media Streaming Application service must be Activated in order to use Music On
Hold.
Note During installation, Unified Communications Manager installs and configures a default Music On Hold audio
source. Music On Hold functionality can proceed by using the default audio source.
Procedure
Step 1 From Cisco Unified CM Administration, choose Tools > Service Activation.
Step 2 Choose a server from the Server drop-down list.
Step 3 Under CM Services , make sure the Cisco IP Voice Media Streaming App service is Activated. If the
service is deactivated, check the service and click Save.
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Custom Features
Configure Music On Hold Server
Configure Music On Hold Server

Before you begin
Make sure one or multiple Music On Hold (MOH) servers are available.
Note The Cisco Unified Communications Manager MOH server is automatically added when the Cisco IP Voice
Media Streaming Application service is activated.
Procedure
Step 1 From Cisco Unified CM Administration, choose Media Resources > Music On Hold Server.
Step 2 Click Find and select the Music On Hold server that you want to update.
Step 3 Select the Host Server.
Step 4 Enter a descriptive Music On Hold Server Name along with a Description.
Step 5 Select the Device Pool you want to use for this server.
Step 6 Configure server capacity by configuring the following fields:
• Maximum Half Duplex Stream—Set this to the maximum number of devices that can be on unicast
music on hold that is streamed from this music on hold server at any given time. You can use the following
formula to calculate the maximum:
Note (Server and deployment capacity) – ([Number of multicast MOH sources] *
[Number of enabled MOH codecs])
• Maximum Multi-cast Connections—Set this to a value that is greater than or equal to the number of
devices that might be placed on multicast MOH at any given time.
Step 7 (Optional) To enable multi-casting, check the Enable Multi-cast Audio Sources on this MOH Server check
box, and configure the multicast IP address ranges.
Step 8 Configure the additional fields in the Music On Hold Server Configuration window. For help with the fields
and their settings, see the online help.
Step 9 Click Save.
Upload Audio File for Music On Hold

Use this procedure if you want to upload customized audio files that you can make available as Music On
Hold audio streams.
Procedure
Step 1 From Cisco Unified CM Administration, choose Media Resources > MOH Audio File Management.
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Custom Features
Configure Music On Hold Audio Source
Step 3 Click Choose File and browse to the file you want to upload. Once you've selected the file, click Open.
Step 4 Click Upload.
The Upload Result window shows the result of the upload. The uploading procedure uploads the file and
performs audio conversions to create codec-specific audio files for MOH. Depending on the size of the original
file, processing may take several minutes to complete.
Step 5 Click Close to close the Upload Result window.

Step 6 Repeat this procedure if you want to upload additional audio files.
Note When you import an audio source file, Unified Communications Manager processes the file and
converts the file to the proper formats for use by the Music On Hold server. Following are examples
of valid input audio source files:
• 16-bit PCM .wav file
• Stereo or mono
• Sample rates of 48 kHz, 44.1 kHz, 32 kHz, 16 kHz, or 8 kHz
Note MOH audio source files do not automatically propagate to other MOH servers in a cluster. You
must upload an audio source file to each MOH server or each server in a cluster separately
Configure Music On Hold Audio Source

Use this procedure to configure Music On Hold audio sources. You can configure audio streams and associate
uploaded files to an audio stream. You can configure up to 500 audio streams.
Note If a new version of an audio source file is available, perform the update procedure to use the new version.
Procedure
Step 1 From Cisco Unified CM Administration, choose Media Resources > Music On Hold Audio Source.
• Click Find and select an existing audio stream.
• Click Add New to configure a new stream.
Step 3 From the MOH Audio Stream Number, select an audio stream.
Step 4 Enter a unique name in the MOH Audio Source Name field.
Step 5 Optional. Check the Allow Multi-casting check box if you want to allow this file to be multi-casted.
Step 6 Configure the audio source:
• Check the Use MOH WAV file source radio button and from the MOH Audio Source File, select the
file you want to assign.
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Custom Features
Configure Fixed Music On Hold Audio Source
• Check the Rebroadcast External Multicast Source radio button and enter the multicast source IP
Address details.
Step 7 In the Announcement Settings for Held and Hunt Pilot Calls section, assign the announcements that you
want to use for this audio source.
Step 8 Configure the remaining fields in the Music On Hold Audio Source Configuration window. For help with
the fields and their settings, see the online help.
Step 9 Click Save.
Configure Fixed Music On Hold Audio Source

For each cluster, you may define one fixed audio source (Source 51). You must set up the fixed audio source
that is configured per cluster on each MOH server. The fixed audio source originates from a fixed device that
uses the local computer audio driver.
Procedure
Step 1 From Cisco Unified CM Administration, choose Media Resources > Fixed MOH Audio Source.
Step 2 Optional. Check the Allow Multi-casting check box if you want to allow this audio source to be multi-casted.
Step 3 Check the Enable check box to enable the fixed audio source. When you check this check box, a Name is
required.
Step 4 In the Announcement Settings for Held and Hunt Pilot Calls area, configure announcements for this audio
source.
Step 5 Configure the fields in the Fixed MOH Audio Source Configuration window. For help with the fields and
their settings, see the online help.
Step 6 Click Save.
Add MOH to Media Resource Group

A Media Resource Group is a logical grouping of media resources. You may associate a media resource group
with a geographical location or a site, as required. You can also form media resource groups to control server
usage, or unicast or multicast service type.
Procedure
Step 1 From Cisco Unified CM Administration, choose Media Resources > Media Resource Group.
• Click Find and select an existing group.
• Click Add New to create a new group.
Step 3 Enter a Name and Description.
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Custom Features
Configure Media Resource Group List
Step 4 In the Available Media Resources list, select the Music On Hold resource and use the down arrow to add
the resource to the Selected Media Resources. Repeat this step for the other media resources you want to
assign to this group.
Step 5 (Optional) Check the Use Multi-cast for MOH Audio check box if you want to allow Music On Hold
multi-casting.
Step 6 Click Save.
Configure Media Resource Group List

Media Resource Group List lists the prioritized media resource groups. An application can select required
media resources from among ones that are available according to the priority order that is defined in a media
resource group list.
Procedure
Step 1 From Cisco Unified CM Administration, choose Media Resources > Media Resource Group List.
• Click Find and select an existing media resource group list.
• Click Add New to create a new media resource group list.
Step 3 Enter a Name for the list.

Step 4 From the Available Media Resource Groups list, select the groups you want to add to this list and use the
down arrow to move them to Selected Media Resource Groups.
Step 5 In the Selected Media Resource Groups list use the up and down arrows to the right of the list to edit the
prioritized order of groups.
Step 6 Click Save.
Add Media Resources to Device Pool

You can make MOH available to devices by assigning the media resource group list that contains the MOH
resource to a device or to the device pool.
Note The device in a call will use the media resource group list that is assigned to the device in the Phone
Configuration window. If none is assigned, it will use the media resource group list for the device pool that
is used for the call.
Procedure
Step 1 From Cisco Unified CM Administration, do either of the following:

• Choose System > Device Pool.
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Custom Features
Configure MOH Service Parameters
• Choose Device > Phone.
Step 2 Click Find and select an existing phone or an existing device pool.
Step 3 From the Media Resource Group List drop-down list, select the media resource group list that contains the
Music On Hold resource.
Step 4 Complete the remaining fields in the configuration window. For more information on the fields and their
Step 5 Click Save.
Configure MOH Service Parameters

Use this procedure to configure optional service parameters for Music On Hold (MOH). For many deployments
the default settings will be sufficient.
Procedure
Step 1 In Cisco Unified CM Administration, choose System > Service Parameters.

Step 2 From the Server drop-down list, select the server.
Step 3 From the Service drop-down list, select Cisco IP Voice Media Streaming.
Step 4 From the Clusterwide Parameters (Parameters that apply to all servers) area, configure optional MOH
service parameters.
Step 5 Click Save.
Step 7 Configure optional MOH parameters. For example, under Clusterwide Parameters (Service), you can assign
the default audio sources for Hold.
Step 8 Click Save.
Note All parameters apply only to the current server except parameters that are in the cluster-wide
group.
View Music on Hold Audio File

View existing Music On Hold audio files that are stored on the system.
Procedure
Step 1 In Cisco Unified CM Administration, choose Media Resources > MOH Audio File Management.
The Music On Hold Audio File Management window appears.
Step 2 View the following information for each record:
• Check box—If the audio file can be deleted, a check box appears before the File Name column.
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Custom Features
Unicast and Multicast Audio Sources
• File Name—This column displays the audio file name.

• Length—This column displays the audio file length in minutes and seconds.
• File Status—This column displays one of the following statuses of an audio file:
• Translation Complete—This status appears after a file is uploaded successfully and is available
for use as audio files for a music on hold audio source.
• In Use—This status appears after you add a Music On Hold audio source that uses this audio file
as its MOH audio source file.
Note You cannot a delete a file with In Use status.

Unicast Music On Hold is the system default option. However, you need to configure for multicast, if required.
Both multicast and unicast configurations present the same audio-source behavior to held parties. Each audio
source is used once, and the stream is split internally and is sent to the held parties. The only difference between
multicast and unicast, in this case, is how the data is sent over the network.
Table 58: Differences Between Unicast and Multicast Audio Sources
Unicast Audio Source Multicast Audio Source
Consists of streams that are sent directly from the Consists of streams that are sent from the MOH server
MOH server to the endpoint that requests an MOH to a multicast group IP address. Endpoints that request
audio stream. an MOH audio stream can join multicast MOH, as
needed.
A unicast MOH stream is a point-to-point, one-way A multicast MOH stream is a point-to-multipoint,

audio RTP stream between the server and the endpoint one-way audio RTP stream between the MOH server
device. and the multicast group IP address.
Unicast MOH uses a separate source stream for each Enables multiple users to use the same audio source
user or connection. As more endpoint devices go on stream to provide MOH.
hold through a user or network event, the number of
MOH streams increases.
An MOH audio source may be configured with an For multicast users, this announcement is not heard.
initial (greeting) announcement, which will be played
to unicast held parties. For unicast MOH users, this
announcement is heard from the beginning.
The additional MOH streams can have a negative Multicast MOH conserves system resources and
effect on network throughput and bandwidth. bandwidth.
Extremely useful in networks in which multicast is Can be problematic in situations in which a network
not enabled or devices are incapable of multicast. is not enabled for multicast or the endpoint devices
are incapable of processing multicast.
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Custom Features
Unicast Audio Source Multicast Audio Source
Includes managing devices only. Includes managing devices, IP addresses, and ports.
No requirement to define the Music On Hold server. Administrators must define at least one audio source
to allow multicasting. To define Music On Hold
servers for multicast, first define the server to allow
multicasting.
Functions without configuring MOH audio source, Functions only if both media resource groups and
MOH server, or media resource group list. media resource group lists are defined to include a
multicast Music On Hold server. For media resource
groups, you must include a Music On Hold server that
is set up for multicast. These servers are labeled as
(MOH) [Multicast]. Also, check the Use Multicast
for MOH Audio check box when you define a media
resource group for multicast.
Note The Multicast MOH Direction Attribute for SIP service parameter determines whether Cisco Unified
Communications Manager sets the direction attribute of the Session Description Protocol (SDP) in its multicast
Music On Hold (MOH) INVITE message to sendOnly or recvOnly.
If your deployment uses SIP phone uses Release 8.4 and earlier for Cisco Unified IP Phones 7940 and 7960,
or SIP phone uses Release 8.1(x) and earlier for Cisco Unified IP Phones 7906, 7911, 7941, and 7961, set
this parameter to sendOnly. Otherwise, leave this parameter set to the default value, recvOnly.
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Custom Features
Music On Hold Interactions
Music On Hold Interactions

Feature Interaction
Multicast Music On Hold Using the multicast MOH over H.323 intercluster trunk feature, you can multicast
over H.323 Intercluster MOH to work over H.323 intercluster trunks (ICT). When a call connects over
Trunks an intercluster trunk and one of the parties presses the Hold key, MOH streams
over the intercluster trunk. If you have turned on the multicast MOH and have
configured the holding party and trunk to use the multicast MOH server, MOH
streams with multicast. Only one multicast MOH stream streams over the trunk
regardless of the number of calls that are put on hold on this trunk.
Additional points regarding this feature:
• This feature does not work if any middle box between Cisco Unified
Communications Managers does not pass the new fields in Terminal
Capability Set (TCS) and OLC message.
• This feature requires no additional configuration for field up multicast MOH,
and applies only between Cisco Unified Communications Managers that
support single-transmitter multicast.
• The feature is On by default, but can be turned off by setting the Send
Multicast MOH in H.245 OLC Message service parameter to False. Setting
this value can resolve interoperability issues that the feature might cause.
Music On Hold Failover The MOH server supports Cisco Unified Communications Manager lists and
and Fallback failover as implemented by the software conference bridge and media termination
point. Upon failover, the system maintains connections to a backup Cisco Unified
Communications Manager, if available.
When a Music On Hold server fails during an active Music On Hold session, the
held party hears no music from this point. However, this situation does not affect
normal call functions.
Call Park and Directed Music On Hold allows users to place calls on hold with music that a streaming
Call Park source provides. Music On Hold allows two types of hold:
• User hold—The system invokes this type of hold when a user presses the
Hold button or Hold softkey.
• Network hold—This type of hold takes place when a user activates the
Transfer, Conference, or Call Park feature, and the hold automatically gets
invoked. This hold type applies to directed call park because directed call
park is a transfer function. However, Directed Call Park uses the Cisco Call
Manager service parameter, Default Network Hold MOH Audio Source, for
the audio source.
Extension Mobility Cross Examples include RSVP Agent, TRP, Music On Hold (MOH), MTP, transcoder,
Cluster—Media resources and conference bridge.
for the visiting phone
Media resources are local to the visiting phone (other than RSVP Agents).
613
Custom Features
Music On Hold Restrictions
Feature Interaction
Hold Reversion Cisco Unified Communications Manager supports MOH on a reverted call if
MOH is configured for a normal held call.
Media Resource Selection Held parties determine the media resource group list that a Cisco Unified
Communications Manager uses to allocate a Music On Hold resource.
Secured Music On Hold Cisco Unified Communications Manager enhances the Cisco IP Voice Media
with SRTP Streaming application service to support Secure Real-Time Protocol (SRTP).
Hence, when you enable the Cisco Unified Communications Manager cluster or
system for security, the MOH server registers with Cisco Unified Communications
Manager as an SRTP capable device. If the receiving device is also SRTP-capable,
the music media is encrypted before streaming to the receiving device.
Make sure of the following:
• Cluster security should be mixed mode—Run the utils ctl set-cluster
mixed-mode CLI command
• SIP trunks in the path support SRTP—The SRTP Allowed check box must
be checked in the Trunk Configuration window for SRTP to work over
the trunk.
• Devices support SRTP—In the Phone Security Profile used by the endpoint,
the Device Security Mode must be Encrypted.

Multicast Music On Hold Computer Telephony Integration (CTI) and media termination point (MTP)
Support devices do not support the multicast Music On Hold feature. If you configure
CTI or MTP devices with a multicast MoH device in the media resource
group list of the CTI device, call control issues may result. CTI and MTP
devices do not support multicast media streaming.
Internet Protocol Support Multicast Music On Hold supports only IPv4. The Cisco IP Voice Media
Streaming Application, which is a component of Music On Hold, supports
both IPv4 and IPv6 audio media connections for unicast Music On Hold.
Multicast Music On Hold supports IPv4 only. Devices with an IP addressing
mode of IPv6 only cannot support multicast.
Distribution of fixed-device Cisco Unified Communications Manager does not support distribution of
audio sources fixed-device (hardware) audio sources across Music On Hold servers within
a media resource group.
Unacceptable Audio Quality Because the G.729a codec is designed for human speech, if you use it with
with G.729a codec Music On Hold for music, it may not provide acceptable audio quality.
614
Custom Features
Cisco Unified Communications A Cisco Unified Communications Manager cluster or system supports only
Manager System Support virtualized deployments on Cisco Unified Computing System (UCS) servers
or other Cisco-approved third-party server configurations. You cannot use
the Music On Hold feature with an external source (USB audio dongle) for
the nodes that provide MOH from an external source.
Multicast Support The administrator can designate a Music On Hold server as either unicast
or multicast, provided that resources exist to support multicast.
Caller-specific MOH Support Caller-specific MOH is not supported when calls are received or transferred
over QSIG tunneling-enabled SIP trunks.
MP3 Format Support The Music On Hold feature does not support the MP3 format.
Interoperability between H.323 Multicast MOH does not support interoperability between H.323 and SIP
and SIP Protocols protocols.
SRTP Support Multicast MoH audio streams are not encrypted and do not support SRTP.
Multicast Streams MTPs do not support multicast streams.
Encryption of Multicast Music Cisco Unified Communications Manager does not support encryption of
On Hold RTP Streams multicast Music On Hold RTP streams. For secure MOH audio, you should
not configure multicast audio sources.
Fixed Music On Hold Device The fixed Music On Hold device cannot specify an audio source that
connects through a USB, because Cisco Unified Communications Manager
does not support USB when running on VMware. However, VMware
supports internal Music On Hold.
MOH Server Failure Cisco Unified Communications Manager takes no action when a Music On
Hold server fails during an active Music On Hold session.
Multicast MOH When an MTP resource gets invoked in a call leg at a site that is using
multicast MOH, Cisco Unified Communications Manager falls back to
unicast MOH instead of multicast MOH.
Provisioning If you do not provision the user and network MOH audio source identifiers,
or if one or both values are invalid, the caller-specific MOH information
in the SIP header is ignored. The call reverts to tone on hold and an invalid
MOH audio source alarm is raised.
Header Values • When both the user and network MOH audio source identifiers are
present in the header, any invalid value is replaced by the default value
(0).
• If both values are zero, or the only value is zero, the header in the
incoming INVITE is ignored.
615
Custom Features
Music On Hold Troubleshooting
MOH Audio Source Identifier • If you provide only one MOH audio source identifier in the SIP header,
including if a comma appears before or after the MOH audio source
identifier value, the same MOH ID is used for both user and network
MOH. The SIP trunk populates both the user and the network MOH
audio source identifiers in the SIP header so that Call Control always
receive both values.
• If there are more than two MOH audio source identifier values
separated by a comma in the header, then the first two values are used.
Subsequent values are ignored.
Administrators for Consistent Administrators are responsible to maintain consistent caller-specific MOH
Caller-specific MOH configurations when multiple Cisco Unified Communications Manager
Configurations clusters are involved.
Original Incoming Caller The original incoming caller to the call center cannot change during the
course of the entire call.
MOH Information The Music On Hold information is shared only across SIP trunks.
Music On Hold Troubleshooting

Music On Hold Does Not Play on Phone
Phone user cannot hear Music On Hold.
• G.729a codec is used with MOH for music, which may not provide acceptable audio quality.
• An MTP resource is invoked in a call leg at a site that is using multicast MoH.
• When an MTP resource gets invoked in a call leg at a site that is using multicast MoH, the caller receives
silence instead of Music On Hold. To avoid this scenario, configure unicast MoH or Tone on Hold instead
of multicast MoH.
616
CHAPTER 50
Self Care Portal
• Self Care Portal Overview, on page 617
• Self Care Portal Task Flow, on page 617
• Self Care Portal Interactions and Restrictions, on page 619
Self Care Portal Overview

From the Cisco Unified Communications Self Care Portal, users can customize features and settings for their
phones. As the administrator, you control access to the portal. Before an end user can access the portal, you
must add the user to the default Standard CCM End Users access control group, or to any access control
group that has the Standard CCM End Users role assignment. In addition, users require their user ID,
password, and the URL with which to access the portal. Users can access the portal via the following URL:
http(s)://<server_name>:<port_number>/ucmuser/
where:
• <server_name> represents the Unified Communications Manager IP address, hostname or fully
qualified domain name
• <port_number> represents the port on which to connect. The port is optional, but is useful in firewall
situations.
• ucmuser is a mandatory subpath that points to Self Care
Optionally, you can also configure enterprise parameters within Cisco Unified Communications Manager in
order to assign which phone settings are available for end users to configure. For example, the Show Call
Forwarding enterprise parameter determines whether users can configure Call Forward via the portal.
Self Care Portal Task Flow

Procedure

Step 1 Grant User Access to the Self Care Portal, on To access the portal, end users must be assigned
page 618 to the Standard CCM End Users access
617
Custom Features
Grant User Access to the Self Care Portal

control group or to any group that has the
Standard CCM End Users role assignment.
Step 2 Configure the Self Care Portal Options, on page Configure enterprise parameters in order to
618 control what configuration options are available
to users whom access the portal.
Grant User Access to the Self Care Portal

To access the portal, end users must be assigned to the Standard CCM End Users access control group or
to any group that has the Standard CCM End Users role assignment.
Procedure
Step 2 Search for the user for whom you want to provide Self-Care access.
Step 3 In the End User section, ensure that the user has a password and PIN configured.
Usually these credentials are entered when a new user is added.
Step 4 In the Permission Information section, click Add to Access Control Group.
Step 5 Click Find and select the Standard CCM End Users group or a customized group that contains the Standard
CCM End Users role.
Note For information on editing and configuring access control groups, and role assignments for access
control groups, refer to the "Configure User Access" chapter of the System Configuration Guide
Step 6 Select Save.
Configure the Self Care Portal Options

Use this procedure to configure Self Care Portal enterprise parameters in order to control what configuration
options are available to users whom access the portal.
Before you begin

Grant User Access to the Self Care Portal, on page 618
Procedure
Step 1 From Cisco Unified Communications Manager Administration, select System > Enterprise Parameters.
Step 2 Under Self Care Portal Parameters, set the Self Care Portal Default Server by selecting one of the available
servers from the drop-down list.
618
Custom Features
Self Care Portal Interactions and Restrictions
This parameter determines which Cisco Unified CM server Jabber uses to display embedded Self Care options
pages. If you select None, Jabber defaults to the Publisher.
Step 3 Configure any of the remaining Self Care Portal Parameters to enable or disable features for the portal. For
help with the fields, refer to the enterprise parameters help.
Step 4 Select Save.

The following table highlights feature interactions and restrictions with the Self-Care Portal.
Feature Interaction or Restriction
Device Onboarding via Activation If you want users to be able to activate their phones via the Self-Care
Codes Portal, the Show Phones Ready to Activate enterprise parameter must
be set to True (this is the default setting).
With this feature, users can obtain their activation code by logging in
to the Self-Care portal. They can either use the phone's video camera to
scan the barcode, or they can enter the code manually on the phone in
order to activate and register the phone.
For more information on activation codes, see the "Device Onboarding
via Activation Codes" chapter of the System Configuration Guide for
Authenticated user https request When an authenticated user makes a request to

https://{CUCM_address}/ucmuser/hostAlive/{host},
the following happens:
• If the request is successful at getting http:{host}/ or if the
request can ping {host} Cisco Unified Communications Manager
returns the string, "true".
• If the request is unsuccessful, Cisco Unified Communications
Manager returns the string "false".
Maximum Login for Extension For end users to be able to configure this setting within the Self-Care
Mobility Portal, an administrator must have checked the Allow End User to set
their Extension Mobility maximum login time option in the associated
User Profile of Cisco Unified CM Administration.
If this option is selected within the User Profile, for all users whom use
the profile, the Self-Care Portal setting overrides the
administrator-configured values of the Intra-cluster Maximum Login
Time and Inter-cluster and Maximum Login Time service parameters
in Cisco Unified Communications Manager.
619
Custom Features
620
CHAPTER 51
Emergency Call Handler
• Emergency Call Handler Overview, on page 621
• Emergency Call Handler Prerequisites, on page 622
• Emergency Call Handler Task Flow, on page 622
• Interactions, on page 629
• Emergency Call Handler Troubleshooting, on page 631
Emergency Call Handler Overview

Emergency Call Handler helps you to manage emergency calls in your telephony network while following
local ordinances and regulations.
When an emergency call is made the following is required:
• The emergency call must be routed to the local Public-Safety Answering Point (PSAP) based on the
location of the caller.
• The caller's location information must be displayed at the emergency operator terminal. The location
information can be obtained from an Automatic Location Information (ALI) database.
The caller's location is determined by the Emergency Location Identification Number (ELIN). An ELIN is a
Direct Inward Dial (DID) number that the PSAP can dial to reconnect to the emergency caller if the emergency
call is cut off or if the PSAP needs to talk to the caller again. The emergency call is routed to the PSAP based
on the location information that is associated with this number.
For multiline phone systems, such as an office system, the ELIN can be associated with more than one telephone
by grouping the phones in an ELIN group. An ELIN group in Emergency Call Handler identifies a location.
The ELINs under this ELIN group must be mapped to the location in the ALI database.
Each location should have as many ELINs created as needed to support simultaneous emergency calls. For
example, to support five simultaneous calls, five ELINs would be needed in an ELIN group.
Note Emergency Call Handler supports a maximum of 100 ELINs per cluster.
Make sure that the mapping of ELIN to the original called party is active until the ELIN is used for the next
emergency call from the same location. If the ELIN mapping is not used, the DN will be active for a maximum
period of 3 hours only.
621
Custom Features
Emergency Call Handler Prerequisites
The following types of phone are supported to use ELIN groups:

• SIP and SCCP IP phones
• CTI ports
• MGCP and SCCP analog phones
• H.323 phones
Emergency Call Handler Prerequisites

Example
Before deploying Emergency Call Handler in your network, we recommend that you test the ALI
submission process. With your service provider's help, test that the PSAP can successfully call back
into your network using the ALI data.
Reserve the ELIN number from your local PSAP. Ordinances and regulations can differ across
different locations and across different companies, so research your security and legal needs before
deploying this feature.
Emergency Call Handler Task Flow

Before you begin
• Review Emergency Call Handler Prerequisites, on page 622
Procedure

Step 1 Enable Emergency Call Handler, on page 623 Enable the Emergency Call Handler feature on
Emergency Call Handler provides essential
emergency call features and supports a limited
number of locations with phone location
assignment by static configuration. If you
require advanced emergency call features, such
as a greater amount of specific locations or
dynamic location assignment, consider Cisco
Emergency Responder.
Step 2 Configure Emergency Location Groups, on Configure an Emergency Location (ELIN)

page 624 Group for a particular site or location.
Step 3 Add a Device Pool to an Emergency Location Configure device pools to use an Emergency
Group, on page 624 Location (ELIN) Group.
622
Custom Features
Enable Emergency Call Handler

Step 4 (Optional) Add Device to an Emergency Configure a particular device to use a particular
Location Group, on page 625 Emergency Location (ELIN) Group. If you want
to use the device pool ELIN Group that is
associated for this device, you can ignore this
section.
Note Configurations that are made at
the device level will overwrite any
configurations that were made at
the device pool level.
Step 5 Enable Route Patterns and Translation Patterns, Enable the Emergency Location (ELIN) service
on page 626 for a route pattern or a translation pattern.
Caution No Calling Party Transformation
masks are set at the Gateway or
Trunk, because these may
transform the ELIN that is set by
Emergency Call Handler.
Note It is mandatory that you enable

either route patterns or translation
patterns, but it is possible to
enable both.
Step 6 (Optional) Use the following procedures to This section provides information about the
perform bulk administration tasks on ELIN Bulk Administration tasks you can use to update
group information and phones: ELIN group information and to add phones to
new ELIN groups. For Bulk Administration,
• Import Emergency Location Group
see the Cisco Unified Communications Manager
Information, on page 627
Bulk Administration Guide, Release 11.0(1).
• Export Emergency Location Group
Information, on page 627
• Update Phones with a new Emergency
Location Group, on page 628
Enable Emergency Call Handler

Enable the Emergency Call Handler feature on Cisco Unified Communications Manager. Emergency Call
Handler provides essential emergency call features and supports a limited number of locations with phone
location assignment by static configuration. If you require advanced emergency call features, such as a greater
amount of specific locations or dynamic location assignment, consider Cisco Emergency Responder.
Note Do not enable this feature if you are already using an external emergency calling solution such as Cisco
Emergency Responder.
If you decide to enable this feature, make sure you disable the external one.
623
Custom Features
Configure Emergency Location Groups
Procedure
Step 1 From Cisco Unified CM Administration, choose Call Routing > Emergency Call Handler > Emergency
Location Configuration.
Step 2 From the Emergency Location Configuration window:
• To enable the Emergency Call Handler feature, check the Enable Emergency Location (ELIN) Support
check box. The setting default is Disabled. When enabled, the settings related to this feature appear in
the Related Settings pane. You must configure these settings for the feature to work. Refer to the tasks
below for further details on how to configure these related settings.
• To disable the Emergency Call Handler feature, uncheck the Enable Emergency Location (ELIN)
Support check box.
Note If you disable this feature, all related settings that are configured will be removed. See the
Related Settings Pane for all configured settings.
Note If you want to disable the feature and you have more than 500 devices associated with ELIN
Groups, then you must manually delete the associations until there are fewer than 500
associations before you can disable the feature.
Step 3 Click Save.
Configure Emergency Location Groups

Configure an Emergency Location (ELIN) Group for a particular site or location.
Before you begin

Enable Emergency Call Handler, on page 623
Procedure
Step 1 From Cisco Unified CM Administration, choose Call Routing > Emergency Call Handler > Emergency
Location (ELIN) Group.
Step 2 In the Emergency Location (ELIN) Group Configuration window, enter a name for the group in the Name
field.
Step 3 In the Number field, enter the pool of DID numbers that are registered in the Public Safety Answering Point
(PSAP).
Step 4 Click Save.
Add a Device Pool to an Emergency Location Group

Configure device pools to use an Emergency Location (ELIN) Group.
624
Custom Features
Add Device to an Emergency Location Group
Before you begin

Configure Emergency Location Groups, on page 624
Procedure
Step 2 In the Find and List Device Pools window, if you are adding an existing device pool, click Find and choose
the device pool from the list. If you are adding a new device pool click Add New.
Step 3 In the Device Pool Configuration window, choose the ELIN group to which you want to add the device pool
from the Emergency Location (ELIN) Group drop-down list. If you are adding a new device pool, fill out
any other required fields.
Step 4 Click Save.
Add Device to an Emergency Location Group

Configure a particular device to use a particular Emergency Location (ELIN) Group. If you want to use the
device pool ELIN Group that is associated for this device, you can ignore this section.
Note Configurations that are made at the device level will overwrite any configurations that were made at the device
pool level.
Note The devices that you add to the ELIN Group, should be added to the ELIN Group that represents the particular
location at which those devices are located.
Before you begin

Add a Device Pool to an Emergency Location Group, on page 624
Procedure

Note If you are using a type of phone that is not an IP phone, go to the relevant configuration page for
that type of phone.
Step 2 In the Find and List Phones window, if you are adding an existing device, click Find and choose the device
you want to configure from the list. If you are adding a new device, click Add New.
Step 3 If you are adding a new phone, choose the type of phone you want to add from the Phone Type drop-down
list and click Next.
625
Custom Features
Enable Route Patterns and Translation Patterns
Step 4 In the Phone Configuration window, choose the ELIN group to which you want to add the device from the
Emergency Location (ELIN) Group drop-down list. If you are adding a new device, fill out any other
required fields.
Step 5 Click Save.
Enable Route Patterns and Translation Patterns

Enable the Emergency Location (ELIN) service for a route pattern or a translation pattern.
Note It is mandatory that you enable either route patterns or translation patterns, but it is possible to enable both.
Before you begin

Add Device to an Emergency Location Group, on page 625
Procedure
Step 1 From Cisco Unified CM Administration, choose one of the following:

• To enable a route pattern, choose Call Routing > Route/Hunt > Route Pattern.
• To enable a translation pattern, choose Call Routing > Translation Pattern.
Step 2 In the Find and List Route Patterns or Find and List Translation Patterns window, click Find and choose
a route pattern or translation pattern from the list.
Step 3 In the Route Pattern Configuration or Translation Pattern Configuration window, check the Is an
Emergency Services Number check box.
Note Check this check box only if you are using Emergency Call Handler and not another external
emergency calling solution such as Cisco Emergency Responder.
Step 4 Click Save.
Bulk Administration of Emergency Location Groups and Phones

• Bulk Administration of Emergency Location Groups and Phones Task Flow, on page 626
Bulk Administration of Emergency Location Groups and Phones Task Flow

This section provides information about the Bulk Administration tasks you can use to update ELIN group
information and to add phones to new ELIN groups. For more information about Bulk Administration, see
the Cisco Unified Communications Manager Bulk Administration Guide, Release 11.0(1).
626
Custom Features
Import Emergency Location Group Information
Note Before you perform these procedures, make sure that you have enable the Emergency Call Handler feature.
See Enable Emergency Call Handler, on page 623.
Procedure

Step 1 Import Emergency Location Group Information, Import Emergency Location (ELIN) Group
on page 627 information using the Bulk Administration Tool.
Step 2 Export Emergency Location Group Information, Export Emergency Location (ELIN) Group
on page 627 information using the Bulk Administration Tool.
Step 3 Update Phones with a new Emergency Location Find and list multiple phones and configure
Group, on page 628 them with a new Emergency Location (ELIN)
Group.
Import Emergency Location Group Information

Import Emergency Location (ELIN) Group information using the Bulk Administration Tool.
Procedure
Step 1 From Cisco Unified CM Administration, choose Bulk Administration > Import/Export > Import.
Step 2 From the File Name drop-down list, choose the name of the .tar file you want to import, and click Next.
Step 3 The Import Configuration section lists all the components of the .tar file. Check the ELIN Group-related
check boxes for the options that you want to import.
Step 4 Choose to run the job immediately or later by clicking the corresponding radio button.
Step 5 To create a job for importing the selected data, click Submit. A message in the Status section notifies you
know that the job was submitted successfully.
Step 6 Use the Job Scheduler option in the Bulk Administration main menu to schedule or activate this job.
Export Emergency Location Group Information

Export Emergency Location (ELIN) Group information using the Bulk Administration Tool.
Before you begin

Import Emergency Location Group Information, on page 627
Procedure
Step 1 From Cisco Unified CM Administration, choose Bulk Administration > Import/Export > Export.
627
Custom Features
Update Phones with a new Emergency Location Group
Step 2 In the Export Data window, in the Job Information pane, enter the .tar file name, without the extension,
in the Tar File Name field. BPS uses this filename to export the configuration details.
Note All files that are exported at the same time get bundled together (.tar) and can be downloaded
from the server.
Step 3 To export ELIN Group information, check the Elin Group check box on the Select items to Export pane.
Step 4 (Optional) Perform these steps:
• To export device pools with ELIN Groups configured, check the Device Pools check box.
• To export phones with ELIN Groups configured, check the Phone check box.
Step 5 In the Job Descripton field, enter the description that you want to override for the job. Export Configuration
is the default description.
Step 6 You can choose to run the job immediately or later by clicking the corresponding radio button.
Step 7 To create a job for exporting the selected data, click Submit. A message in the Status pane notifies you that
the job was submitted successfully.
Step 8 Use the Job Scheduler option in the Bulk Administration main menu to schedule or activate this job.
Update Phones with a new Emergency Location Group

Find and list multiple phones and configure them with a new Emergency Location (ELIN) Group.
Before you begin

Export Emergency Location Group Information, on page 627
Procedure
Step 1 From Cisco Unified CM Administration, choose Bulk Administration > Phones > Update Phone > Query.
Step 2 In the Find and List Phones To Update window, set the parameters for your search and click Find.
Note To update all phones, click Find and do not specify a query.
Step 3 The Find and List Phones To Update window displays the details of the phones that you chose. Click Next.
Step 4 In the Update Phones window, check the Emergency Location (ELIN) Group check box, and choose a
new ELIN Group from the drop-down list.
628
Custom Features
Interactions
Interactions
Feature Interaction
Do Not Disturb Call Calls made by PSAP CallBack will overwrite a Do Not Disturb (DND)
Reject configuration of a destination device.
If DND Call Reject is enabled, when the emergency number is dialed using the
translation pattern, an ELIN will be associated for this outbound emergency call.
If the call is disconnected and the ELIN is called back using PSAP CallBack, the
call is routed to the phone irrespective of the phone's DND settings.
Call Forward All Calls made by PSAP CallBack will overwrite Call Forward All (CFA) settings
of the destination device.
If a phone has CFA enabled and if the emergency number using the translation
pattern is dialed, an ELIN will be associated for this outbound emergency call.
If the call is disconnected and the ELIN is called back using PSAP CallBack, the
call is routed to the phone irrespective of the phone's CFA settings.
Single Number Reach PSAP CallBack will ignore the Single Number Reach (SNR) configuration.
When a phone has SNR enabled with the Remote Destination pointing to a mobile
number. If the emergency number is dialed using the translation pattern, an ELIN
will be associated for this outbound emergency call. If the call is disconnected,
and the ELIN number is called back using PSAP CallBack, the call is routed to
the phone and not to the remote destination.
629
Custom Features
Interactions
Feature Interaction
Extension Mobility PSAP CallBack call will consider Extension Mobility (EM) status.
If you log in with EM profile credentials and dial the emergency number using
the translation pattern, an ELIN will be associated for this outbound emergency
call. If the call is disconnected and the ELIN where the user is still logged in is
called back using PSAP CallBack, the call is routed to the device which initiated
the call.
Note This is the device on which the user is still logged in.
PSAP CallBack will fail if a user logs out of EM before a PSAP CallBack is
performed.
When a user logs in with EM profile credentials, and the emergency number is
dialed using the translation pattern, an ELIN will be associated for this outbound
emergency call. If the call is disconnected and is called back using PSAP
CallBack, if the user has since logged out, the call will not route to the device
that initiated the call and will fail.
PSAP CallBack with a user logged in on a different device.

When a user logs in with EM profile credentials at Phone A and dials the
emergency number using the translation pattern, an ELIN will be associated for
this outbound emergency call. If the call is disconnected, the user should log out
from Phone A. If the user then logs in to another phone, Phone B, with the same
profile, and the ELIN is called back using PSAP CallBack, the call is then be
routed to Phone B with normal priority, meaning CFA settings will be ignored
and DND settings will not be ignored.
PSAP CallBack call with multiple logins.

When a user logs in with EM profile credentials at Phone A and dials the
emergency number using the translation pattern, an ELIN number will be
associated for this outbound emergency call. If the call is disconnected and the
user logs in to another phone, Phone B, with the same profile while the user is
still logged in on Phone A, and the ELIN is called back using PSAP CallBack,
then the call is routed to Phone A only, the device on which the call originated.
Device Mobility A roaming device will use the Roaming Device Pool's ELIN Group for an
outbound emergency call.
Move a device with Device Mobility enabled from its home location to the
Roaming location, a change in IP subnet, so that it gets associated with the
Roaming device pool. If the emergency number is dialed using the translation
pattern, an ELIN is associated for this outbound emergency call. The ELIN
belongs to the ELIN Group that is associated with the Roaming Device Pool.
630
Custom Features
Emergency Call Handler Troubleshooting
Feature Interaction
Shared Lines PSAP CallBack rings only on the device which made the emergency call even if
the line is shared by different devices.
Phone A and Phone B share a Directory Number (DN). If the emergency number
is dialed using the translation pattern, an ELIN is associated for this outbound
emergency call. If the call is disconnected, and the ELIN is called back using
PSAP CallBack, the call is routed to Phone A only, the device from which the
call originated.
Emergency Call Handler Troubleshooting

About Emergency Call Handler Troubleshooting Scenarios
This section provides information about some Emergency Call Handler troubleshooting scenarios in the
following areas:
• Configuration Scenarios
• Outgoing Calls Scenarios
• Incoming Calls Scenarios
Configuration Scenarios
Emergency Calls Get Busy Signals and Are Not Routed
Problem:
Emergency calls get busy signals and are not routed.
Solution:
If a user who is dialing the emergency call is running a reorder tone, perform the following checks:
• Check whether the translation or route pattern for the emergency call has been used. This may require
checking for the device or phone on CSS.
• Check whether the Is an Emergency Services Number check box has been checked for the translation
or route pattern of the emergency call, and that it is correctly routing to the gateway.
If the user who is dialing the emergency call is not reaching the correct gateway or Public Service Answering
Point (PSAP), check that the settings or device pool settings for the phone or device are configured with the
correct Emergency Location (ELIN) Group.
Emergency Location Numbers Are Dialed from Outside Running a Reorder Tone
Problem:
Emergency Location (ELIN) numbers are dialed from outside while running a reorder tone.
631
Custom Features
Outgoing Calls Scenarios
Cause:
In this case the ELINs have been set as DID which is used to identify a caller's location. This should not be
used on any phone or for any other purpose.
Solution:
Check the ELIN configuration information, and unset the ELINs that have been set as DID.
Outgoing Calls Scenarios

Outgoing Emergency Call Does Not Contain Calling Party as Emergency Location Number
Problem:
An outgoing emergency call does not contain the calling party as an Emergency Location (ELIN) number.
Cause:
The translation pattern or route pattern for this ELIN was not configured correctly.
Solution:
Check that the translation pattern or route pattern settings are correctly configured for this ELIN, and make
sure that the Is an Emergency Services number check box is checked on the relevant translation pattern or
route pattern configuration page.
Outgoing Emergency Call Contains Modified Emergency Location Number

Problem:
An outgoing emergency call contains a modified Emergency Location (ELIN) number.
Cause:
The outgoing trunk or route list contains extra transformations that are not required for ELINs.
Solution:
Check the transformations that were applied for the call, and make sure that only the required transformations
for ELINs are present on the outgoing trunk or route list.
Incoming Calls Scenarios

Incoming PSAP Callback Call Fails
Problem:
An incoming PSAP Callback call fails.
Cause:
The device that made the original emergency call was not registered correctly.
Solution:
Check whether the device that made the original emergency call is still registered and whether any Extension
Mobility is involved.
632
Custom Features
Incoming PSAP CallBack Call is Not Routed as Expected

Problem:
An incoming PSAP CallBack call does not get routed as expected.
Cause:
The Emergency Location (ELIN) number does not match the number of the original dialed party.
Solution:
For an ELIN to be successfully reverse mapped to the original dialed party, these two numbers must match.
If there are already transformations at the incoming Gateway or Trunk and significant digits configured, make
sure that the final transformed called party matches the ELIN number.
633
Custom Features
634
CHAPTER 52
Enterprise Groups
• Enterprise Groups Overview, on page 635
• Enterprise Groups Prerequisites, on page 636
• Enterprise Groups Configuration Task Flow, on page 636
• Enterprise Groups Deployment Models (Active Directory), on page 640
• Enterprise Groups Limitations, on page 642
Enterprise Groups Overview

When Enterprise Groups is configured, Cisco Unified Communications Manager includes user groups when
it synchronizes its database with an external LDAP directory. In Cisco Unified CM Administration, you can
view synced groups in the User Groups window.
This feature also helps administrators to:
• Provision users with similar characteristics traits with a comment set of features (for example, the sales
and accounting teams).
• Target messages to all users in a specific group.
• Configure uniform access for all members of a specific group
This feature also helps Cisco Jabber users to quickly build contact lists of users who shares common traits.
Cisco Jabber users can search the external LDAP Directory for user groups and then add them to their contact
list. For example, a Jabber user can search the external LDAP directory and add the sales group to a contact
list, thereby adding all of the sales team members into the contact list as well. If the group gets updated in the
external directory, the user's contact list is updated automatically.
Enterprise Groups is supported with Microsoft Active Directory on Windows as the external LDAP directory.
Note If you disable the Enterprise Groups feature, Cisco Jabber users cannot search for enterprise groups or see
the groups that they already added to their contact lists. If a user is already logged in when you disable the
feature, the group will be visible until the user logs out. When the user logs in again, the group will not be
visible
635
Custom Features
Enterprise Groups Prerequisites
Security Groups
Security Groups are a subfeature of Enterprise Groups. Cisco Jabber users can also search for, and add, security
groups to their contact list. To set up this feature, administrators must configure a customized LDAP filter
and apply it to the configured LDAP directory sync. Security Groups are supported with Microsoft Active
Directory only.
Maximum Allowed Entries

When configuring Enterprise Groups, make sure that you configure contact list maximums that handle groups
• The maximum number of entries that are allowed in a contact list is the sum of the number of entries in
the contact list and the number of entries in groups that are already added to the contact list.
• Maximum entries in contact list = (number of entries in contact list) + (number of entries in groups)
• When the Enterprise Groups feature is enabled, Cisco Jabber users can add the groups to the contact list
if the number of entries in the contact list is less than the maximum allowed entries. If the maximum
allowed entries is exceeded while the feature is disabled, the users are not restricted until the feature is
enabled. If the user continues to be logged in after the feature is enabled, no error message is displayed.
When the user logs out and logs in again, an error message is displayed that asks the users to clear the
excess entries.
Enterprise Groups Prerequisites

This feature assumes that you already have an LDAP Directory sync schedule configured with the below
conditions. For details on how to configure an LDAP Directory sync, see the "Import Users from LDAP
Directory" chapter of the System Configuration Guide for Cisco Unified Communications Manager.
• The Cisco DirSync service must be activated
• The LDAP Directory sync must include both users and groups
• Regular LDAP Directory syncs, as configured with the LDAP Directory Synchronization Schedule
must be scheduled.
Supported LDAP Directories

Only Microsoft Active Directory is supported with enterprise groups.
Enterprise Groups Configuration Task Flow

Complete these tasks to configure the Enterprise Groups feature.
Procedure

Step 1 Verify Group Sync from LDAP Directory, on Confirm that your LDAP Directory sync
page 637 includes both users and groups.
636
Custom Features
Verify Group Sync from LDAP Directory

Step 2 Enable Enterprise Groups, on page 637 Complete this task to enable Cisco Jabber users
to search for enterprise groups in Microsoft
Active Directory and add them to their contact
lists.
Step 3 Enable Security Groups, on page 638 (Optional) If you want Cisco Jabber users to be
able to search for and add security groups to
their contact lists, complete this task flow.
Step 4 View User Groups, on page 640 (Optional) View enterprise groups and security
groups that are synchronized with Cisco Unified
Communications Manager database.
Verify Group Sync from LDAP Directory

Use this procedure to confirm that your LDAP Directory sync includes users and groups.
Procedure
Step 1 From Cisco Unified CM Administration, choose Server > LDAP > LDAP Directory.
Step 2 Click Find and select the LDAP directory from which you are syncing enterprise groups.
Step 3 Confirm that the Synchronize field has Users and Groups selected.
Step 4 Complete any remaining fields in the LDAP Directory configuration window. For help with the fields and
their settings, refer to the online help.
Step 5 Click Save.
Enable Enterprise Groups

Configure the system to include enterprise groups in LDAP Directory syncs.
Procedure
Step 1 From Cisco Unified CM Administration, chooseSystem > Enterprise Parameters.

Step 2 Under User Management Parameters, set the Directory Group Operations on Cisco IM and Presence
parameter to Enabled.
Step 3 Enter a value for the Maximum Enterprise Group Sized to allow Presence Information parameter. The
permitted range is 1 to 200 users with a default value of 100 users.
Step 4 From the Syncing Mode for Enterprise Groups drop-down list configure the LDAP sync that you want to
perform at regular intervals: None, Differential Sync, Full Sync.
Note Refer to the enterprise parameter help for additional assistance in configuring these fields.
637
Custom Features
Enable Security Groups
Step 5 Click Save.
Enable Security Groups

If you want to allow Cisco Jabber users to be able to add a security group to their contact list, complete these
optional tasks to include security groups in an LDAP Directory sync.
Note Security group sync is supported from Microsoft Active Directory only.
Note You cannot add new configurations into an existing LDAP Directory configuration in Cisco Unified
Communications Manager where the initial sync has already occurred.
Procedure

Step 1 Create Security Group Filter, on page 638 Create an LDAP filter that filters both directory
groups and security groups.
Step 2 Synchronize Security Groups from LDAP Add your new LDAP filter to an LDAP
Directory, on page 638 Directory sync.
Step 3 Configure Cisco Jabber for Security Groups, Update existing service profiles to give Cisco
on page 639 Jabber users whom are associated to that service
profile access to search and add security groups.
Create Security Group Filter

Create an LDAP filter that filters security groups.
Procedure
Step 1 From Cisco Unified CM Administration, choose System > LDAP > LDAP Filter.
Step 3 Enter a unique Filter Name. For example, syncSecurityGroups.
Step 4 Enter the following Filter: (&(objectClass=group)(CN=*)).
Step 5 Click Save.
Synchronize Security Groups from LDAP Directory

Add your Security Group filter to an LDAP Directory sync and complete a sync.
638
Custom Features
Configure Cisco Jabber for Security Groups
Note You cannot add new configurations into an existing LDAP Directory configuration in Cisco Unified
Communications Manager if the initial LDAP sync has already occurred.
Note For detailed information on how to set up a new LDAP Directory sync, see the "Configure End Users" part
of the System Configuration Guide for Cisco Unified Communications Manager.
Before you begin

Create Security Group Filter, on page 638
Procedure
Step 1 In Cisco Unified CM Administration, choose System > LDAP > LDAP Directory.
• Click Add New to create a new LDAP Directory.
• Click Find and select the LDAP Directory from which the security groups will be synchronized.
Step 3 From the LDAP Custom Filter for Groups drop-down list, select the security group filter that you created.
Step 4 Click Save.
Step 5 Configure any remaining fields in the LDAP Directory Configuration window. For more information on
Step 6 Click Perform Full Sync Now to synchronize immediately. Otherwise, security groups will be synchronized
when the next scheduled LDAP sync occurs.
Configure Cisco Jabber for Security Groups

Update existing service profiles to allow Cisco Jabber users whom are associated to that service profile to add
security groups from an LDAP directory to their contact lists.
Note For information on how to set up new service profiles and assign them to Cisco Jabber users, see the "Configure
Service Profiles" chapter of the System Configuration Guide for Cisco Unified Communications Manager.
Before you begin

Synchronize Security Groups from LDAP Directory, on page 638
639
Custom Features
View User Groups
Procedure
Step 1 Complete any remaining fields in the Service Profile Configuration window. For help with the fields and
Step 2 Click Find and select the service profile that your Jabber users use.
Step 3 Under Directory Profile, check the Allow Jabber to Search and Add Security Groups check box.
Step 4 Click Save.
Cisco Jabber users who are associated to this service profile can now search and add security groups.
Step 5 Repeat this procedure for all service profiles that your Cisco Jabber users use.
View User Groups

You can view the enterprise groups and security groups that are synchronized with the Cisco Unified
Communications Manager database using the following steps.
Procedure
Step 1 From Cisco Unified CM Administration, choose User Management > User Settings > User Group.
The Find and List User Groups window appears.
A list of user groups that match the search criteria is displayed.
Step 3 To view a list of users that belong to a user group, click on the required user group.
The User Group Configuration window appears.
A list of users that match the search criteria is displayed.
If you click on a user in the list, the End User Configuration window appears.
Enterprise Groups Deployment Models (Active Directory)

The Enterprise Groups feature offers two deployment options for Active Directory.
Important Ensure that Cluster 1 and Cluster 2 have a unique set of UserGroup, UserGroupMember, and
UserGroupWatcherList records before synchronizing data through the Cisco Intercluster Sync Agent service.
If both the clusters have unique sets of records, both the clusters will have a super set of all the records after
synchronization.
640
Custom Features
Enterprise Groups Deployment Models (Active Directory)
Enterprise Groups Deployment Model 1

In this deployment model, Cluster 1 and Cluster 2 synchronize different subsets of users and groups from
Microsoft Active Directory. The Cisco Intercluster Sync Agent service replicates the data from Cluster 2 into
Cluster 1 to build the complete database of users and groups.
Figure 12: Enterprise Groups Deployment Model 1
Enterprise Groups Deployment Model 2

In this deployment model, Cluster 1 synchronizes all the users and groups from Microsoft Active Directory.
Cluster 2 synchronizes only users from Microsoft Active Directory. The Cisco Intercluster Sync Agent service
replicates groups information from Cluster 1 into Cluster 2.
Caution If you are using this deployment model, ensure that you synchronize the groups data in only one cluster. The
Enterprise Groups feature will not work as expected if you fail to do so.
You can verify your configuration on the Cisco Unified CM IM and Presence Administration > Presence >
Inter-Clustering window.
Check the status of the Enterprise Groups LDAP Configuration parameter in the Inter-cluster peer table.
No conflict found means there are no misconfigurations between peers. If there are conflicts found, click the
Enterprise GroupConflicts link, and click the details button which appears. This opens a Reporting window
for a detailed report.
641
Custom Features
Enterprise Groups Limitations
Figure 13: Enterprise Groups Deployment Model 2

Table 59: Enterprise Groups Limitations
Limitation Description
Block Everyone When a Cisco Jabber user enables the "Block Everyone" feature from
within their Cisco Jabber policy settings, the block prevents other Jabber
users from viewing or exchanging IMs and Presence with the blocking
user, unless they are listed as a contact in the blocking user's contact
list.
For example, a Cisco Jabber user (Andy) has enabled Block everyone
within his personal Jabber settings. The following list breaks down how
Andy's block affects other Jabber users whom may or may not be
included in Andy's personal contact list. In addition to the block, Andy
has a personal contact list that:
• Includes Bob—Because Bob is in Andy's personal contact list, he
can still send IMs and view Andy's presence despite the block.
• Omits Carol—Carol cannot view Andy's presence or send IMs due
to the block..
• Omits Deborah as a personal contact. However, Deborah is a
member of an enterprise group that Andy has listed as a
contact—Deborah is blocked from viewing Andy's presence or
sending IMs to Andy.
Note that Deborah is blocked from viewing Andy's presence, or sending

IMs to Andy, despite the fact that she is a member of an enterprise group
in Andy's contact list. For additional details on enterprise group contacts
behavior, see CSCvg48001.
642
Custom Features
Intercluster peering with a 10.x Enterprise Groups is supported for releases 11.0(1) and higher.
cluster
If the synced group includes group members from a 10.x intercluster
peer, users on the higher cluster cannot view the presence of synced
members from the 10.x cluster. This is due to database updates that were
introduced in 11.0(1) for the Enterprise Groups sync. These updates are
not a part of the 10.x releases.
To guarantee that users homed on the higher cluster can view the
presence of group members homed on the 10.x cluster, users on the
higher cluster should manually add the 10.x users to their contact lists.
There are no presence issues for manually added users.
Multilevel grouping Multilevel grouping is not allowed for the group sync.
Group-only synchronization When a user group and users are present in the same search base,
group-only synchronization is not allowed. Instead, the user group as
well as the users are synchronized.
Maximum number of user groups You can synchronize a maximum of 15000 user groups from Microsoft
Active Directory server to the Unified Communications Manager
database. Each user group can contain from 1 to 200 users. You can
configure the exact amount on the Cisco Unified CM IM and Presence
Administration > System > Service Parameters window.
The maximum number of user accounts in the database cannot exceed
160,000.
User group migration If a user group is moved from one organization unit to another, you must
perform a full sync on the original unit followed by a full sync on the
new unit.
Local groups Local groups are not supported. Only groups synchronized from
Microsoft Active Directory are supported.
Group members not assigned to IM Group members that are not assigned to IM and Presence Service nodes
and Presence Service nodes display in the contact list with the presence bubble greyed out. However,
these members are considered when calculating a maximum numbers
of users allowed in the contact list.
Migration from Microsoft Office During migration from Microsoft Office Communication Server, the
Communication Server Enterprise Groups feature is not supported until users are fully migrated
to the IM and Presence Service node.
LDAP synchronization If you change the synchronization option in the LDAP Directory
Configuration window while the synchronization is in progress, the
existing synchronization remains unaffected. For example, if you change
the synchronization option from Users and Groups to Users Only when
the synchronization is in progress, the users and groups synchronization
still continues.
643
Custom Features
Group search functionality over the Group search functionality over the Edge is offered in this release, but
Edge has not been fully tested. As a result, full support for group searches
over the Edge cannot be guaranteed. Full support is expected to be
offered in a future release.
Cisco Intercluster Sync Agent If a group name or a group member name is updated in the external
service periodic synchronization LDAP directory, it gets updated on the Cisco Jabber contact list only
after the periodic Cisco Intercluster Sync Agent service synchronization.
Typically, the Cisco Intercluster Sync Agent service synchronization
occurs every 30 minutes.
Synchronization of users and user If users and user groups are synchronized into the Cisco Unified
groups through different Communications Manager database as part of the same synchronization
synchronization agreements in agreement, the user and group association gets updated as expected in
LDAP configuration Cisco Unified Communications Manager database after synchronization.
However, if a user and user group are synchronized as part of different
synchronization agreements, the user and the group may not get
associated in the database after the first synchronization. The user and
group association in the database depends on the sequence in which the
synchronization agreements are processed. If the users are synchronized
ahead of the groups, then the groups may not be available in the database
for association. In such cases, you must ensure that the synchronization
agreement with groups is scheduled ahead of the synchronization
agreement with the users. Otherwise, after the groups synchronize into
the database, the users will get associated with the groups after the next
manual or periodic sync with the sync type set as Users and Groups.
Users and corresponding group info will be mapped only when the
agreement sync type is set as Users and Groups
.
Tested OVA information for Tested Scenario

Enterprise Groups
In a Intercluster deployment with two clusters Cluster A and Cluster B:
Cluster A has 15K OVA and 15K users enabled for IM and Presence
Service out of 160K users that are synced from Active Directory. The
tested and supported average number of enterprise groups per user on
15K OVA cluster is 13 enterprise groups .
Cluster B has 25K OVA and 25K users enabled for IM and Presence
Service out of 160K users that are synced from Active Directory. The
tested and supported average number of enterprise groups per user on
25K OVA is 8 enterprise groups.
The tested and supported sum of user's personal contacts in roster and
the contacts from enterprise groups that are in a user's roster is less than
or equal to 200.
Note In environments with more than 2 clusters these numbers
are not supported.
644
Custom Features
Export Contact List When you export the user's contact list using Bulk Administration >
Contact List > Export Contact List, the Contact List CSV file doesn't
include the details of enterprise group they had in Jabber client.
645
Custom Features
646
PA R T XIII
Device Management
• Headset Management, on page 649
• Video Endpoints Management, on page 665
CHAPTER 53
Headset Management
• Headset Management Overview, on page 649
• Feature Compatibility for Headset Management, on page 649
• Workflow: Configure Headset Serviceability, on page 651
• Headset Template Management, on page 655
• Firmware Management, on page 660
• Headset Inventory Management, on page 660
• Headset Troubleshooting and Diagnostics , on page 663
Headset Management Overview

Headset Management enhances your Cisco headset deployment, letting administrators manage headset
serviceability from Cisco Unified Communications Manager. From Cisco Unified CM Administration,
administrators can:
• Remotely configure headset settings such as wireless power range, audio bandwidth, and Bluetooth
on/off.
• Define and control the headset firmware.
• Get a detailed inventory of all the headsets in your deployment.
• Diagnose and troubleshoot headsets with Remote PRT, headset metrics in Call Management Records
(CMR), and alarms.
Feature Compatibility for Headset Management

Cisco Headset Management is supported in Unified Communications Manager from the following releases:
• Release 11.5(1)SU7 for 11.x releases
• Release 12.5(1)SU1 for 12.x releases
Along with the Unified Communications Manager version, feature support is dependent on the firmware
versions of Cisco Headsets, Cisco IP Phone and Cisco Jabber. The following table lists the available headset
management features depending on the headset, phone, and Unified Communications Manager versions you
use.
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Device Management
Feature Compatibility for Headset Management
Note The Cisco Headset Management feature is not supported in 12.0(x) or 12.5(1). For earlier versions, you may
have a limited support for sending headset configuration templates for IP phones manually via the
defaultheadsetconfig.json configuration file and TFTP. Refer to your headset Administration
Guide for details.
Table 60: Headset Serviceability Features for Cisco IP Phones
New Serviceability Feature Unified Unified Unified Unified Unified Unified

CM 12.5(1) CM CM 12.5(1) CM CM 12.5(1) CM
or earlier 12.5(1)SU1 or earlier 12.5(1)SU1 or earlier 12.5(1)SU1
+ Phone and + Phone and + Phone and
Firmware above** + Firmware above** + Firmware above** +
12.1(1) or Phone 12.5(1) Phone 12.5(1)SR3 Phone
earlier Firmware Firmware Firmware
12.1(1) or 12.5(1) 12.5(1)SR3
earlier
COP file installation required X X X X X —
Manual remote configuration — — X N/A X —
Headset firmware management on — — — — — X

Unified CM
Remote headset configuration — — — — — X

through Unified CM
Headset inventory on Unified CM — — — — — X*
Configuration Reset on the phone — — — — X X

UI
Headset Call Management Records — — — — — X*

(CMR)
• * This feature is only available on headsets with Headset Firmware 1.5 or later.
• **This feature is not supported in the 12.0.x and 12.5(1) releases.
• N/A When you upgrade to Unified CM 11.5(1)SU7 or higher from an earlier version, most Cisco IP
Phones will upgrade automatically to Phone Firmware 12.5(1)SR3 or higher versions.
650
Device Management
Third-Party Headset Support
Table 61: Headset Serviceability Features for Cisco Jabber
New Serviceability Feature Unified Unified Unified Unified Unified Unified

CM 12.5(1) CM CM 12.5(1) CM CM 12.5(1) CM
or earlier 12.5(1)SU1 or earlier 12.5(1)SU1 or earlier 12.5(1)SU1
+ Jabber and + Jabber and + Jabber and
version above** + version above** + version above** +
12.5(1) or Jabber 12.6(1) Jabber 12.6(1)MR Jabber
earlier version version version
12.5(1) or 12.6(1) 12.6(1)MR
earlier
COP file installation required X X X X X X
Headset firmware management — — — — — X

through Unified CM
Remote headset configuration — — — X — X

through Unified CM
Headset inventory on Unified CM — — — X* — X*
Local configuration reset — — — — X X
Local UI configuration — — X X X X
Local headset version display — — — — X X
• * This feature can only detect headsets with Headset Firmware 1.5 or later.
• **This feature is not supported in the 12.0.x and 12.5(1) releases.
Third-Party Headset Support

If you are deploying third-party headsets, Cisco Unified Communications Manager supports headset inventory
management with limited information for the third-party headsets right from the Cisco Unified CM
Administration interface. Cisco Unified Communications Manager does not support headset configuration
templates, firmware, diagnostics, and headset CMRs for third-party headsets.
Workflow: Configure Headset Serviceability

Use the following workflow to guide you through the setup of your Cisco Headset Serviceability feature.
After you complete this workflow, you can configure headset settings, maintain headset latest firmware loads,
headset association to users, enable headset-based Extension Mobility, and maintain inventory.
Procedure

Step 1 Activate Cisco Headset Service, on page 652 Turn on Cisco Headset Service in Cisco
Unified Serviceability.
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Device Management
Activate Cisco Headset Service
Step 2 Prepare Your Headset COP Files, on page 653 Make sure you install and upgrade the latest
headset firmware using a COP file.
Step 3 Configure User Profiles for Headset Users, on If you haven't yet configured User Profiles, use
page 653 this procedure to set up profiles for your users.
If all User Profiles are configured, you can skip
this task.
Step 4 Apply User Profiles to End Users, on page 655 Assign User Profiles to your end users. If you've
already assigned User Profiles, you can skip
this task.
Step 5 Configure a Headset Template, on page 659 Configure default settings and firmware for a
Cisco headset template. Associate User Profiles
to the template such that users whom use that
User Profile are assigned to this headset
template.
Step 6 View Headset Inventory, on page 661 Check that you can see your deployed headset
inventory through the Cisco Unified CM
interface.
Activate Cisco Headset Service

Before you can begin administering Cisco Headsets and Accessories through the Cisco Unified CM
Administration interface, turn on in Cisco Unified Communications Manager Serviceability.
Note Cisco Headset service should be activated on all the Unified Communications Manager nodes wherever Cisco
CallManager service is already running. Ensure that you activate the Cisco Headset service on the Unified
Communications Manager nodes where you want to administer headsets using the Cisco Unified CM
Administration interface. The Cisco CallManager service will be automatically activated when you enable
the Cisco Headset service. Deactivate the Cisco CallManager service if you do not need it.
Procedure
Step 1 From Cisco Unified CM Administration, navigate to Cisco Unified Serviceability and click Go.
Step 2 Select Tools > Service Activation.
Step 3 Check the Cisco Headset Service check box from the CM Services section and select Save.
What to do next
Prepare your Headset COP Files.
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Device Management
Prepare Your Headset COP Files
Prepare Your Headset COP Files

You can install and upgrade the latest headset firmware using a COP file. A headset COP file contains all the
firmware versions of different headset models along with their configuration data.
Note Ensure that the Cisco Headset service is up and running before the COP file is installed.
Ensure that the headset COP file is installed on all nodes of Unified Communications Manager.
1. Install or upgrade the COP file to the Unified Communications Manager system before you can start using
your Cisco headsets .
When you connect your headset to the endpoints, the headset template configuration changes are applied. If
you make any updates to the headset template configurations on Unified Communications Manager, the
endpoints apply these configuration updates on the connected headsets .
All configuration updates depend on the version of the headset template in the COP file. If the headset template
version is higher in the latest COP file, the configuration file on Unified Communications Manager is updated.
If the configuration file in the COP file is upgraded, the headset template version in Unified Communications
Manager is updated irrespective of the version of the template and vice versa. The following list shows the
various template version update scenarios after a COP file upgrade:
• If the Unified Communications Manager is currently installed with the headset template version 1-10
and you upgrade your Unified Communications Manager server that has headset template version 1-12,
then the chosen headset template version is 1-12. Unified Communications Manager opts for the higher
headset template version.
and you upgrade your Unified Communications Manager server that has headset template version 1-9,
then the chosen headset template version is 1-10. Unified Communications Manager opts for the higher
headset template version.
and you install a COP file that has headset template version 1-12, then the chosen headset template
version is 1-12. Headset template installed with the COP files is the preferred option.
and you install a COP file that has headset template version 1-9, then the chosen headset template version
is 1-9. Headset template installed with the COP files is the preferred option.
• If you had a COP file installed that has headset template version 1-12 and you upgrade your Unified
Communications Manager server having headset template version 1-10, then the chosen headset template
version is 1-12. Unified Communications Manager opts for the higher headset template version.
Configure User Profiles for Headset Users

If you haven't yet configured User Profiles for your users, use this procedure to set up profiles. Your headset
templates will be assigned to users via their User Profile. If you've already configured User Profiles, you can
skip this task.
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Device Management
Configure User Profiles for Headset Users
Note Configure multiple User Profiles for different groups of users as per your deployment needs. By default, all
User Profiles get assigned to the System default headset template. You can assign them to customized templates
when you configure your headset template.
Procedure
Step 1 From Cisco Unified CM Administration, choose User Management > User Settings > User Profile.
Step 3 Enter a Name and Description for the user profile.
Step 4 Assign a Universal Device Template to apply to users' Desk Phones, Mobile and Desktop Devices, and
Remote Destination/Device Profiles.
Step 5 Assign a Universal Line Template to apply to the phone lines for users in this user profile.
Step 6 If you want the users in this user profile to be able to use the self-provisioning feature to provision their own
phones, do the following:
a) Check the Allow End User to Provision their own phones check box.
b) In the Limit Provisioning once End User has this many phones field, enter a maximum number of
phones the user is allowed to provision. The maximum is 20.
Step 7 If you want Cisco Jabber users associated with this user profile to be able to use the Mobile and Remote
Access feature, check the Enable Mobile and Remote Access check box.
Note By default, this check box is selected. When you uncheck this check box, the Jabber Policies
section is disabled and No Service client policy option is selected by default.
Note This setting is mandatory only for Cisco Jabber users. Non-Jabber users do not need this setting
to be able to use Mobile and Remote Access. The Mobile and Remote Access feature is applicable
only for Jabber Mobile and Remote Access users and not to any other endpoints or clients.
Step 8 Assign the Jabber policies for this user profile. From the Jabber Desktop Client Policy, and Jabber Mobile
Client Policy drop-down list, choose one of the following options:
• No Service—This policy disables access to all Cisco Jabber services.
• IM & Presence only—This policy enables only instant messaging and presence capabilities.
• IM & Presence, Voice and Video calls—This policy enables instant messaging, presence, voicemail, and
conferencing capabilities for all users with audio or video devices. This is the default option.
Note Jabber desktop client includes Cisco Jabber for Windows users and Cisco Jabber for Mac users.
Jabber mobile client includes Cisco Jabber for iPad and iPhone users and Cisco Jabber for Android
users.
Step 9 If you want the users in this user profile to set the maximum login time for Extension Mobility or Extension
Mobility Cross Cluster through the Cisco Unified Communications Self Care Portal, check the Allow End
User to set their Extension Mobility maximum login time check box.
Note By default Allow End User to set their Extension Mobility maximum login time check box
is unchecked.
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Device Management
Apply User Profiles to End Users
Step 10 Click Save.
Apply User Profiles to End Users

Associate your users to the User Profiles that you've created. The User Profile must be associated with the
end user, and the MAC of the device must be added under the controlled devices to apply the headset template
configuration changes.
Note If you've already assigned all users to the appropriate User Profiles, you can skip this task.
Procedure
Step 1 To add a new end user to the Unified Communications Manager database manually, perform the following:
a) In Cisco Unified CM Administration, choose User Management > End User.
b) Click Add New.
c) Enter the User ID and Last name.
d) Choose the User Rank from the drop-down list.
e) Complete the fields in the End User Configuration window. For field descriptions, see the online help.
f) Click Save.
Step 2 To associate the end user with the device, perform the following:
a) In Cisco Unified CM Administration, choose Device > Phone.
b) Select the Cisco IP Phone or device.
c) Under Device Information, select User as the Owner and select the Owner User ID.
d) Click Save and Apply Config for the configuration changes to take effect.
Headset Template Management

You can assign headset templates to user profiles in Cisco Unified Communications Manager to configure
default headset settings for your users. The headset template provides the option to associate User Profiles.
Unified Communications Manager supports the following types of headset templates:
Standard Default Headset Configuration Template

This is the system default template that contains the factory default settings for all headset model series. This
template contains the headset settings supported by the latest headset firmware installed on your system for
all your headset model series. You cannot edit the default settings though you can change the profile
configuration setting.
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Note The Standard Default Headset Configuration template is created only when the Cisco Headset Service is
activated in the Cisco Unified Serviceability user interface.
By default, all User Profiles are associated to the standard headset template unless the administrator associates
these user profiles to any of the custom defined headset templates. You can make copies of the standard default
headset template to create custom template with customized values of the parameters including the headset
firmware version.
System Generated Custom Headset Template

For some earlier releases that did not support the full Cisco Headset Serviceability feature, administrators
could configure and deploy headset templates manually via the defaultheadsetconfig.json
configuration file and TFTP. If you used this method on a previous release, and then upgrade to this release,
the config file is converted to the System Generated Custom Headset Template and displays in the Headset
Template Configuration window. Following the upgrade, users and devices that used the config file are
associated to this custom template.
Custom Headset Configuration Template

From Cisco Unified CM Administration, use the Device > Headset > Headset Template window to customize
headset templates as per your deployment needs. You can assign different headset parameters to different
models in the same template. You can also assign different firmware loads to different headset models. The
custom headset settings can be assigned to specific sets of users by associating the User Profile(s) to the
Custom Headset Template.
Table 62: Headset Template Configuration Settings
Field Description
Headset Template Configuration
Name Enter a unique name to identify the headset template.
Description Enter a description that identifies use of the template.
Model and Firmware Settings
Choose Model Series Choose any supported headset model that offers reliable, high-quality sound for
your device.
Add For a standard template, you can view the default pre-defined firmware versions
and settings of the headset models. You cannot edit the default values.
For customized templates, click Add to add a new headset model and
corresponding settings. You cannot add another existing headset model in the
same template. You can add different headset models in a customized template;
however, you can only use one firmware per headset model. For more information
on headset parameters, see the "Headset Configuration Parameters" table below.
For Standard Default Headset Template Configuration, you can only edit settings
by installing a headset COP file.
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Field Description
Firmware Select the required firmware version.

• Remain on current version—Choose this option if you want the headset to
remain on the existing firmware version (that is, the headset firmware version
is not upgraded to the latest firmware version on the system).
• Latest—Choose this option if you want to upgrade the headset firmware
version to the latest firmware version on the system.
Delete For customized templates, click Delete to remove the headset model from the
headset template.
Profile Configuration
Available User Profiles Lists the configured User Profiles that are available to use with this headset
template.
To associate a User Profile to this template, select the profile and click the down
arrow to move the template to Assigned User Profiles.
Note By default, all User Profiles get assigned to the Standard Default
Headset Configuration Template. To associate a User Profile to a
different template, create the new template and assign the User
Profile to the new template.
Assigned User Profiles Lists the User Profiles that will use this headset configuration template. For users
assigned to this profile, the settings in this headset configuration template are
applied to their Cisco headsets during registration.
Click the arrows to add new User Profiles from the Available User Profiles list.
The following table describes the parameters in each headset template.
Note On-premises and multiplatform headset serviceability features are unavailable through an RJ-9 connection.
Table 63: Cisco Headset 500 Series Parameters
Parameter Range Default Notes
Speaker Volume 0 – 15 7 Controls the level of sound in the headset. 0 is

very low while 15 is loud.
Configure this setting based on the ambient
noise in the office environment.
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Microphone Gain Softer – Louder Default Gain controls how loud the user sounds to other
people on the call. Softer means users sound
quiet while Louder means users sound much
louder.
Configure this setting based on the ambient
noise in the office environment.
Sidetone Off – High Low Controls how much of a user's own voice they
can hear through their headset.Off turns off the
sidetone while High means that users receive
much more feedback from their headset
microphones.
Equalizer Warmest – Brightest Default Controls the Equalizer settings. Warmer settings
mean users hear more bass in their headsets,
while a brighter setting means users hear more
treble.
Audio Bandwidth Wide Band, Narrow Wide Controls the Digital Enhanced cordless
Band Band Telecommunications (DECT) codec in the Cisco
Headset 560 Series.
In a dense DECT environment, set the field to
Narrow Band to limit the Cisco Headset 560
Series to the G.727 codec.
Bluetooth On, Off On Controls the use of Bluetooth on the Cisco

Headset 560 Series with Multibase. When this
parameter is set to Off, the base deletes all
devices paired with it.
Conference On, Off On Controls the use of the conferencing feature on

the Cisco Headset 560 Series. Conferencing
allows up to three guest headsets to pair with
the same base at once.
See Cisco Headset 500 Series User Guide for
more information on conferencing.
Firmware Source Allow from UCM or Allow Controls the headset's firmware upgrade source.
Cisco Cloud (firmware from
By default, users can upgrade their headset
will upgrade only), UCM or
through a devices and software connected to
Restrict to UCM only Cisco
Unified CM or through a cloud-connected
(firmware may Cloud
device or software. You can restrict your
upgrade or downgrade)
headsets to only accept firmware changes
through a Unified CM source.
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Configure a Headset Template
DECT Radio Range Autorange, Medium Medium Controls the maximum distance between the
Range, Short Range Range Cisco Headset 560 Series and its base.
By default, the bases have a DECT range of
over 330 feet (100 meters) in ideal conditions.
If you configure the DECT radio range to
Medium Range or Short Range, the headset
base consumes less power but users can't move
as far from the base while on a call. Configure
DECT radio range to Short Range for high
density headset deployment.
For more detailed information on DECT
deployment, refer to the white paper on Cisco
Headset deployment, How to Deploy DECT at
Work for the Cisco Headset 560 Series.
Headset dock behavior On, Off On Controls how the Cisco Headset 560 Series
behaves if you lift the headset off the base when
you have an incoming call.
Configure a Headset Template

Use this procedure to configure a headset template with customized settings that you can apply to Cisco
headsets. You can create a customized template or use the system-defined Standard Default Headset Template.
Note The Standard Default Headset Configuration Template is a system-defined template. You can assign new
User Profiles to the Standard Default Headset Template but you can't edit the template. By default, all user
profiles are assigned to this template. To disassociate a user profile from this template, you must assign the
profile to a new template.
Procedure
Step 1 From Cisco Unified CM Administration, choose Device > Headset > Headset Template.
• To edit an existing template, select the template.
• To create a new template, select any existing template and click Copy. The existing settings are applied
to your new template.
Step 3 Add a Name and Description for the template.

Step 4 Under Model and Firmware Settings, assign any customized headset settings that you want to apply to this
template. To add a new setting, click the Add button and configure the settings.
Step 5 Use the up and down arrows to move the User Profiles that you want to assign to this template to the Assigned
Users Profiles list box. All users whom are assigned to those profiles will also be assigned to this headset
template.
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Firmware Management
Step 6 Click Save.

Step 7 Use the Set to Default button to return to the default template settings.
For a Standard Default Headset Configuration Template, the Apply Config button takes effect for the following:
• Devices owned by users you added to the Assigned User Profile list
• Anonymous devices
For a Customized Headset Configuration Template, the Apply Config button takes effect only for devices
owned by users you added to the Assigned User Profiles list.
Firmware Management
Most phones and devices connected to the Unified Communications Manager support the Cisco Headset 500
Series and Cisco Headset 700 Series. Install the latest phone firmware release and device package before
connecting your headset to a phone. When the headset first connects, it downloads the required firmware and
begins the upgrade process.
For a given headset model, the following two firmware options are supported:
• Remain on current version—Choose this option if you want the headset to remain on the existing
firmware version (that is, the headset firmware version is not upgraded to the latest system firmware
version).
• Latest—Choose this option to upgrade or downgrade the headset. The system installs and runs the chosen
software, even if that firmware is an older release from what the headset currently has.
For example, if you choose 1-5-1-10 as the latest, that firmware will be installed on the headset regardless
of whether the headset currently has 1-5-1-9 or 1-5-1-11.
Firmware Considerations
• Users assigned to the standard headset template will always receive the latest headset firmware and
settings.
• Settings shown in the Headset Template Configuration (both Standard and Custom) are always set to the
Latest firmware for all headset model series.
Headset Inventory Management

Cisco IP Phones send headset inventory data to Unified Communications Manager whenever the headset is
in a connected or disconnected state. Unified Communications Manager stores the inventory data so you can
generate an Inventory Summary Report or Custom Inventory Report for all headsets deployed in this server.
Report information includes: headset serial and model number, docking station details, firmware, configuration
templates used, vendor details, and headset connection status to devices.
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Headset Inventory Settings
Headset Inventory Settings

From Cisco Unified CM Administration, use the Device > Headset > Headset Inventory window to view a
full list of all headsets deployed on your server. You can use this information to generate reports for all
deployed headsets. If you click the Serial Number of the device, you can view details of individual headsets
in a pop-up window.
Table 64: Headset Inventory Settings
Field Description
Serial Number Serial Number of the headset. This number is unique for every individual headset.
The Cisco Headset 520 and 530 Series report the serial number found on the USB
controller. The Cisco Headset 560 and 700 Series report the headset serial number
found on inside of the left armband.
Note For non-Cisco headsets, the Device Name is used as the Serial
Number. Using the same non-Cisco headset with multiple phones
creates duplicate headset records.
Model Model number of the headset.
Vendor Displays vendor details.
Type Indicates the type of headset connection: Wired, DECT Wireless, or Unknown.
Firmware Displays the most current firmware load of the headset.
User Displays information of the end user using the phone or device.
Template Display the name of the headset configuration template.
Status (since) Displays the status of the headset activities. It can be: Connected or Disconnected.
Dock Model Displays the type of docking model station.
Device Name Name of the device to which the headset is connected to.
Device Model Displays the Cisco IP Phone or Cisco Jabber model number. For example,
CP-8865 is a Cisco IP Phone model. CSF is a device type for either Cisco Jabber
for Mac or Cisco Jabber for Windows.
Software Version Displays the latest version of the software used. It can be a phone firmware or a
Jabber software version.
Headset Age (days) Displays the age of the headset. If the record is deleted, the headset age is reset.
View Headset Inventory

You can view a full list of all headsets deployed on your server. You can use this information to generate
reports for all deployed headsets.
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Headset Inventory Summary
Procedure
Step 1 From Cisco Unified CM Administration, choose Device > Headset > Headset Inventory.
• Select Find to see a full list of headsets deployed on your server.
• Enter a one or more search criteria into the search box and select Find.
Headset Inventory Summary

From Cisco Unified CM Administration, you can use the Device > Headset > Headset Inventory Summary
window to view an aggregate summary of your deployed headsets in the Headset Inventory Summary
window.
Headset Inventory by Model
Field Description
Headset Model The headset model number.
Quantity Lists the number of headsets for each model type in your deployment.
Note Click the link in the Quantity column to navigate to the detailed
Headset Inventory page, filtered by model type.
Headset Inventory by Status

Click the hyperlinks in the Headset Model, Active, Inactive, or Unassigned columns to navigate to the
detailed Headset Inventory page for each status.
Field Description
Headset Model The headset model number.
Active The headset has connected within the last 30 days.
Inactive The headset hasn't connected in the last 30 days.
Unassigned The user ID doesn't exist in the system or the inventory record doesn't have a
user ID mapping.
Get an Aggregate Summary of Your Deployed Headsets

You can view an aggregate summary of your deployed headsets in the Headset Inventory Summary window.
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Headset Troubleshooting and Diagnostics
Procedure
In Cisco Unified CM Administration, select Device > Headset > Headset Inventory Summary.
You can view a breakdown of headset inventory by model or by headset status.
Headset Troubleshooting and Diagnostics

You can configure Unified Communications Manager or Cisco Unified Real-Time Monitoring Tool (RTMT)
to collect Problem Report Tool (PRT) logs for headsets connected to Cisco IP Phones. The PRT includes data
on call quality, codecs used, audio settings, wireless settings, and alert logs.
Unified Communications Manager stores the call diagnostics details for Headsets. Cisco IP Phones send
headset diagnostics data in Headset-Stat header either in a BYE message or a 200 OK response to BYE
message to update the CMRs in Unified Communications Manager.
Cisco IP Phones share the headset diagnostics data with Unified Communications Manager and this information
is stored in the following fields in the CMR record:
• SN—Serial number of the headset.
• Metrics—Headset metrics such as RSSI frame errors, connection drop reason, beacon moves, audio
settings, and DECT bandwidth.
For detailed information on how to export and view CMR records, see the Call Reporting and Billing
Administration Guide for Cisco Unified Communications Manager.
Note Headset CMR records apply to Cisco Headset 500 series, but not to 700 series.
Generate PRT for Endpoints on Unified CM

Use this procedure to trigger the Problem Reporting Tool (PRT) on the endpoints.
Procedure

Step 2 Click Find and select one or more phones that the headset connects to.
Step 3 Click Generate PRT for Selected to collect PRT logs for the headsets used by the selected phones.
Step 4 Click Save.
Cisco Unified Communications Manager sends SIP Notify messages to remotely trigger the log collection on
the phone and upload it to the log server configured in the “Customer support upload URL” parameter.
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Generate PRT for Endpoints on RTMT
Generate PRT for Endpoints on RTMT

Devices or endpoints generate alarms for each critical event for diagnostics and troubleshooting. These alarms
are generated using the Problem Reporting Tool (PRT) available in the Trace Collection menu or the Device
Monitoring menu of the Cisco Unified Real-Time Monitoring Tool (RTMT) user interface.
Procedure
Step 1 Open the Trace and Log Central options.

Step 2 In the Trace & Log Central tree hierarchy, choose Generate PRT.
The Generate PRT wizard appears.
Step 3 Enter the Device name as configured in the Find and List Phones page in the Cisco Unified CM Administration
user interface.
Step 4 Click Generate PRT.
The generated report is uploaded at the Customer support upload URL. The download option is available
only if the Customer support upload URL parameter is configured at the Enterprise, Profile, or Device level
in the Cisco Unified CM Administration user interface.
Note Check the Customer support upload URL parameter in the Enterprise, Profile, or Device level
configuration page settings. Else, PRT generation fails.
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CHAPTER 54
Video Endpoints Management
• Video Endpoints Management Overview, on page 665
• Video Endpoints Management Feature Compatibility, on page 666
• Migration Considerations for Video Endpoints Provisioning, on page 667
• Video Endpoints Migration Report, on page 668
• Provisioning and Migration Scenarios, on page 669
Video Endpoints Management Overview

This feature simplifies the administrator's job of provisioning and managing Cisco TelePresence video
endpoints. An administrator can provision settings for Cisco TelePresence endpoints in Unified Communications
Manager and then push those Product-Specific Configuration settings to endpoints.
Prior to Release 12.5(1)SU1, only a limited set of Product-Specific Configurations were pushed from Unified
Communications Manager to the endpoint resulting in a partial configuration of the endpoint. Administrator
had to rely on Cisco TelePresence Management Suite or TelePresence Endpoint's web interface to configure
all the settings. The Phone Configuration window in Unified Communications Manager contains a complete
Product-Specific Configuration layout for Cisco TelePresence endpoints that matches what users see on their
endpoint. This update lets administrators apply settings on behalf of users and then push those settings to
users.
Note The Bulk Administration Tool (BAT) Phone Template Configuration page also displays the new
model-specific configurations in a tabbed layout, supporting the complete list of endpoint parameters. You
can import the entire set of parameters or modify a specific parameter in the endpoint in bulk.
Video endpoints managements feature provides the following benefits:

• TelePresence endpoints can be fully provisioned from Unified Communications Manager—Endpoints
parameters listed in the Unified Communications Manager user interface are in the same order as listed
in the Advanced Configuration settings of your Cisco TelePresence model. For more information on
the various advanced parameters, see the respective model in the Collaboration Endpoints Administrator
Guides.
• New Product-Specific Configuration Layout—New layout details the model-specific configurations
in a tabbed layout. This is an upgrade from the earlier flat format that provided access only to a limited
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Video Endpoints Management Feature Compatibility
set of parameters. The new layout ensures that you have a complete list of Cisco TelePresence settings
on the Cisco Unified CM Administration interface.
• Automatic migration of the configuration data from the video endpoints—This simplifies the deployment
of endpoints by automatically synching data from endpoints to Unified Communications Manager and
vice versa. Endpoint configurations can be fully restored in case of reset to factory settings or Product
Returns & Replacements (RMA) swaps.
Note Any endpoint that supports Collaboration Endpoint (CE) Software 9.8 or higher can use this new provisioning
layout for the Product-Specific Configuration fields on the Phone Configuration page. If you are using a CE
software version prior to 9.8, you will be able to view all the new set of advanced parameters; but, the new
set of parameters functions only if you upgrade your CE Software version to 9.8 or higher. The subset of
parameters supported is marked with a “#” to the right of each parameter value in the user interface. You must
load a device pack onto Unified Communications Manager if a device type is capable of supporting the new
provisioning framework, but does not show the additional parameters.
Video Endpoints Management Feature Compatibility

Following table details the video endpoints management feature compatibility with Unified Communications
Manager and Collaboration Endpoint (CE) versions:
Unified Communications CE Endpoint Version Expected Behavior

Manager Version
12.5(1) SU1 9.8 and above Devices added prior to 12.5(1) SU1:
• Advanced Configuration UI (Tabbed Layout)
for successfully backed up devices
• Limited Configuration UI (Flat Layout) for
devices yet to be backed up
New device added through UI/BAT/AXL:

• Advanced Configuration UI
Note It's highly recommended that you run

CE 9.8 or higher.
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Migration Considerations for Video Endpoints Provisioning
Unified Communications CE Endpoint Version Expected Behavior

Manager Version
12.5(1) SU1 9.7 and below Devices added prior to 12.5(1) SU1:
• Limited Configuration UI
New device added through UI/BAT/AXL:

• Advanced Configuration UI with only a
limited set of parameters taking into effect
Note For migrations with CE 9.7 or earlier,

you cannot maintain the existing
endpoint configuration during
migration. When the migrated device
registers, Unified CM overwrites the
existing configuration with default
settings.
12.5(1) and below 9.8 and above Limited Configuration UI
Migration Considerations for Video Endpoints Provisioning

Auto Backup After Unified Communications Manager Upgrade
When upgrading to Unified Communications Manager 12.5(1)SU1, the existing configuration data for the
supported endpoint types is automatically migrated from endpoints to Unified Communications Manager.
1. Upgrade Unified Communications Manager to version 12.5(1)SU1 or later.
2. Endpoints register to Unified Communications Manager.
3. Unified Communications Manager then sends a SIP Notify message to endpoints requesting for the full
set of Product-Specific Configuration parameters.
4. Endpoints that are upgraded to CE 9.8 and above send full set of configuration data to Unified
Communications Manager (in xConfiguration format) using a SIP REFER message.
5. Unified Communications Manager processes this configuration data and populates the complete list of
Cisco TelePresence settings (Advanced Configuration UI) on the Cisco Unified CM Administration
interface.
Note Unified Communications Manager server displays the complete endpoint configuration settings in the new
layout only if Unified CM is able to successfully back up data from the endpoint.
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Video Endpoints Migration Report
Configuration Control Modes

Based on the deployment needs, administrators can configure various configuration control modes in the
Cisco Unified CM Administration interface. You can decide whether you want to control the configuration
settings centrally from the endpoints or Unified Communications Manager or both of them together.
Navigate to the Product-Specific Configuration Layout section on the Phone Configuration page and choose
the Configuration Control Mode under “General Settings” in the Miscellaneous tab to control the various
modes. Following are the various Configuration Control Modes:
• Unified CM and Endpoint (Default)—Use this mode if you want Unified Communications Manager
and endpoint to operate as the multi-prime source for provisioning endpoint data. If Unified CM and
Endpoint is the configured mode, any update made via an endpoint locally is synched with the Unified
CM server.
• Unified CM—Use this mode if you want Unified Communications Manager to operate as the centralized
primary source for provisioning endpoint data and does not want to accept any configurations done from
the endpoints locally.
• Endpoint—Use this mode if you want endpoints to operate as the centralized primary source of
configuration data. In this mode, endpoint ignores any configuration data from Unified Communications
Manager and doesn’t sync back the changes done locally. This mode is typically used when an Audiovisual
(AV) integrator is installing the endpoints and wants to control configuration from the endpoint.
Note In the Endpoint mode, CE devices continue to accept that limited set of parameters supported prior to release
12.5(1)SU1. Unified Communications Manager indicates these parameters with a "#" symbol. CE devices
will ignore the extended set of parameters supported from the 12.5(1)SU1 release onwards.
On-demand Configuration Pull Functionality

Administrators can use the Get Config from Phone option to pull configuration changes from the CE 9.8
endpoint devices on-demand at that given point.
Navigate to the Product-Specific Configuration Layout section on the Phone Configuration page and click
the Get Config from Phone button on the top corner of the page to pull any data configuration from the CE
9.8 endpoints on-demand. This option is enabled only if the endpoint is in the registered state.
Video Endpoints Migration Report

Video Endpoint with Extended Configuration Backup is the new filter is introduced on the Find and List
Phones window for release 12.5(1)SU1. Administrators can search for details on how many CE endpoints got
migrated automatically and how many CE endpoints did not. Based on this information, they can take corrective
measures.
Note In the Find and List Phones window, the Video Endpoint with Extended Configuration Backup filter is
applicable only for video endpoints running Collaboration Endpoint (CE) Software 9.8 or higher.
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Provisioning and Migration Scenarios
Provisioning and Migration Scenarios

The following table describe various provisioning and migration scenarios. All of these scenarios assume that
your TelePresence video endpoints are upgraded to a CE release that supports Product-Specific Configuration
provisioning from Unified CM. In Unified CM, these settings appear in the Product-Specific Configuration
section, but on the endpoint, they appear under Advanced Configuration.
Table 65: Provisioning and Migration Scenarios for Video Endpoints
Task Existing Configuration What to do

Summary
Provisioning New Video • Brand new device With Unified CM at a minimum release 12.5(1)SU1 and
Endpoints the CE endpoint at 9.8, you can provision new endpoints
• Device is not and manage the product-specific configurations from
provisioned on Unified Unified CM.
CM
• No existing settings on
the device or on
Unified CM
Migrating Existing Video • Existing device If you are migrating existing video endpoints from a Cisco
Endpoints from VCS TelePresence Video Communications Server to Cisco
• Device is not Unified Communications Manager:
provisioned on Unified
CM Adding Phones via Phone Configuration window in
Unified CM:
• Device is configured,
but Unified CM does • Add the phone to Unified CM, but DO NOT CLICK
not have any of the Save.
configurations
• Register the phone. After registration, the existing
Advanced Configuration settings from the phone are
uploaded to Unified CM and display in
Product-Specific Configurations in the Phone
• In the Phone Configuration window, configure the

new settings and click Save. The provisioned settings
download to the phone.
For a detailed procedure, see Add Migrating Video
Endpoint to Unified CM, on page 670
Adding Phones via Bulk Administration

Make sure that the csv file or BAT Template that you use
for provisioning does not include the Product-Specific
Configuration fields.
Adding Phones via AXL
Make sure that the AXL request does not include any
Product-Specific Configuration fields.
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Add Migrating Video Endpoint to Unified CM
Task Existing Configuration What to do

Summary
Upgrading from an Earlier • Existing device So long as the CE endpoint is at a supported version, when
Release of Unified CM with you upgrade Unified CM, the Advanced Configuration
Registered Video Endpoints • Device is provisioned settings from the endpoint get pulled into Unified CM
on a pre-12.5 release of automatically following device registration and display
Unified CM under the Product-Specific Configuration section of the
• Unified CM has a Phone Configuration window.
limited set of After registration, you can set the Configuration Control
Product-Specific Mode in addition to whatever settings you want.
Configuration settings
for the device

If you are migrating existing Cisco TelePresence Video Endpoints from a Cisco TelePresence Video
Communications Server to Unified Communications Manager, use this procedure to add the CE endpoint into
Unified CM via the Phone Configuration window so that the existing Advanced Configurations from the
endpoint can be managed from the Phone Configuration window in Unified CM.
Note Make sure to follow this procedure closely. The settings from the endpoint do not automatically upload to
Unified CM until after device registration.
Note This procedure uses the Add New from Template setting in the Unified CM Phone Configuration window.
You can also use tools like Bulk Administration or AXL to add the endpoint.
Before you begin

It's highly recommended that you upgrade firmware to CE 9.8 or higher before you migrate. With CE 9.7 or
earlier, Unified CM overwrites the existing endpoint configuration during registration with default settings.
Procedure

Step 2 Click Add New from Template and enter the following phone details:
• Select the model from the Phone Type drop-down list.
• Enter the MAC Address of the endpoint
• From the Device Template, select a Universal Device Template.
• Select the Directory Number that you want to add to the phone. If none exists, click New and configure
a directory number.
• From the User drop-down list, select the user whom will own the device.
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Step 3 Click Add. The Phone Configuration displays with the universal device template settings filling out the
phone configuration. The Product-Specific Configuration section also appears, but with default settings,
rather than the existing settings from the phone.
Note You can also add the device using the Phone Configuration window's Add New, but this method
requires that you enter settings manually.
Step 4 DO NOT CLICK Save. If you save settings, Unified CM does not load existing settings from the phone. If
you saved by mistake, go straight to the troubleshooting Note at the bottom of this procedure for recovery
steps.
Step 5 Register the phone.
During registration, the existing Advanced Configuration settings from the phone get pulled into Unified
CM and display in the Phone Configuration window's Product-Specific Configuration section.
Step 6 In the Phone Configuration window, configure how you want endpoint settings to be managed by configuring
the Configuration Control Mode field:
• Unified CM and Endpoint (Default)—Use this mode if you want Unified Communications Manager
and endpoint to operate as the multi-prime source for provisioning endpoint data. If Unified CM and
Endpoint is the configured mode, any update made via an endpoint locally is synched with the Unified
CM , and any change made on Unified CM syncs to the endpoint.
• Unified CM—Use this mode if you want Unified Communications Manager to operate as the centralized
primary source for provisioning endpoint data and does not want to accept any configurations done from
the endpoints locally.
• Endpoint—Use this mode if you want endpoints to operate as the centralized primary source of
configuration data. In this mode, the endpoint maintains existing settings, ignores any configuration data
from Unified Communications Manager, and doesn’t sync back the changes done locally. This mode is
typically used when an Audiovisual (AV) integrator is installing the endpoints and wants to control
configuration from the endpoint.
Note If you want to maintain existing settings on the endpoint, it's recommended to choose Endpoint
mode, at least until after the endpoint has completed the registration process in full. You can
switch the configuration to one of the other modes after you complete this procedure.
Step 7 Configure any phone settings that you want. For more information on the fields and their configuration options,
see Online Help.
Step 8 Click Save.
The provisioned settings in Unified Communications Manager download to the endpoint.
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Note If you clicked Save mistakenly in the Phone Configuration window prior to device registration, the existing
Advanced Configuration settings from the endpoint will not load to Unified CM when the device registers.
To recover, do the following prior to device registration:
• In Unified CM, set the Configuration Control Mode to Endpoint and click Save.
• Let the phone register to Unified CM.
• After registration, return to the device configuration in the Phone Configuration window and click the
Get Config from Device button. The setting results in the existing Advanced Configurations on the
phone getting pulled into Unified CM. Note that this button does not appear until after device registration.
• Return to Step 6 of the procedure in order to complete the configuration.
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PA R T XIV
Advanced Call Processing
• Configure Call Control Discovery, on page 675
• Configure External Call Control, on page 685
• Configure Call Queuing, on page 695
• Configure Call Throttling, on page 707
• Configure Logical Partitioning, on page 711
• Configure Location Awareness, on page 721
• Configure Flexible DSCP Marking and Video Promotion, on page 729
• Separate Calling Party Number and Billing Number in SIP, on page 737
• SIP OAuth Mode, on page 753
CHAPTER 55
Configure Call Control Discovery
• Call Control Discovery Overview, on page 675
• Call Control Discovery Prerequisites, on page 675
• Call Control Discovery Configuration Task Flow, on page 675
• Call Control Discovery Interactions, on page 682
• Call Control Discovery Restrictions, on page 683
Call Control Discovery Overview

Use Call Control Discovery (CCD) to advertise Unified Communications Manager information along with
other key attributes, such as directory number patterns. Other call control entities that use the Service
Advertisement Framework (SAF) network can use the advertised information to dynamically configure and
adapt their routing operations. All entities that use SAF advertise their directory number patterns along with
other key information. Other remote call control entities can learn the information from this broadcast and
adapt the routing operations of the call.
Call Control Discovery Prerequisites

• SAF-enabled SIP or H.323 intercluster (non-gatekeeper controlled) trunks
• Remote call control entities that support and use the SAF network; for example, other Unified
Communications Manager or Cisco Unified Communications Manager Express servers
• Cisco IOS routers that are configured as SAF forwarders
Call Control Discovery Configuration Task Flow

Procedure

Step 1 See the documentation that supports your Configure a Cisco IOS router as the SAF
Cisco IOS router. Cisco Feature Navigator forwarder.
(http://www.cisco.com/go/cfn) allows you to
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Call Control Discovery Configuration Task Flow

determine which Cisco IOS and Catalyst OS
software images support a specific software
release, feature set, or platform.
Step 2 Configure SAF Security Profile, on page 677 Configure the SAF security profile for the SAF
forwarder to provide a secure connection
between the SAF forwarder and Unified
Step 3 Configure SAF Forwarders, on page 678 Configure the SAF forwarders, which are
Cisco IOS routers configured for SAF. They
notify the local cluster when remote
call-control entities advertise their hosted DNs
patterns. In addition, the SAF forwarder
receives publishing requests from the local
cluster for each configured and registered trunk
that is configured; the publishing request
contains the Hosted DN patterns for the Cisco
Unified Communications Manager, the PSTN
failover configuration, the listening port for
the trunk, and, for SIP trunks, the SIP route
header field, which contains a URI for the
trunk.
Step 4 Configure SIP or H.323 Intercluster Trunks, Configure SIP or H.323 intercluster
on page 678 (non-gatekeeper controlled) trunks for SAF
support. The local cluster uses SAF-enabled
trunks that are assigned to the CCD requesting
service to route outbound calls to remote
call-control entities that use the SAF network.
Step 5 Configure Hosted DN Groups, on page 679 Configure hosted DN groups, which are
collections of hosted DN patterns. After you
assign a hosted DN group to the CCD
advertising service, the CCD advertising
service advertises all the hosted DN patterns
that are a part of the hosted DN group. You
can assign only one hosted DN group per CCD
advertising service.
Step 6 Configure Hosted DN Patterns, on page 679 Configure hosted DN patterns, which are
directory number patterns that belong to
Unified Communications Manager; the CCD
advertising service advertises these patterns to
other remote call-control entities that use the
SAF network. You associate these patterns
with hosted DN groups, which allow you to
easily associate multiple patterns to a CCD
advertising service.
676
Configure SAF Security Profile

Step 7 Configure the Advertising Service, on page Configure the call control discovery
680 advertising service, which allows Unified
Communications Manager to advertise the
hosted DNs for the cluster and the PSTN
failover configuration to remote call-control
entities that use the SAF network.
Step 8 Configure the Partition for Call Control Configure a call control discovery partition to
Discovery, on page 680 ensure that the learned patterns are inserted
into digit analysis under this partition.
Step 9 Configure the Requesting Service, on page 680 To ensure that your local cluster can detect
advertisements from the SAF network,
configure one call control discovery requesting
service to listen for advertisements from
remote call control entities that use the SAF
network. In addition, the CCD requesting
service ensures that learned patterns are
inserted into the digit analysis.
Step 10 Block Learned Patterns, on page 681 Block learned patterns that remote call control
entities send to the local Unified
Communications Manager. Perform this
procedure on learned patterns that you no
longer want to use.
Configure SAF Security Profile

Configure the SAF security profile for the SAF forwarder to provide a secure connection between the SAF
forwarder and Unified Communications Manager.
Tip Use the same username and password that you entered on the router (SAF forwarder).
Before you begin

Configure a Cisco IOS router as the SAF forwarder. (See the Cisco Feature Navigator at http://www.cisco.com/
%20go/cfn.)
Procedure
Step 1 From Cisco Unified CM Administration, choose Advanced Features > SAF > SAF Security Profile.
Step 2 Configure the fields on the SAF Security Profile Configuration window.
For more information on the fields and their configuration options, see the system Online Help.
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Configure SAF Forwarders
Step 3 Click Save.
Configure SAF Forwarders

Configure the SAF forwarders, which are Cisco IOS routers configured for SAF. They notify the local cluster
when remote call-control entities advertise their hosted DNs patterns. In addition, the SAF forwarder receives
publishing requests from the local cluster for each configured and registered trunk that is configured; the
publishing request contains the Hosted DN patterns for the Cisco Unified Communications Manager, the
PSTN failover configuration, the listening port for the trunk, and, for SIP trunks, the SIP route header field,
which contains a URI for the trunk.
Tip If more than one node appears in the Selected Cisco Unified Communications Managers pane, append @
to the client label value; otherwise, errors can occur if each node uses the same client label to register with
the SAF forwarder.
Procedure
Step 1 From Cisco Unified CM Administration, choose Advanced Features > SAF > SAF Forwarder.
Step 2 Configure the fields on the SAF Forwarder Configuration window.
Step 3 Click Save.
Configure SIP or H.323 Intercluster Trunks

Configure SIP or H.323 intercluster (non-gatekeeper controlled) trunks for SAF support. The local cluster
uses SAF-enabled trunks that are assigned to the CCD requesting service to route outbound calls to remote
call-control entities that use the SAF network.
Procedure

• For SIP trunks:
a. From the Trunk Service Type Type drop-down list, choose Call Control Discovery. You cannot
change the trunk service type after you select it from the drop-down list.
b. Click Next.
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Configure Hosted DN Groups
c. Configure the fields on the Trunk Configuration window. For more information on the fields and
• For intercluster (non-gatekeeper controlled) trunks:

a. Click Next.
b. Check the Enable SAF check box.
c. Configure the other fields on the Trunk Configuration window. For more information on the fields
Step 4 Click Save.
Configure Hosted DN Groups

Configure hosted DN groups, which are collections of hosted DN patterns. After you assign a hosted DN
group to the CCD advertising service, the CCD advertising service advertises all the hosted DN patterns that
are a part of the hosted DN group. You can assign only one hosted DN group per CCD advertising service.
Procedure
Step 1 From Cisco Unified CM Administration, choose Call Routing > Call Control Discovery > Hosted DN
Group.
Step 2 Configure the fields on the Hosted DN Groups Configuration window.
Step 3 Click Save.
Configure Hosted DN Patterns

Configure hosted DN patterns, which are directory number patterns that belong to Unified Communications
Manager; the CCD advertising service advertises these patterns to other remote call-control entities that use
the SAF network. You associate these patterns with hosted DN groups, which allow you to easily associate
multiple patterns to a CCD advertising service.
Procedure
Step 1 From Cisco Unified CM Administration, choose Call Routing > Call Control Discovery > Hosted DN
Patterns.
Step 2 Configure the fields on the Hosted DN Patterns Configuration window. For more information on the fields
and their configuration options, see the system Online Help.
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Configure the Advertising Service
Step 3 Click Save.
Configure the Advertising Service

Configure the call control discovery advertising service, which allows Unified Communications Manager to
advertise the hosted DNs for the cluster and the PSTN failover configuration to remote call-control entities
that use the SAF network.
Procedure
Step 1 From Cisco Unified CM Administration, choose Call Routing > Call Control Discovery > Advertising
Service.
Step 2 Configure the fields in the Advertising Service Configuration window. For more information on the fields
Step 3 Click Save.
Configure the Partition for Call Control Discovery

Configure a call control discovery partition to ensure that the learned patterns are inserted into digit analysis
under this partition.
Note The CCD partition does not appear under Call Routing > Class of Control > Partition in Cisco Unified
Communications Manager Administration.
Procedure
Step 1 From Cisco Unified CM Administration, choose Call Routing > Call Control Discovery > Partition.
Step 2 Configure the fields in the Call Control Discovery Partition Configuration window. For more information
on the fields and their configuration options, see the system Online Help.
Step 3 Click Save.
Configure the Requesting Service
Caution Updating the Learned Pattern Prefix or Route Partition fields can affect system performance. To avoid
system performance issues, we recommend that you update these fields during off-peak hours.
680
Block Learned Patterns
To ensure that your local cluster can detect advertisements from the SAF network, configure one call control
discovery requesting service to listen for advertisements from remote call control entities that use the SAF
network. In addition, the CCD requesting service ensures that learned patterns are inserted into the digit
analysis.
Procedure
Step 1 From Cisco Unified CM Administration, choose Call Routing > Call Control Discovery > Requesting
Service.
Step 2 Configure the fields in the Requesting Service Configuration window. For more information on the fields
Step 3 Click Save.
Configure your remote call control entity to use the SAF network. (See the documentation for your remote
call control entity.)
Block Learned Patterns

Block learned patterns that remote call control entities send to the local Unified Communications Manager.
Perform this procedure on learned patterns that you no longer want to use.
Before you begin

Configure your remote call control entity to use the SAF network. See the documentation that supports your
remote call control entity.
Procedure
Step 1 From Cisco Unified CM Administration, choose Call Routing > Call Control Discovery > Block Learned
Patterns.
Step 3 Configure one of the following fields:
• In the Learned Pattern field, enter the exact learned pattern that you want to block. You must enter the
exact pattern that you want Cisco Unified Communications Manager to block.
• In the Learned Pattern Prefix field, enter the prefix to block a learned pattern based on the prefix that
is prepended to the pattern.
Example:
For Learned Pattern, enter 235XX to block 235XX patterns.
Example:
For Learned Pattern Prefix, enter +1 to block patterns that use +1.
Step 4 In the Remote Call Control Entity field, enter the name of the remote call control entity that advertises the
pattern that you want to block.
681
Call Control Discovery Interactions
Step 5 In the Remote IP field, enter the IP address for the remote call control entity where you want to block the
learned pattern.
Step 6 Click Save.
Call Control Discovery Interactions

Table 66: Call Control Discovery Interactions
Feature Interaction
Alarms Cisco Unified Serviceability provides alarms to support the call control discovery
feature. For information about how to configure alarms, see the Cisco Unified
Serviceability Administration Guide at http://www.cisco.com/c/en/us/support/
products-maintenance-guides-list.html.
BLF Subscriptions For a user to subscribe BLF status of a SAF learned pattern, Unified
Communications Manager sends a SIP subscribe message over a SIP trunk to
the remote cluster.
This functionality is supported with only SAF-enabled SIP trunks.
Bulk Administration Tool In the Bulk Administration Tool, you can import and export the configuration
for SAF security profiles, SAF forwarder, CCD advertising service, CCD
requesting service, hosted DN groups, and hosted DN patterns.
Call Detail Records Unified Communications Manager supports redirecting onBehalfOf as

SAFCCDRequestingService with a redirection reason as
SS_RFR_SAF_CCD_PSTNFAILOVER, which indicates that the call is redirected
to a PSTN failover number.
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Call Control Discovery Restrictions
Feature Interaction
Incoming Called Party The H.323 protocol does not support the international escape character +. To
Settings ensure that the correct DN patterns are used with SAF and call control discovery
for inbound calls over H.323 gateways or trunks, you must configure the incoming
called party settings in the service parameter, device pool, H.323 gateway, or
H.323 trunk windows; that is, configure the incoming called party settings to
ensure that when a inbound call comes from a H.323 gateway or trunk, Unified
Communications Manager transforms the called party number back to the value
that was originally sent over the trunk or gateway.
For example, a caller places a call to +19721230000 to Unified Communications
Manager A.
Unified Communications Manager A receives +19721230000 and transforms
the number to 55519721230000 before sending the call to the H.323 trunk. In
this case, your configuration indicates that the international escape character +
should be stripped and 555 should be prepended for calls of International type.
For this inbound call from the trunk, Unified Communications Manager B receives
55519721230000 and transforms the number back to +19721230000 so that digit
analysis can use the value as it was sent by the caller. In this case, your
configuration for the incoming called party settings indicates that you want 555
to be stripped and +1 to be prepended to called party numbers of International
type.
Digest Authentication Unified Communications Manager uses digest authentication (without TLS) to
authenticate to the SAF forwarder. When Unified Communications Manager
sends a message to the SAF forwarder, Unified Communications Manager
computes the SHA1 checksum and includes it in the MESSAGE-INTEGRITY
field in the message.
QSIG The QSIG Variant and ASN.1 ROSE OID Encoding settings in the H.323
Configuration window are advertised by the CCD advertising service. These
settings affect decoding of QSIG messages for inbound tunneled calls; for call
control discovery, they do not affect outgoing calls.
The remote call-control entity determines whether QSIG tunneling is required
for outgoing calls over H.323 trunks. If the remote call-control entity advertises
that QSIG tunneling is required, the QSIG message is tunneled in the message
of the outgoing call, even if the H.323 Configuration window in Cisco Unified
CM Administration indicates that QSIG support is not required.

All clusters are limited to advertised or learned routes within the same autonomous system (AS).
683
684
CHAPTER 56
Configure External Call Control
• External Call Control Overview, on page 685
• External Call Control Prerequisites, on page 686
• External Call Control Configuration Task Flow, on page 686
• External Call Control Interactions, on page 692
• External Call Control Restrictions, on page 694
External Call Control Overview

External call control lets an adjunct route server make call routing decisions for Unified Communications
Manager by using the Cisco Unified Routing Rules Interface. When you configure external call control,
Unified Communications Manager issues a route request that contains the calling party and called party
information to the adjunct route server. That server receives the request, applies appropriate business logic,
and returns a route response that instructs your system on how to route the call and any additional call treatment
to apply.
The adjunct router influences how your system allows, diverts, or denies calls; modifies calling and called
party information; plays announcements to callers; resets call history so that adjunct voicemail and IVR servers
can properly interpret calling and called party information; and logs reason codes that indicate why calls were
diverted or denied.
External call control provides the following functions:
• Best Quality Voice Routing—The adjunct route server monitors network link availability, bandwidth
usage, latency, jitter, and MOS scores to ensure that calls are routed through voice gateways that deliver
the best voice quality to all call participants.
• Least Cost Routing—The adjunct route server is configured with carrier contract information such as
local access and transport area (LATA) and inter-LATA rate plans, trunking costs, and burst utilization
costs to ensure that calls are routed over the most cost effective links.
• Ethical Wall—The adjunct route server is configured with corporate policies that determine reachability,
for example, whether user 1 is allowed to call user 2.
685
External Call Control Prerequisites
External Call Control Prerequisites

This feature requires the Cisco Unified Routing Rules XML Interface, which directs your system on how to
handle calls.
For more information, see the Cisco Unified Routing Rules Interface Developers Guide (CURRI documentation)
at https://developer.cisco.com.
External Call Control Configuration Task Flow

Procedure

Step 1 Configure a Calling Search Space for External Configure a calling search space for your
Call Control, on page 687 system to use when the route server sends a
divert obligation. A calling search space
comprises an ordered list of route partitions that
you assign to devices. Calling search spaces
determine the partitions that calling devices
search when they attempt to complete a call.
Step 2 Configure an External Call Control Profile, on Configure an external call control profile to
page 688 provide the URIs for the adjunct route server,
a calling search space that is used for diverting
calls, a timer that indicates how long your
system waits for a response from the adjunct
route server, and so on.
Step 3 Assign a Profile to a Translation Pattern, on For the translated patterns that you want to use
page 688 with external call control, assign an external
call control profile to the pattern. When a call
occurs that matches the translation pattern, your
system immediately sends a call routing query
to an adjunct route server, and the adjunct route
server directs your system on how to handle the
call.
Step 4 (Optional) Import the Route Server Certificate If the route server uses HTTPS, import the
into the Trusted Store, on page 689 certificate for the route server into the trusted
store on your system node. You must perform
this task on each node in the cluster that can
send routing queries to the route server. If you
use HTTPS for the primary or secondary web
service URIs in the external call control profile,
your system uses certificates to mutually
authenticate through a TLS connection to the
configured adjunct route servers.
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Configure a Calling Search Space for External Call Control

Step 5 (Optional) Export the Self-Signed Certificate If the route server uses HTTPS, export the Cisco
to the Route Server, on page 689 Unified Communications Manager self-signed
certificate to the route server. You must perform
this task for each node in the cluster that can
send routing queries to the route server. To
ensure that the primary and redundant route
servers can authenticate with Cisco Unified
Communications Manager through HTTPS, you
must generate a self-signed certificate that you
can import to each adjunct route server that
sends directives to your system.
Perform this procedure for each node in the
cluster that can contact the primary and
redundant adjunct route server.
Step 6 (Optional) Configure the Chaperone Function, Configure chaperone functionality if your
on page 690 routing rules from the route server state that a
chaperone must monitor or record a call. A
chaperone is a designated phone user who can
announce company policies in the call, monitor
the call, and record the call.
Step 7 (Optional) Configure Customized Follow this procedure if your routing rules
Announcements, on page 691 require that an announcement is played for some
calls and you do not want to use the
Cisco-provided announcements.
Configure a Calling Search Space for External Call Control

Configure a calling search space for your system to use when the route server sends a divert obligation. A
calling search space comprises an ordered list of route partitions that you assign to devices. Calling search
spaces determine the partitions that calling devices search when they attempt to complete a call.
Procedure

687
Configure an External Call Control Profile

Partitions box.
Step 8 Click Save.
Configure an External Call Control Profile

Configure an external call control profile to provide the URIs for the adjunct route server, a calling search
space that is used for diverting calls, a timer that indicates how long your system waits for a response from
the adjunct route server, and so on.
Procedure
Step 1 From Cisco Unified CM Administration, choose Call Routing > External Call Control Profile.
• Click Find and then choose an existing external call control profile from the resulting list to modify the
settings for an an existing external call control profile, enter search criteria.
• Click Add New to add a new external call control profile.
Step 3 Configure the fields on the External Call Control Profile Configuration window. For more information on
the fields and their configuration options, see the system Online Help.
Step 4 Click Save.
Assign a Profile to a Translation Pattern

Configure an external call control profile to provide the URIs for the adjunct route server, a calling search
space that is used for diverting calls, a timer that indicates how long your system waits for a response from
the adjunct route server, and so on.
Procedure
• Click Find and then choose an existing translated pattern from the resulting list to modify the settings
for an existing translated pattern, enter search criteria, .
• Click Add New to add a new translated pattern.
Step 3 From the External Call Control Profile drop-down list, choose the external call control profile that you want
to assign to the pattern.
688
Import the Route Server Certificate into the Trusted Store
Step 4 Configure other fields as needed in the Translation Pattern Configuration window. For more information
on the fields and their configuration options, see the system Online Help.
Step 5 Click Save.
Import the Route Server Certificate into the Trusted Store

If the route server uses HTTPS, import the certificate for the route server into the trusted store on your system
node. You must perform this task on each node in the cluster that can send routing queries to the route server.
If you use HTTPS for the primary or secondary web service URIs in the external call control profile, your
system uses certificates to mutually authenticate through a TLS connection to the configured adjunct route
servers.
Procedure
Step 1 From Cisco Unified Operating System Administration, choose Security > Certificate Management.
Step 2 Click Upload Certificate.
Step 3 In the Upload Certificate popup window, click CallManager-trust from the Certificate Name drop-down
list, and browse to the certificate for the adjunct route server.
Step 4 After the certificate appears in the Upload File field, click Upload.
Step 5 (Optional) Perform this procedure again if your system can contact a redundant adjunct route server.
Export the Self-Signed Certificate to the Route Server

If the route server uses HTTPS, export the Unified Communications Manager self-signed certificate to the
route server. You must perform this task for each node in the cluster that can send routing queries to the route
server. To ensure that the primary and redundant route servers can authenticate with Unified Communications
Manager through HTTPS, you must generate a self-signed certificate that you can import to each adjunct
route server that sends directives to your system.
Perform this procedure for each node in the cluster that can contact the primary and redundant adjunct route
server.
Procedure
Step 1 From Cisco Unified Operating Administration, choose Security > Certificate Management.
Step 2 In the Certificate List window, click Generate New.
Step 3 From the Certificate Name drop-down list, choose CallManager.
Step 4 Click Generate New.
Step 5 From the Find and List Certificates window, choose the CallManager.pem certificate that you just created.
Step 6 After the certificate file data appears, click Download to download the certificate to a location that you can
use for exporting the certificate to the adjunct route server.
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Configure the Chaperone Function
Step 7 Export the certificate to each adjunct route server that sends directives.
Configure the Chaperone Function

Configure chaperone functionality if your routing rules from the route server state that a chaperone must
monitor or record a call. A chaperone is a designated phone user who can announce company policies in the
call, monitor the call, and record the call.
Unified Communications Manager provides the following capabilities to support chaperone functionality, as
directed by the adjunct route server:
• Redirect an incoming call to a chaperone, hunt group, or a list of chaperones.
• Provide a chaperone with the ability to record a call.
When the chaperone is connected to the caller or when the chaperoned conference is established, the Record
softkey or programmable line key (PLK) (depending on the phone model) is active on the phone so that the
chaperone can invoke call recording. Call recording occurs for only the current call, and call recording stops
when the current call ends. Messages that indicate the status of recording may display on the phone when the
chaperone presses the recording softkey or PLK.
Procedure
Step 1 For phones on which you want to enable recording, set the Built-in-Bridge to On in the Phone Configuration
window.
Step 2 Create a recording profile:
a) Choose Device > Device Settings > Recording Profile.
b) Create a Call Recording Profile for the phones that can record chaperoned conferences.
Step 3 Apply the recording profile to the line appearance.
Step 4 Add a SIP trunk to point to the recorder.
Step 5 Create a route pattern that points to the SIP trunk.
Step 6 Configure the following service parameters:
a) Play Recording Notification Tone to Observed Target
b) Play Recording Notification Tone to Observed Connected Target
Step 7 Assign the Standard Chaperone Phone softkey template to the phone that the chaperone uses.
Step 8 Perform the following steps from Call Routing > Directory Number for a new phone or from Device >
Phone if the phone is already configured:
a) Configure only one directory number (DN) for the chaperone phone.
b) For the DN on the chaperone phone, choose Device Invoked Call Recording Enabled from the Recording
Option drop-down list.
c) For the DN on the chaperone phone, enter 2 for the Maximum Number of Calls setting, and enter 1 for
the Busy Trigger setting.
Step 9 For Cisco Unified IP Phones that support the Record softkey, configure the Standard Chaperone Phone softkey
template so that only the Conference, Record, and End Call softkeys display on the phone in a connected
state.
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Configure Customized Announcements
Step 10 For Cisco Unified IP Phones that support the record programmable line key (PLK), configure the PLK in the
Phone Button Template Configuration window.
Step 11 (Optional) If you have more than one chaperone in your cluster, add the chaperone DN to the chaperone line
group that you plan to assign to the chaperone hunt list.
This step ensures that an available chaperone monitors the call.
Configure Customized Announcements

Follow this procedure if your routing rules require that an announcement is played for some calls and you do
not want to use the Cisco-provided announcements.
Tip Do not use embedded spaces for the announcement identifier.
If other language locales are installed, you can upload other .wav files for this announcement to use with those
locales.
Procedure
Step 1 From Cisco Unified CM Administration, choose Media Resources > Announcement.
• To add a new announcement:
a) Click Add New.
b) In the Announcement Identifier field, enter an announcement identifier.
c) In the Description, enter a description of the announcement.
d) From the Default Announcement drop-down list, choose a default Cisco-provided announcement if
desired.
e) Click Save.
• To upload a custom .wav file for the announcement:
a) Click Upload File.
b) From the Locale drop-down list, choose the locale language for the announcement.
c) Click Choose File, and then choose a .wav file to upload.
d) Click Upload File.
e) When the upload finishes, click Close to refresh the window and show the uploaded announcement.
691
External Call Control Interactions

Table 67: External Call Control Interactions
Feature Interaction
Best Call Quality Routing You can set up routing rules on the adjunct route server that determine which
gateway to use for a call, taking voice quality into consideration. For example,
gateway A provides the best voice quality, so it is used for the call. In this case,
the adjunct route server monitors network link availability, bandwidth usage,
latency, jitter, and mean opinion scores (MOS) to ensure that calls are routed
through voice gateways that deliver the best voice quality to all call participants.
Call Detail Records External Call Control functions can be displayed in call detail records; for
example, the call detail record can indicate whether the adjunct route server
permitted or rejected a call. In addition, the call detail record can indicate whether
Unified Communications Manager blocked or allowed calls during which it did
not receive a decision from the adjunct route server.
Call Forward External Call Control intercepts calls at the translation pattern level, while Call
Forward intercepts calls at the directory number level. External Call Control has
a higher priority than Call Forward; for calls that invoke Call Forward, Unified
Communications Manager sends a routing query to the adjunct route server if
the translation pattern is assigned to an External Call Control profile. Call Forward
is triggered only when the adjunct route server sends a Permit decision with a
Continue obligation to the Cisco Unified Communications Manager.
Note The Call Diversion Hop Count service parameter that supports
External Call Control and the Call Forward Call Hop Count
service parameter that supports Call Forward are independent of
each other.
Call Pickup When a phone user tries to pick up a call by using the Call Pickup feature, External
Call Control is not invoked; Unified Communications Manager does not send a
routing query to the adjunct route server for that portion of the call.
Chaperones A chaperone is a designated phone user who can announce company policies to
the call, monitor the call, and record the call, if required. Chaperone restrictions
exist so that the parties that are involved in the call cannot converse without the
presence of the chaperone.
692
Feature Interaction
Cisco Unified Mobility Unified Communications Manager allows the route decision from the adjunct
route server for the following Cisco Unified Mobility features:
• Mobile Voice Access
• Enterprise Feature Access
• Dial-via-Office Reverse Callback
Unified Communications Manager does not send a routing query for the following
Cisco Unified Mobility features:
• Cell pickup
• Desk pickup
• Session handoff
Conferences When a phone user creates a conference, External Call Control may be invoked
for the primary call and consultative call.
Directory Numbers When you configure directory numbers as four- or five-digit extensions (enterprise
extensions), you need to configure two translation patterns if on-net dialing
supports four or five digits. One translation pattern supports globalizing the calling
and called numbers, and a second translation pattern supports localizing the
calling and called numbers.
Do Not Disturb By default, the DND setting for the user takes effect when the user rule on the
adjunct route server indicates that the adjunct route server sent a continue
obligation. For example, if the adjunct route server sends a continue obligation,
and the user has DND-R enabled, Unified Communications Manager rejects the
call.
Emergency Call Handling Caution We strongly recommend that you configure a very explicit set of
patterns for emergency calls (for example, 911 or 9.11) so that the
calls route to their proper destination (for example, to Cisco
Emergency Responder or a gateway) without having to contact the
route server for instructions on how to handle the call.
Transfer When a phone user transfers a call, External Call Control may be invoked for
both the primary call and consultative call. However, Unified Communications
Manager cannot enforce any routing rules from the adjunct route server between
the party that transfers and the target of the transfer.
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External Call Control Restrictions
External Call Control Restrictions

Table 68: External Call Control Restrictions
Adding Parties The chaperone cannot use the phone to add parties to a conference after the
conference begins, because the call must be put on hold for the chaperone to add
parties.
The other parties on the conference may add additional parties to the conference.
The configuration for the Advanced Ad Hoc Conference Enabled service
parameter, which supports the Cisco CallManager service, determines whether
other parties can add participants to the conference. If the service parameter is
set to True, other parties can add participants to the conference.
Call Transfer The chaperone cannot use the phone to transfer the conference call to another
party.
Conference Log Out When the chaperone leaves the conference, the entire conference ends.
Conference Softkey After the chaperone creates a conference, the Conference softkey, if available,
is disabled on the phone.
Hold The chaperone cannot use the phone to put the conference call on hold.
Recording If the chaperone starts recording before the feature makes a consultative call to
the party that will join the conference, Unified Communications Manager suspends
recording while the chaperone makes the consultative call; recording resumes
after the conference is established.
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CHAPTER 57
Configure Call Queuing
• Call Queuing Overview, on page 695
• Call Queuing Prerequisites, on page 696
• Call Queuing Task Flow, on page 697
• Call Queuing Interactions, on page 703
• Call Queuing Restrictions, on page 704
• Performance and Scalability for Hunt Pilots with Call Queuing, on page 704
Call Queuing Overview

Unified Communications Manager provides Call Queuing to place callers in a queue until hunt members are
available to answer them. An administrator can set the default so callers receive an initial greeting announcement
before the call is extended to an agent or the default can be changed so the initial announcement plays only
after the caller is put in the queue followed by Music On Hold or Tone On Hold. If the caller remains in queue
for a specified period of time, a secondary announcement is played at a configured interval until the call can
be answered or until the maximum wait timer expires.
When an incoming call reaches the hunt pilot, the following functions are provided:
• A caller may be connected to an initial customizable greeting announcement before proceeding.
• If one or more line members are logged in to the hunt pilot and are in an idle state, and if no calls are
queued, the call is extended to the line member that has been idle for the longest period of time.
• If no line members answer a call, that caller is not placed in queue. The call is routed to a new destination
or disconnected, based on the setting When no hunt members answer, are logged in, or registered.
• If a line member does not answer a queue-enabled call, that line member is logged off the hunt group
only if the setting Automatically Logout Hunt Member on No Answer is selected in the Line Group
setting window.
• Calls are placed in queue only if all members are busy.
• A caller who is waiting in queue may hear Music On Hold and a repeating (customizable) periodic
announcement.
• After a line member becomes idle, the caller with the longest wait time across multiple hunt groups is
extended to the idle line member. If the idle line member does not answer the call, the caller is returned
to the previous position in the queue.
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Call Queuing Prerequisites
• If a queued call exceeds its maximum wait time or the maximum number of callers allowed in queue is
exceeded, the call can be routed to an alternate number or it can be disconnected, depending on how the
hunt pilot is configured. The alternate number can be one of the following:
• A hunt pilot DN with queuing either enabled or disabled
• A voicemail DN
• A line DN
• A shared DN
• Line members can display the queue status of their queue-enabled hunt pilots. The queue status display
provides the following types of information:
• Hunt pilot pattern
• Number of queued callers on each hunt pilot
• Longest waiting time
Call queuing works in conjunction with existing hunt pilots, but there are no changes in the behavior of the
hunting operation for either queuing or nonqueuing hunt pilots. Hunt pilots that have call queuing enabled
provide the following features:
• Queuing-enabled hunt pilot calls can only be received by line members one call at a time. Two
queuing-enabled hunt pilot calls cannot be offered to a line member. A line member can receive calls
directly to the DN or from non-queuing hunt pilots.
• Line members who do not answer calls that are routed by hunt pilots are automatically logged out. A
line member is automatically logged out of a device if the line member receives a queuing-enabled hunt
pilot call and does not answer the call before timeout occurs. In the case of a shared-line deployment,
all devices configured with the same shared line are logged out. You can configure this behavior from
the Line Group setting window by selecting Automatically Logout Hunt Member on No Answer. Line
members are logged out only if this check box is checked.
For information about Call Queuing monitoring or announcements monitoring, see Cisco Unified Real Time
Monitoring Tool Administration Guide.
You can configure the inbound calls to change to the connected call state before playing the queuing
announcement while the call is extended to a hunt member in the queuing-enabled hunt pilot.
Call Queuing Prerequisites

• Cisco IP Voice Media Streaming (IPVMS) Application, which should be activated on at least one node
in the cluster
• Cisco CallManager service that is running on at least one server in the cluster
• Cisco RIS Data Collector service that is running on the same server as the Cisco CallManager service
• Cisco Unified Communications Manager Locale Installer, if you want to use non-English phone locales
or country-specific tones
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Call Queuing Task Flow
Call Queuing Task Flow

Procedure

Step 1 Configure Announcements, on page 697 Configure announcements through uploading
.wav files.
Step 2 Configure Music On Hold, on page 698 Configure Music On Hold (MoH) audio source.
Step 3 Configure Hunt Pilot Queuing, on page 701 Enables call queuing hold option for the calls
in a queue until they are answered.
Step 4 Automatically Logout Hunt Member on No Allows line members to log off the hunt list
Answer, on page 703 automatically.
Configure Announcements
Cisco Unified Communications Manager allows you to:
• use the existing Cisco-provided announcements,
• change the message or tone that you want an announcement to play,
• insert custom announcement .wav files,
• assign the locale for the announcement,
• change the description for the announcement,
• change the message or tone that you want an announcement to play.
Feature announcements are used by specific features such as Music On Hold (MoH) in association with Hunt
Pilot call queuing or External Call Control.
There are up to 50 feature announcements available. These announcements can be Cisco-provided audio files
or uploaded custom .wav files.
All custom announcement .wav files must be uploaded to all servers in the cluster.
Procedure
Step 1 In Cisco Unified Communications Manager, select Media Resources > Announcements.
The Find and List Announcements window displays.
Step 2 Select a hyperlink to the announcement you want to use.
Example:
Hyperlink—Wait_In_Queue_Sample
You can edit the announcement description or choose a customized announcement if uploaded.
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Configure Music On Hold
Step 3 To upload a .wav file to use as a custom announcement, click Upload File.
The Upload File window opens.
Step 4 In the Upload File window, choose the locale, enter the filename or browse to select the .wav file, and click
Upload File.
The upload process begins, and may take a few minutes depending on the file. The Status is updated after
processing is complete.
Step 5 Click Close to close the upload window.

The Announcement Configuration window refreshes to update the uploaded file status.
Step 6 To play the customized announcement, ensure that the Enable check box is checked in the Announcement
by Locale pane in the Announcements Configuration window.
Step 7 After you make the changes in the Announcements Configuration window, click Save.
What to do next
You must upload the announcement on each node in the cluster, because the announcement files are not
propagated between servers in a cluster. Browse to Cisco Unified Communications Manager Administration
on each server in the cluster and repeat the upload process.
Configure Music On Hold

You can configure Music On Hold (MoH) to play an optional initial greeting announcement when a caller is
first put on hold and also to play a periodic repeating announcement. These announcements can use one of
the Cisco-provided audio files or a file that is uploaded into the system.
Perform the following procedure to add or update a Music On Hold audio source, to associate an existing
audio source with an audio stream number, or to upload a new custom audio source.
Procedure
Step 1 From the Cisco Unified Communications Manager, choose Media Resources > Music On Hold Audio
Source.
The Find and List Music On Hold Audio Sources window appears.
Step 2 To add a new Music On Hold audio source, click Add New. To update a Music On Hold audio source, locate
a specific Music On Hold audio source. Based on the search criteria you specify, the system displays search
results for the record that matches all the criteria.
Step 3 Enter the appropriate settings, as described in Audio Source Fields for Music On Hold, on page 699.
Step 4 Click Save.
The list box at the bottom of the window shows the new Music On Hold audio source. The MOH Audio
Source File Status pane shows the MOH audio translation status for the added source.
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Audio Source Fields for Music On Hold

Table 69: Music On Hold Audio Source Information
Field Description
MOH Audio Stream Use this field to choose the stream number for this MOH audio source. Click the
Number drop-down list and choose a value from the list. For existing MOH audio sources,
the value appears in the MOH Audio Source title.
MOH Audio Source File Use this field to choose the file for this MOH audio source. Click the drop-down
list and choose a value.
MOH Audio Source Enter a unique name in this field for the MOH audio source. This name includes
Name up to 50 valid characters, such as letters, numbers, spaces, dashes, dots (periods),
and underscores.
Allow Multicasting Check this check box to specify that the selected MOH audio source allows
multicasting.
MOH Audio Source File This pane displays the following information about the source file for the selected
Status MOH audio source:
• InputFileName
• ErrorCode
• ErrorText
• DurationSeconds
• DiskSpaceKB
• LowDateTime
• HighDateTime
• OutputFileList
• MOH Audio Translation completion date
Note OutputFileList includes information on ULAW, ALAW, G.729,

and Wideband wav files and status options.
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Table 70: Announcement Settings
Field Description
Initial Announcement Choose an initial announcement from the drop-down list.

Note To select MoH with no initial announcement, choose the Not
Selected option.
Click the View Details link to view the following Initial Announcement
information:
• Announcement Identifier
• Description
• Default Announcement
Note • Played by MOH server only when the Audio Source Allow
Multi-casting is not checked and the Initial Announcement
for queuing-enabled hunt pilot calls field is set to Play
announcement if call is queued.
• Played by ANN if Allow Multi-casting check box is checked
or if Initial Announcement for queuing-enabled hunt pilot
calls is set to Play announcement before routing to Hunt
Member.
Initial Announcement for Choose one of the following to determine when to play the initial announcement:
queuing-enabled hunt
• Play announcement before routing to Hunt Member
pilot calls
• Play announcement if call is queued
Periodic Announcement Choose a periodic announcement from the drop-down list.

Note To select MoH with no periodic announcement, choose the Not
Selected option.
Click the View Details link to view the following Periodic Announcement
information:
• Announcement Identifier
• Description
• Default Announcement
Note The MOH server always plays the periodic announcement regardless
of other settings.
Periodic Announcement Enter a value (in seconds) that specifies the periodic announcement interval.
Interval Valid values are 10 to 300. The default value is 30.
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Configure Hunt Pilot Queuing
Field Description
Locale Announcement Locale Announcement depends upon the locale installation package that has been
installed.
Note • Prompts played by MOH will use the setting for Locale
Announcement.
• Prompts played by ANN will use the User Locale of the calling
party.
Table 71: Music On Hold Audio Sources
Field Description
(list of MoH audio This list box shows the MOH audio source that you add. Select the audio stream
sources) number of an MOH audio source to configure that MoH audio source.
Audio source ID is an ID that represents an audio source in the Music On Hold
server. The audio source can include either a file on a disk or a fixed device from
which a source stream Music On Hold server obtains the streaming data. An
MOH server can support up to 51 audio source IDs. Each audio source, represented
by an audio source ID, can stream as unicast and multicast mode, if needed.
Note If you select <None> , the system default MoH audio source service
parameter (Default Network Hold MoH Audio Source ID) is used
for the MoH audio source.
Upload File To upload an MOH audio source file that does not appear in the drop-down list,
click Upload File. In the Upload File window, either enter the path of an audio
source file or navigate to the file by clicking Browse. After you locate the audio
source file, click the Upload File button to complete the upload. After the audio
file gets uploaded, the Upload Result window displays the result of the upload.
Click Close to close this window.
Note When you upload a file, the file is uploaded to the Unified
Communications Manager server and performs audio conversions
to create codec-specific audio files for MOH. Depending on the size
of the original file, processing may take several minutes to complete.
Note Uploading an audio source file to an MOH server uploads the file
only to one MOH server. You must upload an audio source file to
each MOH server in a cluster by using Cisco Unified
Communications Manager Administration on each server. MOH
audio source files do not automatically propagate to other MOH
servers in a cluster.

When a hunt pilot has more calls distributed through the call distribution feature than its hunt members can
handle at any given time, call queuing holds these calls in a queue until they can be answered.
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When queuing is enabled, both Forward Hunt No Answer and Forward Hunt Busy are automatically disabled.
Conversely, if Forward Hunt No Answer or Forward Hunt Busy is enabled, queuing is automatically disabled.
Procedure
Step 1 In Cisco Unified Communications Manager Administration, select Call Routing > Route/Hunt > Hunt
Pilot to configure hunt pilots.
Step 2 Select the hunt pilot that you need to configure for Queuing.
Step 3 Navigate to the Queuing section of the Hunt Pilot Configuration window.
Step 4 Check the Queue Calls check box to enable queuing.
Step 5 Choose a Music On Hold (MoH) source from the drop-down list box to be used to play announcements and
provide queue hold treatments.
The MoH source can be configured as unicast or multicast. The caller-side Media Resource Group List (MRGL)
takes precedence for multicast or unicast.
If you do not select a source, the default Network Hold MoH/MoH Source and Announcements is used.
The MoH source announcement locale is used to determine the language used for the announcement. Only
one type of language announcement can be played per hunt pilot.
Step 6 In the Maximum Number of Callers Allowed in Queue field, enter an integer value for the number of callers
allowed in the queue for this hunt pilot.
The default value is 32. The field range is from 1 to 100.
Step 7 Choose one of the following options when the maximum number of callers in the queue is reached:
• If you want subsequent calls to be disconnected, select Disconnect the call.
• If you want subsequent calls to be routed to a secondary destination, select Route the call to this
destination. Provide a specific device DN, shared line DN, or another hunt pilot DN.
• (Optional) You may also select Full Queue Calling Search Space from the drop-down list. Used to
determine which partition to search when attempting to complete a call.
Step 8 In the Maximum Wait Time in Queue field, enter an integer value to set the maximum wait time, in seconds,
in a queue.
The default value is 900 seconds. The field range is from 10 to 3600 seconds.
Step 9 Choose one of the following options when the maximum wait time is reached:
• If you want that call to be disconnected, select Disconnect the call.
• If you want that call to be routed to a secondary destination, select Route the call to this destination.
Provide a specific device DN, shared line DN, or another hunt pilot DN.
• (Optional) You may also select Maximum Wait Time Calling Search Space from the drop-down list.
Used to determine which partition to search when attempting to complete a call.
Step 10 When no line members are logged in or registered at the time of an incoming call, choose one of the following
options:
• If you need that call to be disconnected, select Disconnect the call.
• If you need that call to be routed to a secondary destination, select Route the call to this destination.
Provide a specific device DN, shared line DN, or another hunt pilot DN.
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Automatically Logout Hunt Member on No Answer
• (Optional) You may also select No hunt members logged in or registered Calling Search Space from
the drop-down list. Used to determine which partition to search when attempting to complete a call.
Step 11 Click Save.
Automatically Logout Hunt Member on No Answer

Allows line members to log off the hunt list automatically. If an agent does not answer a queuing-enabled
hunt pilot call, that agent will be logged off of the hunt group and will not receive additional hunt pilot calls
unless he presses the "HLOG" soft key on the phone to log into the hunt pilot.
Line members can log back in using the "HLOG" softkey or PLK.
Procedure
Step 1 In Cisco Unified Communications Manager Administration, choose Call Routing > Route/Hunt > Line
Group to configure line groups.
Step 2 Choose the line group that you need to configure from the Find and List Line Groups window.
Step 3 Navigate to the Hunt Options section of the Line Group Configuration window.
Step 4 Ensure that the Automatically Logout Hunt Member on No Answer check box is checked.
Step 5 Click Save.
Call Queuing Interactions

Feature Interaction
SIP Rel1XX Options If a call is routed to a queuing-enabled hunt pilot through SIP ICT, the SIP ICT
uses the SIP profile that has SIP Rel1XX Options set to Send PRACK if 1XX
contains SDP. As a result, the initial announcement is played to every call before
the call is extended to the line member.
The above existing interaction for SIP ICT does not apply if Connect Inbound
Call before Playing Queuing Announcement checkbox is checked under
DeviceDevice Settings SIP Profile > Trunk Specific Configuration in Cisco
If Connect Inbound Call before Playing Queuing Announcement checkbox
is not checked the interaction for SIP ICT remains the same. However, it does
not guarantee the initial announcement can always be heard by a caller from the
PSTN side. The initial announcement will not be heard by a caller from the PSTN
side if the PSTN provider doesn't open the voice path until a Connect message
is received on the call.
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Call Queuing Restrictions
Feature Interaction
Hunt Pilots and Hunt • The logoff notification functionality for hunt groups changes when Call
Groups Queuing is enabled for a hunt pilot. If Call Queuing is enabled for a hunt
pilot, the Hunt Group Logoff Notification does not play when users log out
of a hunt group or are logged off because they missed their turn in the queue.
• If the hunt list has multiple line groups, these line groups must have the
same setting for Automatically Logout Hunt Member on No Answer.
• Hunt Pilot still queues calls, even when all hunt members are logged out.The
line group members should not be added in more than one line group and
even if they are added in second line groups, those second line groups should
not be in the same Hunt list.
• All hunt options must be set to Try Next Member, then Try Next Group in
the hunt list.
Call Queuing Restrictions

The following general restrictions apply to call queuing:
• H.323 Fast Start does not support Call Queuing.
• Queue status PLK is supported only with the following LCD display phones for both SCCP and SIP:
6921, 6941, 6945, 6961, 7911G, 7931G, 7942G, 7945G, 7962G, 7965G, 7975G, 8961, 8945, 8941, 9951,
9971, 7800 and 8800 series.
• Log Out of Hunt Groups (HLog) is not compatible with Cisco Extension Mobility Cross Cluster (EMCC);
Call Queuing should not be deployed with EMCC.
• Unified Communications Manager does not support Unified Mobility with Call Queuing.
• In a H323 to SIP interworking scenario, the user may not hear initial announcement, MoH, periodic
announcement or observe call failure in a native call queuing flow due to interworking delays. In such
a scenario it is advised to use only SIP protocol.
Performance and Scalability for Hunt Pilots with Call Queuing

The following performance and scalability restrictions apply:
• A single Unified CM Cluster supports a maximum of 15,000 hunt list devices.
• A single Unified CM Subscriber supports a maximum of 100 hunt pilots with call queuing enabled per
node
• Hunt list devices may be a combination of 1500 hunt lists with ten IP phones in each hunt list, 750 hunt
lists with twenty IP phones in each hunt list, or similar combinations
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Note When using the broadcast algorithm for call coverage, the number of hunt list
devices is limited by the number of busy hour call attempts (BHCA). Note that
a BHCA of 10 on a hunt pilot pointing to a hunt list or hunt group containing 10
phones and using the broadcast algorithm is equivalent to 10 phones with a BHCA
of 10.
• The maximum number of hunt pilots is 100 per Unified CM subscriber node with call queue enabled
when configured with 32 callers which is allowed in the queue. The total number of queue slots per node
(the value of "Maximum Number of Callers Allowed in Queue" for all Call Queuing Enabled Hunt Pilots
on the node combined) is limited to 3200. The maximum number of simultaneous callers in a queue for
each hunt pilot is 100, meaning 100 callers per hunt pilot is allowed in a queue and the maximum number
of hunt pilots is reduced to 32. The maximum number of members across all hunt lists does not change
when call queuing is enabled.
• The maximum wait time in queue for each hunt pilot that you can configure ranges from 0 to 3600
seconds (default 900). An increase in the number of hunt lists can require you to increase the dial plan
initialization timer that is specified in the Unified Communications Manager service parameters. We
recommend that you set the dial plan initialization timer to 600 seconds if you have 1500 hunt lists
configured.
• We recommend having no more than 35 directory numbers for a single line group when using broadcast
algorithms with call queuing. Additionally, the number of broadcast line groups depends on the busy
hour call completion rate (BHCC). If there are multiple broadcast line groups in a Unified CM system,
the number of maximum directory numbers in a line group must be less than 35. The number of busy
hour call attempts (BHCA) for all the broadcast line groups should not exceed 35 calls set up per second.
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706
CHAPTER 58
Configure Call Throttling
• Call Throttling Overview, on page 707
• Call Throttling Configuration Task Flow, on page 708
Call Throttling Overview

Call Throttling allows your system to automatically throttle or deny new call attempts. The system takes this
action when conditions cause users to experience a delay in the interval between going off hook and receiving
a dial tone.
Some factors that can result in this delay are as follows:
• Heavy call activity
• Low CPU availability
• Routing loops
• Disk I/O limitations
• Disk fragmentation
The system uses the values that are specified in the call throttling parameters to determine a possible delay
to dialtone and also to determine when conditions no longer require call throttling.
When throttling is necessary to prevent excessive delay to dialtone, the system enters a Code Yellow state
and new call attempts are throttled (denied).
When the system calculates the delay to dialtone as being over the threshold that is configured in the call
throttling service parameters, Unified Communications Manager rejects new calls. When call throttling
activates, a user who attempts a new call receives a reorder tone and, depending on the phone model, may
also receive a prompt on the phone display.
Call throttling effectively prevents the type of excessive delays that can cause a user to complain to the system
administrator or question whether the system is down or the phone is broken. Your system constantly monitor
the system to anticipate when such latency could occur.
When the delay to dialtone is within the guidelines of the call throttling service parameters, Unified
Communications Manager stops throttling calls by exiting the Code Yellow state and new calls are again
allowed.
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Call Throttling Configuration Task Flow
Call Throttling Configuration Task Flow

Procedure

Step 1 Configure Call Throttling, on page 708 Enables Call throttling automatically when your
system detects conditions such as heavy call
activity, low CPU availability, and disk
fragmentation.
Step 2 Configure Memory Throttling, on page 708 Configures memory throttling for your system.
Configure Call Throttling

Call throttling occurs automatically when your system detects conditions such as heavy call activity, low CPU
availability, and disk fragmentation. The system automatically exits throttling when these conditions are fixed.
Call Throttling is configured via advanced service parameters. For many deployments, the default settings
are sufficient.
Caution We recommend that you not modify call throttling parameters unless advised to do so by customer support.
Procedure
Step 5 Under Call Throttling, configure values for the cll throttling service parameters. For parameter help
descriptions, click the parameter name in the GUI.
• Code Yellow Entry Latency
• Code Yellow Exit Latency Calendar
• Code Yellow Duration
• Max Events Allowed
• System Throttle Sample Size
Step 6 Click Save.
Configure Memory Throttling

Use this procedure to configure memory throttling for your system.
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Procedure
Step 2 From the Server drop-down list, select a Unified Communications Manager server.
Step 5 Set the Enable Memory Throttling parameter to True.
Step 6 Configure values for the additional service parameters in the Memory Throttling area. For parameter help,
click the parameter name in the GUI.
Step 7 Click Save.
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710
CHAPTER 59
Configure Logical Partitioning
• Logical Partitioning Overview, on page 711
• Logical Partitioning Configuration Task Flow, on page 711
• Logical Partitioning Interactions, on page 718
• Logical Partitioning Restrictions, on page 719
Logical Partitioning Overview

With logical partitioning, you can support PSTN and VoIP calls on a single system while meeting regulatory
requirements for call separation. For example, under regulatory constraints in India, all calls that are received
from or sent to an external phone must be handed off to and carried by a local or long-distance service provider
over the full length of the connection, with the applicable toll charges. You can create a single Unified
Communications Manager cluster that routes calls appropriately to the PSTN or the VoIP network according
to the caller's location and the phone number being called.
logical partitioning defines which sets of VoIP devices are allowed to communicate with each other. Users
do not have to remember to use one line for PSTN and one line for VoIP. Phones making off-net calls are
only allowed to talk to a PSTN gateway. It's like having two networks to separately handle your VoIP and
PSTN calls, but without the expense of dual infrastructure.
Logical Partitioning Configuration Task Flow

Procedure

Step 1 Enable Logical Partitioning, on page 712 Enable Logical Partitioning.
Step 2 To Configure Geolocations, on page 712, Configuring geolocations is a two-step process:

perform the following subtasks: defining locations and assigning them to
devices. You also can set the default location
• Create Geolocations, on page 713
to be used by all devices in the cluster.
• Assign Geolocations, on page 713
• Set the Default Geolocation, on page 714
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Enable Logical Partitioning

Step 3 Configure a Logical Partitioning Default Policy, Set up a default policy for devices that are not
on page 714 associated with a geolocation or geolocation
filter. The policy allows or denies PSTN calls
between these devices.
Step 4 Configure Devices to Avoid Logical You can specifically exempt devices and device
Partitioning Checks, on page 714 pools from the partitioning checks.
Step 5 To Configure Geolocation Filters, on page 715, Logical partitioning assigns a unique identifier
perform the following subtasks: to each device based on its location. When one
device calls another, these identifiers are used
• Create Geolocation Filter Rules, on page
to determine whether the call is allowed and
715
what routing is appropriate. You can choose
• Assign Geolocation Filters, on page 716 which fields are used to create this identifier.
• Set the Default Geolocation Filter, on page For example, you can apply different policies
716 based on the room or floor within a building.
Step 6 Define a Set of Logical Partitioning Policy Define a set of logical partitioning policies for
Records, on page 717 allowing or denying calls between geolocations.
Before calls between geolocations are allowed
to proceed, the system checks to be sure that
calls are allowed between the specified
geolocations based on these policies.
Step 7 (Optional) Enable Location Conveyance, on Configure location conveyance if you want to
page 717 communicate geolocation information about
devices across clusters.
Enable Logical Partitioning

Use this procedure to turn on the Logical Partitioning feature.
Procedure
Step 2 For the Enable Logical Parititioning enterprise parameter, choose True from the drop-down list.
Step 3 Click Save.
Configure Geolocations
Configuring geolocations is a two-step process: defining locations and assigning them to devices. You also
can set the default location to be used by all devices in the cluster.
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Create Geolocations
Procedure

Step 1 Create Geolocations, on page 713 Configure geolocations to specify geographic
locations. These are used to associate devices
with regulatory features such as logical
partitioning. Geolocations are used in policy
decisions, such as in-country regulations.
Step 2 Assign Geolocations, on page 713 Assign a geolocation to a device or device pool.
Step 3 Set the Default Geolocation, on page 714 Specify a default geolocation for all devices and
device pools in this cluster.
Create Geolocations
Use this procedure to create geolocations that you can assign to the devices in your system. You can use the
geolocations for logical partitioning.
Procedure
Step 1 From Cisco Unified CM Administration, choose System > Geolocation Configuration.
Step 3 Enter a Name for the geolocation.
Step 4 Configure the fields on the Geolocation Configuration window. For more information on the fields and their
Step 5 Click Save.
Step 6 Repeat this procedure to create additional geolocations.
Assign Geolocations
Assign a geolocation to a device or device pool.
Procedure
Step 1 From Cisco Unified CM Administration, choose one of the following menu items:
• Device > Phone
• Device > Trunk
• Device > Gateway
• System > Device Pool

• Click Find to modify the settings for an existing device or device pool. Enter search criteria, and then
choose an existing device or device pool from the resulting list.
713
Set the Default Geolocation
• Click Add New to add a new device or device pool. For devices, choose device types and protocols as
needed and click Next.
Step 3 From the Geolocation drop-down list, choose a geolocation that you configured.
Step 4 Click Save.
Set the Default Geolocation

Specify a default geolocation for all devices and device pools in this cluster.
Procedure
Step 2 From the Default Geolocation drop-down list, choose a Geolocation that you configured. The default value
is Unspecified.
Step 3 Click Save.
Step 5 (Optional) If you need to override this default for a specific device or device pool, enter the value on either
the Device Configuration or Device Pool Configuration window, and then click Save.
Configure a Logical Partitioning Default Policy

Set up a default policy for devices that are not associated with a geolocation or geolocation filter. The policy
allows or denies PSTN calls between these devices.
Procedure
Step 1 From Cisco Unified CM Administration, choose Call Routing > Logical Partitioning Policy Configuration
Step 3 Configure the fields on the Logical Partition Policy Configuration window. For more information on the
Step 4 Click Save.
Note If a policy that contained the value Allow is then later changed to Deny, then it remains Deny.
The opposite is also true. A policy previously set to Deny, later changed to Allow is an Allow.
The Cisco Unified Reporting > Geolocation Policy Report can help you identify policies that
overlap.
Configure Devices to Avoid Logical Partitioning Checks

You can specifically exempt devices and device pools from the partitioning checks.
714
Configure Geolocation Filters
Procedure
• Device > Phone
• Device > Trunk

• Click Find to modify the settings for an existing device or device pool. Enter search criteria and then
Step 3 From the Geolocation drop-down list, choose Unspecified.

Step 4 Click Save.
Configure Geolocation Filters

Logical partitioning assigns a unique identifier to each device based on its location. When one device calls
another, these identifiers are used to determine whether the call is allowed and what routing is appropriate.
You can choose which fields are used to create this identifier. For example, you can apply different policies
based on the room or floor within a building.
Procedure

Step 1 Create Geolocation Filter Rules, on page 715 Geolocation filters allow you to specify which
fields are used to create a geolocation identifier.
This feature is used to make policy decisions
on a subset of the geolocation objects.
Step 2 Assign Geolocation Filters, on page 716

Step 3 Set the Default Geolocation Filter, on page 716 Configure the Default Geolocation Filter
enterprise parameter to specify a default
geolocation filter for a cluster. This parameter
determines the default geolocation filter setting
for all devices and device pools that are not
associated with a geolocation filter.
Create Geolocation Filter Rules

Use this procedure to create geolocation filters that you can use for logical partitioning decisions.
715
Assign Geolocation Filters
Procedure
Step 1 From Cisco Unified CM Administration, choose System > Geolocation Filter.
Step 3 Enter a Name and Description for the filter.
Step 4 Check the check boxes that correspond to the items you want to use for logical partitioning decisions.
Step 5 Configure the fields on the Geolocation Filter Configuration window. For more information on the fields
Step 6 Click Save.
Step 7 Repeat these steps to create additional geolocation filters.
Assign Geolocation Filters
Procedure
• Device > Phone
• Device > Trunk

• Click Find to modify the settings for an existing device or device pool. Enter search criteria and then
Step 3 From the Geolocation Filter drop-down list, choose a geolocation filter that you configured.
Step 4 Click Save.
Set the Default Geolocation Filter
Procedure
Step 2 From the Default Geolocation drop-down list, choose a Geolocation that you configured. The default value
is Unspecified.
Step 3 Click Save.
716
Define a Set of Logical Partitioning Policy Records
Step 5 (Optional) If you need to override this default for a specific device or device pool, specify the default
geolocation filter value on either the Device Configuration or Device Pool Configuration window, and then
click Save.
Define a Set of Logical Partitioning Policy Records

Define a set of logical partitioning policies for allowing or denying calls between geolocations. Before calls
between geolocations are allowed to proceed, the system checks to be sure that calls are allowed between the
specified geolocations based on these policies.
Procedure
Step 1 From Cisco Unified CM Administration, choose Call Routing > Logical Partitioning Policy Configuration.
• Click Find to modify the settings for an existing logical partitioning policy. Enter search criteria and
then choose an existing logical partitioning policy from the resulting list.
• Click Add New to add a new logical partitioning policy.
Step 3 Configure the fields on the Logical Partitioning Policy Configuration window. For more information on
the fields and their configuration options, see the system Online Help.
Note If any policy is left blank without any configuration values, it will become a blank geolocation
policy and configuring a Logical Policy for a specific Device Type with the blank Logical
Partitioning configurations makes Unified Communications Manager add the policy value (Allow
or Deny) in the configured device type.
Step 4 Click Save.
Enable Location Conveyance

Location Conveyance is an optional configuration that lets you share geolocation information across clusters.
Procedure

• Click Find and select an existing trunk.
• Click Add New to configure a new trunk.
Step 3 Complete the fields in the Trunk Configuration window. For more information on the fields and their
Step 4 In the Geolocation Information area, select a Geolocation and Geolocation Filter.
Step 5 To enable Location Conveyance, check the Send Geolocation Information check box.
717
Logical Partitioning Interactions
Step 6 Click Save.
Logical Partitioning Interactions

Table 72: Logical Partitioning Interactions
Feature Interaction
Ad Hoc Conference, Join, Logical partitioning handling does not take place in the following circumstances:
Join Across Lines, Call
• When all participants are VoIP phones.
Forwarding, Call Transfer
• When the geolocation or geolocation filter does not associate with a device.
Barge, cBarge, and Logical partitioning handling does not take place in the following circumstances:
Remote Resume
• When both the caller and the callee devices are VoIP phones, logical
partitioning policy checks are ignored.
• For the participants in cBarge/Barge, no logical partitioning policy checking
exists, and you cannot prevent logical-partitioning-denied scenarios.
Cisco Unified Mobility Logical partitioning handling does not take place in the following circumstances:
• Geolocation or geolocation filter does not associate with the involved devices.
• No logical partitioning support exists when a dual-mode phone is used.
CTI Handling Logical partitioning handling does not take place in the following circumstances:
• When a geolocation or geolocation filter does not associate with any device,
handling does not occur.
• When all the involved devices specify VoIP phones, handling does not occur.
Extension Mobility Logical partitioning handling does not take place in the following circumstances:
• A geolocation or geolocation filter does not associate with a VoIP phone
that is logged on to Cisco Extension Mobility, nor does it associate with the
calling party or called party device.
• The VoIP phone that is logged on to Cisco Extension Mobility calls or
receives a call from a VoIP phone.
Meet-Me Conference Logical partitioning handling does not take place in the following circumstances:
• When all participants are VoIP phones, handling does not occur.
• When geolocation or geolocation filter does not associate with a device, no
policy check takes place for that device.
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Logical Partitioning Restrictions
Feature Interaction
Route Lists and Hunt Logical partitioning handling does not take place in the following circumstances:
Pilots
• When both the calling party and called party devices are VoIP phones,
• All devices must associate with both a geolocation and geolocation filter.
If any device does not associate with both geolocation and geolocation filter,
Shared Line Logical partitioning handling does not take place in the following circumstances:
• When both the caller and the callee devices are VoIP phones, no handling
occurs.
• When geolocation or geolocation filter does not associate with any device,
no handling occurs.

Table 73: Logical Partitioning Restrictions
Barge/cBarge Barge/cBarge does not occur; the call instance is dropped.

For the participants in cBarge/Barge, no logical partitioning policy checking
exists, and you cannot prevent logical-partitioning-denied scenarios.
BLF Presence BLF Presence notifications are not checked for a logical partitioning policy.
Cisco Extension Mobility When Cisco Extension Mobility logs in to a phone in a different geolocation,
outgoing PSTN calls can occur when Local Route Groups are configured.
Incoming PSTN calls are not placed to the phone but receive a reorder tone.
Cisco Unified The system does not support the logical partitioning feature for calls that involve
MeetingPlace Cisco Unified MeetingPlace or Cisco Unified MeetingPlace Express.
Conferences The logical partitioning checks are not supported for participants across
conferences in conference chaining.
For example, meet-me and adhoc chained conferences can have participants that
are logical partitioning denied.
H.225 Cisco Unified Communications Manager does not communicate geolocation

gatekeeper-controlled information over a H.225 gatekeeper-controlled trunk.
trunk
719
H.323 and MGCP Cisco Unified Communications Manager does not communicate geolocation info
Gateways to H.323 or MGCP gateways.
Communication to a SIP gateway can be disabled through the SIP trunk check
box.
Mobility Cell Pickup Logical partitioning deny handling takes place after call is answered on the mobile
phone.
The logical partitioning policy check does not occur before the call is placed to
the mobile phone (as it happens for a basic SNR call). The system checks the
logical partitioning policy after the mobile phone answers the call.
Q.SIG intercluster trunk Intercluster trunks (ICT) with the Q.SIG protocol are not allowed to communicate
geolocation inforomation for the caller or receiving device. The ICT configuration
for “Send Geolocation Information” is disabled when the Q.SIG tunneled protocol
is selected.
Reorder Tones No reorder tone (fast busy tone) is provided on IOS H.323 and SIP gateways
upon release of connected calls due to logical partitioning policies.
Shared Line Active Call For a restricted logical partitioning scenario, the shared line drops the active call
information for the duration of the call, even if a feature moves the shared-line
call to the allowed category.
User Agent Server The logical partitioning policy checks in the logical partitioning-aware cluster
that receives this geolocation may cancel the call if the policy is denied.
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CHAPTER 60
Configure Location Awareness
• Location Awareness Overview, on page 721
• Location Awareness Prerequisites, on page 723
• Location Awareness Configuration Task Flow, on page 723
• Location Awareness Restrictions, on page 727
Location Awareness Overview
Important Meraki Access Points support for Location Awareness is applicable only from Release 12.5(1)SU6 onwards
and Release 14SU1 onwards.
Location Awareness allows administrators to determine the physical location from which a phone connects
to the company network. For wireless networks, you can view the wireless access point infrastructure, and
which mobile devices currently associate to those access points. For wired networks, you can view the Ethernet
switch infrastructure and see which devices are currently connected to those switches. This allows you to
determine the building, floor, and cube from which a call was placed.
Note Currently, wired phones do not support Location Awareness.
You can view your network infrastructure from Cisco Unified CM Administration > Advanced Features >
Device Location Tracking Services > Switches and Access Points > Find and List Switches and Access
Points window.
This feature updates the Unified Communications Manager database dynamically with the following
information:
• Network infrastructure devices such as switches and wireless access points, including IP addresses,
hostnames, and BSSID info (where applicable) for each infrastructure device.
• Associated endpoints for each infrastructure device, including:
• For wireless networks, the list of devices that are currently associated to a wireless access point.
• For wired networks, the list of devices and device types that are currently connected to an ethernet
switch.
721
Wireless Network Updates
Cisco Emergency Responder Integration

Location Awareness helps integrated applications such as Cisco Emergency Responder to determine the
physical location of a user who places an emergency call. When Location Awareness is enabled, Cisco
Emergency Responder learns of a new device to infrastructure association within minutes of a mobile device
associating with a new wireless access point, or a desk phone being connected to a new ethernet switch.
When Cisco Emergency Responder first starts up, it queries the Unified Communications Manager Database
for the current device to network infrastructure associations. Every two minutes following, the Cisco Emergency
Responder checks for updates to the existing associations. As a result, even if a mobile caller places an
emergency call while in a roaming situation, Cisco Emergency Responder can quickly determine the physical
location of the caller and send emergency services to the appropriate building, floor, or cube.
Wireless Network Updates

To enable Location Awareness for your wireless infrastructure, you can configure Unified Communications
Manager to synchronize with a Cisco Wireless LAN Controller. You can synchronize Unified Communications
Manager with up to fifty controllers. During the synchronization process, Unified Communications Manager
updates its database with the access point infrastructure that the controller manages. In Cisco Unified CM
Administration, you can view the status for your wireless access points, including the list of mobile clients
that are associated to each access point.
As mobile clients roam between access points, SIP and SCCP signaling from the endpoint communicates the
new device to access point association to Unified Communications Manager, which updates its database.
Cisco Emergency Responder also learns of the new association by querying the Unified Communications
Manager database every few minutes for new endpoints that have changed their association. As a result, if a
mobile client places an emergency call, Cisco Emergency Responder has accurate information on the physical
location of the user whom placed the call.
If you have a regular synchronization schedule for your Wireless Access Point controllers, Unified
Communications Manager adds and updates access points from the database dynamically following each
synchronization.
Using Bulk Administration to insert Access Points

If you are using a third-party wireless access point controller, or if you want to export your access points from
Cisco Prime Infrastructure, you can use the Bulk Administration Tool to bulk insert your wireless access point
infrastructure from a CSV file into the Unified Communications Manager database. Following the bulk insert,
the next location update from the mobile device updates the database with the current access point association.
However, Bulk Administration does not allow you to update your access point infrastructure dynamically as
new access points get added to your wireless network. If a mobile call gets placed through an access point
that was added after the bulk insert, that access point will not have a record in the database, Unified
Communications Manager will not be able to match the BSSID of the new access point, and will mark the
infrastructure for the wireless device as UNIDENTIFIED AP.
For detailed information on the Bulk Administration Tool, refer to the "Manage Infrastructure Devices" chapter
of the Bulk Administration Guide for Cisco Unified Communications Manager.
Supported Endpoints for Location Awareness

The following endpoints support tracking via Location Awareness:
• Cisco Uniifed Wireless IP Phone 7925G
722
Location Awareness Prerequisites
• Cisco Unified Wireless IP Phone 7925G-EX

• Cisco Unified Wireless IP Phone 7926G
• Cisco Jabber clients—supported as of 12.5(1)SU1
• Cisco Wireless IP Phone 8821—supported as of 12.5(1)SU1
• Webex App—supported as of 12.5(1)SU1
These endpoints provide upstream infrastructure information, such as BSSID, through Station Info messages
to Cisco Unified Communications Manager. Cisco Emergency Responder uses AXL Change Notifications
to track these devices through the associated access point.
For device tracking to work, wireless access points must be defined in Cisco Unified Communications Manager.
You can do this by syncing a wireless access point controller or using Bulk Administration to import wireless
access point infrastructure.
Location Awareness Prerequisites

This feature allows you to synchronize the Cisco Unified Communications Manager database with multiple
Cisco Wireless LAN Controllers. You must also set up your Cisco Wireless LAN Controller hardware and
your infrastructure of access points. For details, see your controller documentation.
Location Awareness Configuration Task Flow

Complete the following tasks to set up Location Awareness in Cisco Unified Communications Manager.
Before you begin
Procedure

Step 1 Start Services for Wireless Infrastructure In Cisco Unified Serviceability, start services
Synchronization, on page 724 that support the Location Awareness feature.
Step 2 Configure Wireless Access Point Controller, Synchronize the database with a Cisco wireless
on page 724 access point controller. The sync imports the
wireless infrastructure into the database.
Tip Set up a sync schedule for
automatic updates.
Step 3 Insert Infrastructure Devices, on page 725 Optional. If you want to add your wireless
infrastructure from Cisco Prime Infrastructure,
or if you are using a third-party wireless LAN
controller, use Bulk Administration to update
the database from a CSV file.
723
Start Services for Wireless Infrastructure Synchronization

Note This method does not allow you
to set up automatic updates.
Step 4 Deactivate Infrastructure Device from Tracking, Optional. If your synchronization includes
on page 726 access points that you do not want to track (for
example, if the synchronization pulls in access
points from a lab), you can deactivate the access
point and Cisco Unified Communications
Manager will not track updates to the access
point.
Start Services for Wireless Infrastructure Synchronization

Use this procedure to start services that support synchronization with a Cisco Wireless LAN Controller in
support of the Location Awareness feature.
Procedure
Step 1 Log in to Cisco Unified Serviceability and choose Tools > Service Activation.
Step 3 Make sure that the following services are checked:
• Cisco CallManager
• Cisco AXL Web Service
• Cisco Wireless Controller Synchronization Service
Step 4 Optional. If you want to use Bulk Administration to import your network infrastructure from a CSV file, make
sure that Bulk Provisioning Service is checked.
Step 5 Click Save.
Configure Wireless Access Point Controller

Use this procedure to synchronize the database with a Cisco wireless access point controller. During the sync,
Unified Communications Manager updates its database with the wireless access point infrastructure that the
controller manages. You can add up to fifty wireless access point controllers.
Procedure
Step 1 From Cisco Unified CM Administration, choose Advanced Features > Device Location Tracking Services >
Wireless Access Point Controllers.
Step 2 Select the controller that you want to configure:
• Click Find and select the controller to edit an existing controller.
• Click Add New to add a new controller.
724
Insert Infrastructure Devices
Step 3 In the Name field, enter the IP address or hostname for the controller.
Step 4 Enter a Description for the controller.
Step 5 Complete the SNMP settings that will be used for SNMP messaging to the controller:
a) From the SNMP Version drop-down list, select the SNMP version protocol that the controller uses.
b) Complete the remaining SNMP authentication fields.For more information on the fields and their
c) Click the Test SNMP Settings to confirm that you entered valid SNMP settings.
Step 6 If you want to configure scheduled syncs to regularly update the database:
a) Check the Enable scheduled synchronization to discover Infrastructure Devices check box.
b) In the Perform a Re-sync Every fields, create the synchronization schedule.
Step 7 Click Save.
Step 8 (Optional) To update the database immediately, click Synchronize.
Optional. If the synchronization pulls in access points that you do not want to track (for example, lab equipment
or access points that are not in use) you can remove the access point from tracking.
Insert Infrastructure Devices

Use this procedure to complete a bulk import of your wireless Access Point infrastructure from a CSV file
into the Unified Communications Manager database. You can use this procedure to import a CSV file that
was exported from Cisco Prime Infrastructure or if you want to import access points from a third-party wireless
Access Point controller.
Before you begin

You must have a data file in comma separated value (CSV) format with the following delineated columns:
• AccessPoint or Switch Name
• IPv4 Address
• IPv6 Address
• BSSID—Required for Wireless Access Protocol (WAP) infrastructure devices
• Description—A location identifier, a combination of switch type and location, or another meaningful
identifier
Note You can define both an IPv4 and IPv6 address, or you can define an IPv4 or an IPv6 address.
Note For the BSSID value, enter the BSSID mask, ending in 0, that uniquely identifies the access point as opposed
to the BSSIDs for the individual channels on the access point.
725
Deactivate Infrastructure Device from Tracking
Procedure
Step 1 Choose Bulk Administration > Infrastructure Device > Insert Infrastructure Device.
The Insert Infrastructure Device Configuration window displays.
Step 2 In the File Name field, choose the CSV data file that you created for this transaction.
Step 3 In the Job Information area, enter the Job description.
The default description is Insert Infrastructure Device.
Step 4 Select when you want to run the job:

• Select the Run Immediately radio button, if you want to run the job immediately.
• Select the Run Later radio button, if you want to schedule the job for later.

If you chose to run the job immediately, the job runs.
Step 6 If you chose to run the job later, schedule when the job runs:
a) Choose Bulk Administration > Job Scheduler.
b) Click Find and select the job that you just created.
c) In the Job Scheduler window, schedule when you want to run the job.
d) Click Save.
At the scheduled time, the job runs.
Deactivate Infrastructure Device from Tracking

If the synchronization includes access points or switches that you do not want to track (for example, if the
sync pulls in lab equipment or access points that are not in use), you can deactivate the access point or switch
from tracking. Unified Communications Manager will not update the status for the access point or switch.
Procedure
Step 1 In Cisco Unified CM Administration, choose Advanced Features > Device Location Tracking Services >
Switches and Access Points.
Step 2 Click Find and select the switch or access point that you want to stop tracking.
Step 3 Click Deactivate Selected.
Related Documentation
After you complete your system configuration, and your system is up and running, you can use tasks in the
following chapter to manage your infrastructure on an ongoing basis:
"Manage Infrastructure", Administration Guide for Cisco Unified Communications Manager and IM and
Presence Service
726
Location Awareness Restrictions

Meraki Access Points The Location Awareness feature does not support Meraki access points.
727
728
CHAPTER 61
Configure Flexible DSCP Marking and Video
Promotion
• Flexible DSCP Marking and Video Promotion Overview, on page 729
• Custom QoS Settings for Users, on page 730
• Traffic Class Label, on page 731
• DSCP Settings Configuration Task Flow, on page 731
• Flexible DSCP Marking and Video Promotion Interactions, on page 735
• Flexible DSCP Marking and Video Promotion Restrictions, on page 735
Flexible DSCP Marking and Video Promotion Overview

Devices and applications use Differentiated Services Code Point (DSCP) markings to indicate the Quality of
Service (QoS) treatment of IP communications. For example, desktop video endpoints may use multimedia
conferencing AF41 marking for video media streams, while high-definition video room systems may use
real-time interactive CS4 marking. When an application sends and receives IP communications to and from
the same type of application, the DSCP markings are symmetric, and the QoS treatments of the IP
communications that each application sends and receives are the same. However, when an application sends
and receives media to and from a different type of application, the DSCP markings may be asymmetric, and
the QoS treatments of the IP communications that each application sends and receives may be inconsistent.
For example, the QoS treatment of the video media stream that a video room system receives from a desktop
video endpoint may be inadequate to support the expected quality of the video room system.
Devices and applications are subjected to Call Admission Control (CAC) to ensure that adequate bandwidth
is available for the duration of established sessions. The bandwidth that is utilized by established sessions is
updated as the sessions begin and end. Attempts to establish new sessions that would exceed the available
bandwidth are blocked. The amount of bandwidth available may be tracked independently for devices and
applications of different types. For example, independent tracking of bandwidth may be available for desktop
video endpoints and high-definition video room systems to send and receive video media streams.
When devices and applications of the same type send and receive communications, the same type of bandwidth
deductions are made in each direction. However, when devices and applications of different types send and
receive communications, different types of bandwidth deductions must be made in each direction. Moreover,
the bandwidth deductions are usually symmetric in amount, by design, to reflect the usual behavior of an IP
network. As a result, when devices and applications of different types send and receive communications, the
total bandwidth deductions may be up to double the amount of network bandwidth that is actually utilized.
729
Custom QoS Settings for Users
This inconsistency in bandwidth accounting may cause attempts to establish new sessions to be blocked
unnecessarily.
The Flexible DSCP Marking and Video Promotion feature allows you to configure a Video Promotion policy
that reconciles the inconsistency in bandwidth accounting in favor of the application that receives more
favorable CAC and QoS treatment. For example, if a session between a desktop video endpoint and a
high-definition video room system is reconciled in favor of the video room system, then the reconciliation is
deemed a promotion for the desktop video endpoint.
When reconciliation is in effect between devices and applications of different types, bandwidth is deducted
only for the type of application that is favored by reconciliation. If sufficient bandwidth is available for a
session of this type to be admitted, the device or application of the type that is not favored by reconciliation
is instructed to change the DSCP markings that it uses to those that are used by the device or application of
the type that is favored by reconciliation. For example, if a desktop video endpoint is promoted in a session
with a high-definition video room system, bandwidth accounting takes place as if the desktop video endpoint
were an application of the same type as the video room system. The desktop video endpoint is instructed to
change its DSCP markings to those that are used by the video room system. The QoS treatment is consistent
in both directions, bandwidth is deducted for a session between devices and applications of the same type as
the video room system, and bandwidth is not deducted for a session between devices and applications of the
same type as the desktop video endpoint.
When you activate the Flexible DSCP Marking and Video Promotion feature, Unified Communications
Manager dynamically signals desktop video devices a Traffic Class Label that is indicative of the DSCP
marking for each negotiated media stream.
Custom QoS Settings for Users

You can customize Quality of Service (QoS) settings within a SIP profile and apply those settings to your
users. The SIP Profile Configuration window has been enhanced with the following types of QoS settings:
• Custom DSCP values for audio and video streams
• Custom UDP port ranges for audio and video streams
Custom DSCP Values for Audio and Video

You can configure DSCP values for audio and video calls within a SIP profile and apply them to the SIP
phones that use that profile. The SIP Profile Configuration window includes custom DSCP settings for the
following types of calls:
• Audio calls
• Video calls
• Audio portion of a video call
• TelePresence calls
• Audio portion of a TelePresence call
If your company has a set of employees, such as a sales force, or a CEO, who require higher QoS priority
settings than the majority of your employees, you can use the SIP profile configurations to configure custom
DSCP values for those users. The settings within the SIP profile override the corresponding clusterwide service
parameter settings.
730
Traffic Class Label
Custom UDP Port Ranges for Audio and Video

You can configure separate UDP port ranges for the audio stream and video stream of a SIP call. Because
video typically requires considerably more bandwidth than audio, creating dedicated port ranges for each
media type simplifies network bandwidth management. It also protects against audio stream degradation by
guaranteeing that the audio stream will have a dedicated channel that is separate from the higher-bandwidth
video stream.
You can apply this configuration by setting the Media Port Ranges field in the SIP profile to Separate Port
Ranges for Audio and Video. You can then apply the configuration to a phone by associating the SIP profile
to a phone.
Traffic Class Label

The Flexible DSCP and Video Promotion feature uses the Traffic Class Label (TCL) to instruct the SIP
endpoint dynamically to mark its DSCP on a per call basis, based on the Video Promotion policy that you
configure. Because TCL is a SIP Session Description Protocol (SDP) attribute that is defined per media line,
the TCL and its associated DSCP markings can be different for the audio media line and the video media line
of a video call. You can choose different DSCP markings for the audio stream and the video stream of the
video call.
DSCP Settings Configuration Task Flow

Perform the following tasks to configure DSCP values and a video promotion policy for your network.
Procedure

Step 1 Configure Flexible DSCP Marking and Video Configure a video promotion policy to handle
Promotion Policy, on page 731 the different types of video.
Step 2 Configure Custom QoS Policy for Users, on If your company has users that require higher
page 733 priority than other users in your company,
configure a SIP Profile that includes custom
DSCP values for audio and video streams. For
example, if your company has a telephone sales
force or CEO whom require higher priority, you
can apply the customized SIP profile to those
users' phones.
Configure Flexible DSCP Marking and Video Promotion Policy

Follow these steps to configure a video promotion policy to handle the different types of video.
Procedure
731
Flexible DSCP Marking and Video Promotion Service Parameters
Step 2 From the Server drop-down list, choose the server where you want to configure the parameters.
Step 3 From the Service drop-down list, choose the Cisco CallManager (Active) service.
If the service does not display as active, ensure that the service is activated in Cisco Unified Serviceability.
Step 4 To configure a Video Promotion policy that promotes desktop video endpoints to immersive video endpoints,
set the Use Video BandwidthPool for Immersive Video Calls parameter to False and set the Video Call
QoS Marking Policy parameter to Promote to Immersive.
Step 5 To configure other parameters, scroll to the appropriate area of the Service Parameter Configuration window
and update the parameter values. See Flexible DSCP Marking and Video Promotion Service Parameters, on
page 732 for information about the service parameters and their configuration options.
Step 6 Click Save.
Flexible DSCP Marking and Video Promotion Service Parameters
Note For more information about the service parameters, click the parameter name or click the question mark (?)
icon that displays in the Service Parameter Configuration window.
Table 74: Flexible DSCP Marking and Video Promotion Service Parameters
Parameter Description
Clusterwide Parameters This section of service parameters includes clusterwide DSCP values for a wide
(System - QoS) range of audio and video call types, including DSCP for audio calls, video calls,
the audio portion of a video call, TelePresence calls, and the audio portion of a
TelePresence call.
It is highly recommended that you keep these parameters set to the default value
unless a Cisco support engineer instructs otherwise.
Clusterwide Parameters
(Call Admission Control)
Video Call QoS Marking This parameter allows you to configure a Promote to Immersive policy that
Policy reconciles bandwidth allocation inconsistencies between a desktop video endpoint
and a Cisco TelePresence immersive video endpoint in favor of the immersive
endpoint. When promotion is performed, the audio and video bandwidth are
reserved from the immersive bandwidth pool allocation. The policy of Promote
to Immersive takes effect only for calls between an immersive video device and
a desktop video device that supports flexible DSCP marking.
Clusterwide Parameters (System - Location and Region)
Default Intraregion Max This parameter specifies the default maximum total bit rate for each immersive
Immersive Video Call Bit video call within a particular region, when the Use System Default option is
Rate (Includes Audio) selected as the Max Immersive Video Call Bit Rate in the Region Configuration
window for the relationship of the region with itself.
732
Configure Custom QoS Policy for Users
Default Interregion Max This parameter specifies the default maximum total bit rate for each immersive
Immersive Video Call Bit video call between a particular region and another region, when the Use System
Rate (Includes Audio) Default option is selected as the Max Immersive Video Call Bit Rate in the
Region Configuration window for the relationship of the region with the other
region.
Use Video This parameter specifies whether Unified Communications Manager reserves
BandwidthPool for bandwidth from the desktop video bandwidth pool for immersive video calls.
Immersive Video Calls
Configure Custom QoS Policy for Users

Perform the following tasks to set up a custom Quality of Service (QoS) policy for users. You may want to
apply a custom policy if a set of users within your company has different QoS requirements from the rest of
the company such as telephone sales force or a CEO.
Procedure

Step 1 Configure Custom QoS Settings in SIP Profile, Configure a SIP Profile with customized DSCP
on page 733 values and a UDP port range for audio and
video streams.
Step 2 Apply Custom QoS Policy to a Phone, on page Apply the SIP Profile to a phone. The DSCP
734 settings in the SIP Profile override the DSCP
clusterwide service parameter settings..
Configure Custom QoS Settings in SIP Profile

Configure custom DSCP values and UDP port ranges for the phones that use this SIP Profile. You can use
these settings to configure a customized QoS policy that you can apply to specific phones and users within
your network. You may want to do this if you want to apply specific QoS settings to specific users within
your enterprise, such as a sales force, or a CEO.
Procedure
Step 2 Perform either of the following steps:
• Click Find and select an existing SIP Profile.
• Click Add New to create a new SIP Profile.
Step 3 From the Media Port Ranges field, select whether you want to assign a single UDP port range that handles
both audio and video media, or separate port ranges for audio and video streams.
• If you want to configure a single port range for audio and video media, enter the range of ports in the
Start Media Port and Stop Media Port fields. The possible port values are between 2048 and 65535.
733
Apply Custom QoS Policy to a Phone
• If you want separate port ranges for audio and video streams, enter the range of audio ports using the
Start Audio Port and Stop Audio Port fields. Enter the range of video ports using the Start Video Port
and Stop Video Port fields. The possible port values for each are between 2048 and 65535.The two port
ranges must not overlap.
Step 4 In the following fields, configure customized DSCP values for audio and video streams.
• DSCP for Audio Calls
• DSCP for Video Calls
• DSCP for Audio Portion of Video Calls
• DSCP for TelePresence Calls
• DSCP for Audio Portion of TelePresence Calls
Note By default, each of the above fields is configured to use the value from a corresponding service
parameter. If you assign new values, the new value overrides the service parameter setting.
Step 5 Complete the remaining fields in the SIP Profile Configuration window. For help with the fields and their
settings, refer to the online help.
Step 6 Click Save.
Apply Custom QoS Policy to a Phone

Use this procedure to apply a SIP Profile that contains customized QoS settings, including DSCP values and
a UDP port range for audio and video media. When you apply this SIP profile to a phone, the phone uses the
custom settings from the SIP Profile.
Procedure

Step 2 Perform any one of the following steps:
• Click Find and select an existing phone.
• Click Add New to create a new phone.
Step 3 From the SIP Profile drop-down list, select the SIP profile that you set up with the custom DSCP values and
UDP port range values.
Step 4 Complete the remaining fields in the Phone Configuration window. For more information on the fields and
their configuration options, see the system Online Help.
Step 5 Click Save.
734
Flexible DSCP Marking and Video Promotion Interactions
Flexible DSCP Marking and Video Promotion Interactions

Table 75: Flexible DSCP Marking and Video Promotion Interactions
Device Interaction
SIP Intercluster Trunks The Flexible DSCP Marking and Video Promotion feature is supported over SIP
intercluster trunks.
Skinny Client Control The Flexible DSCP Marking and Video Promotion feature is supported for SCCP
Protocol (SCCP) Devices devices.
Pass-Through MTPs If pass-through MTPs are inserted in a call, Unified Communications Manager
signals the MTP to mark the packets with the DSCP marking that is expected
from the endpoint device that originally emitted the packet for the video stream.
If the two endpoints on a call use different DSCP markings (for example, a Cisco
TelePresence immersive video endpoint and a desktop video endpoint without
Video Promotion), the MTPs preserve the DSCP marking in each stream direction.
Flexible DSCP Marking and Video Promotion Restrictions

Table 76: Flexible DSCP Marking and Video Promotion Restrictions
Trunks and gateways The Flexible DSCP Marking and Video Promotion feature is not supported over
H.323 trunks and Media Gateway Control Protocol (MGCP) gateways.
Multilevel Precedence and Cisco recommends that you do not use the Flexible DSCP Marking and Video
Preemption Promotion feature with Multilevel Precedence and Preemption (MLPP) service
calls. When you need MLPP service functionality, Cisco recommends that you
set the Video Call QoS Marking Policy and Use Video BandwidthPool for
Immersive Video Calls service parameters to their default values. With default
values for the Video Call QoS Marking Policy and Use Video BandwidthPool
for Immersive Video Calls service parameters, Unified Communications Manager
and endpoints use MLPP DSCP markings for the media packets.
SIP video endpoints The Flexible DSCP Marking and Video Promotion feature is dependent on desktop
SIP video endpoint support. Currently, only Cisco DX650 series SIP phones
provide the required endpoint support.
735
Flexible DSCP Marking and Video Promotion Restrictions
736
CHAPTER 62
Separate Calling Party Number and Billing
Number in SIP
• External Presentation Name and Number Overview, on page 737
• Call Processing, on page 738
• Directory Number Overview, on page 740
• SIP Profile Overview, on page 744
• SIP Trunk Overview, on page 746
• Intercluster SME Call Flows, on page 751
External Presentation Name and Number Overview

Cisco Unified Communications Manager Administration can be configured to contain separate calling party
and presentation number.
In previous releases, Cisco Unified Communications Manager cannot be configured on a per line basis to
have a different number sent to the PSTN in the FROM and PAID header. If a group of users is configured
to present the same Calling Line Identification Number to PSTN users, which is a nongeographic E.164
number and cannot be used for billing. Therefore, the users actual DDI must be sent in a different field than
the presentation number. With this release, Cisco Unified Communications Manager supports External
Presentation Name and Number that is different from existing Identification Name and Number. The configured
Presentation Name and Number are for display purpose on the following devices:
• SIP
• SCCP
• Single Number Reach Destination (SNRD)
• CTIRD
• SparkRD
Configuration Overview
You can configure the external presentation name and number feature on the following pages:
• Directory Number Configuration
737
Call Processing
• SIP Profile Configuration

• Trunk Configuration
Note • When you configure the External Presentation Information on the SIP Profile Configuration page, the
value of External Presentation Number and External Presentation Name on the SIP Profile
Configuration page is used, overriding the settings configured on the Directory Number page.
• When you configure the Presentation Information on the Trunk Configuration page, the value of
Presentation Number and Presentation Name on the Trunk Configuration page is used, overriding the
settings configured on the SIP Profile Configuration and Directory Number Configuration pages.
Call Processing
This section describes the incoming and outgoing call behavior when you configure the external presentation
name and number feature.
Incoming Call Process

Cisco Unified Communications Manager looks for FROM and PAID header information when there is a call
initiated from the PSTN network. The FROM header contains the external presentation name and number (if
configured). However, this is not the real identity of a user, it is used only for display purpose. The PAID
header contains the identity (original DN or DDI) of the user.
If FROM and PAID headers have different numbers and Enable External Presentation Name and Number
option is enabled in the SIP Profile Configuration page and Display External Presentation Name and
Number service parameter value is set to True, then Cisco Unified Communications Manager displays the
FROM header information (configured external presentation name and number) on the called device. Similarly,
if an option is disabled, Cisco Unified Communications Manager displays PAID header information (user's
original DN or DDI) on the called device.
Note • By default, Enable External Presentation Name and Number field is unchecked.
• Default value of a service parameter Display External Presentation Name and Number is False.
Invite Received from the PSTN Network
From: "Customer Care" <sip:[email protected]>;

To: <sip:[email protected]>
P-Asserted-Identity: "Your personal adviser <sip:[email protected]>
Remote-Party-ID: "Your personal adviser <sip:[email protected]>
In the preceding example, FROM header contains a number different from the PAID header. If you
check Enable External Presentation Name and Number check box and set Display External
738
Outgoing Call Process
Presentation Name and Number value to True, Cisco Unified Communications Manager displays
Customer Care / 1800000 on the called device.
If you uncheck the Enable External Presentation Name and Number check box or set the Display
External Presentation Name and Number to False, then Cisco Unified Communications Manager
displays Your personal adviser / [email protected] on the called device.
Outgoing Call Process

Let us assume, a user configured with External Presentation Name and Number initiates a call to a PSTN
network through the SIP trunk with Enable External Presentation Name and Number configured in its
SIP Profile. Then, Cisco Unified Communications Manager sends the configured External Presentation
Information in the FROM header of the outgoing SIP message and displays on the called device.
If Enable External Presentation Name and Number option is disabled or External Presentation Number
and External Presentation Name fields are not configured, the Cisco Unified Communications Manager
sends the directory number information in the FROM and PAID headers and displays on the called device.
External Presentation Number Mask Operation

Cisco Unified Communications Manager allows you to mask the external presentation number, to be displayed
on the called device. You can mask the presentation number on the Directory Number Configuration, SIP
Profile Configuration, and Trunk Configuration pages.
When you enter the digits in the External Presentation Number field with trailing X, the value of X is
replaced with the directory number information starting from right to left.
Mask Operation on Directory Number Configuration

If you mask External Presentation Number as 180011XXXX on Directory Number Configuration page for
a Directory Number 5551234, then Cisco Unified Communications Manager displays the presentation number
as 1800111234 on the called device.
Mask Operation on SIP Profile Configuration

Let us assume External Presentation Number on Directory Number page is 180011XXXX and if you mask
External Presentation Number on SIP Profile Configuration page as 180022XXXX for a Directory Number
5551234, then Cisco Unified Communications Manager displays the presentation number as 1800221234
on the called device.
Mask Operation on Trunk Configuration

Let us assume External Presentation Number on Directory Number page is 180011XXXX and on SIP
Profile Configuration page is 180022XXXX. If you mask Presentation Number on Trunk Configuration
page as 180033XXXX for a Directory Number 5551234, then Cisco Unified Communications Manager
displays the presentation number as 1800331234 on the called device.
739
Directory Number Overview
Directory Number Overview

In Cisco Unified Communications Manager Administration, use the Call Routing > Directory Number menu
path to configure Directory Numbers (DNs). Using Cisco Unified Communications Manager Administration,
you can configure and modify the DNs that are assigned to specific phones.
A new section External Presentation Information is added on the Directory Number Configuration page.
The administrator can now configure the presentation name and number of their choice to display on the
supported devices for external calls. If an administrator does not want to show users' identity, they have a
privilege to display configured External Presentation Number and External Presentation Name as
Anonymous on the called party device.
Directory Number Configuration Tasks

Procedure

Step 1 Add a new end user using one of the following If your system is synchronized with a company
methods: LDAP directory, you can import the new end
user directly from LDAP.
• Import an End User from LDAP, on page
740 Otherwise, you can add and configure the end
• Add an End User Manually, on page 741 user manually.
Step 2 Assign a phone to new or existing end user by You can use the 'Add New Phone' procedure to
performing one of the following tasks: configure a new phone for the end user using
settings from a universal device template.
• Add New Phone for End User , on page
742 You can also use the 'Move' procedure to assign
• Move an Existing Phone to a End User, an existing phone already configured or
on page 743 pre-configured.
Step 3 Configure External Presentation Information To configure the external presentation number
on DN, on page 743 and external presentaion name for DNs that are
assigned to specific phones.
Import an End User from LDAP

Perform the following procedure to manually import a new end user from a company LDAP directory. If your
LDAP synchronization configuration includes a feature group template with a user profile that includes
universal line and device templates and a DN pool, the import process automatically configures the end user
and primary extension.
Note You cannot add new configurations (for example, adding a feature group template) into an LDAP directory
sync after the initial sync has occurred. If you want to edit an existing LDAP sync, you must either use Bulk
Administration, or configure a new LDAP sync.
740
Add an End User Manually
Before you begin

Before you begin this procedure make sure that you have already synchronized Cisco Unified Communications
Manager with a company LDAP directory. The LDAP synchronization must include a feature group template
with universal line and device templates.
Procedure
Step 1 In Cisco Unified CM Administration, choose System > LDAP > LDAP Directory.
Step 2 Click Find and select the LDAP directory to which the user is added.
Step 3 Click Perform Full Sync.
Cisco Unified Communications Manager synchronizes with the external LDAP directory. Any new end users
in the LDAP directory are imported into the Cisco Unified Communications Manager database.
What to do next
If the user is enabled for self-provisioning, the end user can use the Self-Provisioning Interactive Voice
Response (IVR) to provision a new phone. Otherwise, perform one of the following tasks to assign a phone
to the end user:
• Add New Phone for End User , on page 742
• Move an Existing Phone to a End User, on page 743
Add an End User Manually

Perform the following procedure to add new end user and configure them with an access control group and
a primary line extension.
Note Make sure that you have already set up an access control groups that has the role permissions to which you
want to assign your user. For details, see the "Manage User Access" chapter.
Before you begin

Verify that you have a user profile configured that includes a universal line template. If you need to configure
a new extension, Cisco Unified Communications Manager uses the settings from the universal line template
to configure the primary extension.
Procedure
Step 1 In Cisco Unified CM Administration, choose User Management > User/Phone Add > Quick User/Phone
Add.
Step 2 Enter the User ID and Last Name.
Step 3 From the Feature Group Template drop-down list, select a feature group template.
Step 4 Click Save.
741
Add New Phone for End User
Step 5 From the User Profile drop-down list, verify that the selected user profile includes a universal line template.
Step 6 From the Access Control Group Membership section, click the + icon.
Step 7 From the User is a member of drop-down list, select an access control group.
Step 8 Under Primary Extension, click the + icon.
Step 9 From the Extension drop-down list, select a DN that displays as (available).
Step 10 If all line extensions display as (used), perform the following steps:
a) Click the New... button.
The Add New Extension popup displays.
b) In the Directory Number field, enter a new line extension.
c) From the Line Template drop-down list, select a universal line template.
d) Click OK.
Cisco Unified Communications Manager configures the directory number with the settings from the
universal line template.
Step 11 (Optional) Complete any additional fields in the Quick User/Phone Add Configuration window.
Step 12 Click Save.
What to do next
Perform one of the following procedures to assign a phone to this end user:
Add New Phone for End User

Perform the following procedure to add a new phone for a new or existing end user. Make sure that the user
profile for the end user includes a universal device template. Cisco Unified Communications Manager uses
the universal device template settings to configure the phone.
Before you begin

Perform one of the following procedures to add an end user:
• Add an End User Manually, on page 741
• Import an End User from LDAP, on page 740
Procedure
Step 1 In Cisco Unified CM Administration, choose User Management > User/Phone Add > Quick/User Phone
Add.
Step 2 Click Find and select the end user for whom you want to add a new phone.
Step 3 Click the Manage Devices.
The Manage Devices window appears.
Step 4 Click Add New Phone.
742
Move an Existing Phone to a End User
The Add Phone to User popup displays.

Step 5 From the Product Type drop-down list, select the phone model.
Step 6 From the Device Protocol drop-down list select SIP or SCCP as the protocol.
Step 7 In the Device Name text box, enter the device MAC address.
Step 8 From the Universal Device Template drop-down list, select a universal device template.
Step 9 If the phone supports expansion modules, enter the number of expansion modules that you want to deploy.
Step 10 If you want to use Extension Mobility to access the phone, check the In Extension Mobility check box.
Step 11 Click Add Phone.
The Add New Phone popup closes. Cisco Unified Communications Manager adds the phone to the user and
uses the universal device template to configure the phone.
Step 12 If you want to make additional edits to the phone configuration, click the corresponding Pencil icon to open
the phone in the Phone Configuration window.
Move an Existing Phone to a End User

Perform this procedure to move an existing phone to a new or existing end user.
Procedure
Step 1 In Cisco Unified CM Administration, choose User Management > User/Phone Add > Quick/User Phone
Add.
Step 2 Click Find and select the user to whom you want to move an existing phone.
Step 3 Click the Manage Devices button.
Step 4 Click the Find a Phone to Move To This User button.
Step 5 Select the phone that you want to move to this user.
Step 6 Click Move Selected.
Configure External Presentation Information on DN

Perform the following procedure to configure the external presentation information for DNs that are assigned
to specific phones.
Before you begin

• Check the Enable External Presentation Name and Number check box on the SIP Profile Configuration
page.
• Perform one of the following procedures to add an end user:
• Add an End User Manually, on page 741
• Import an End User from LDAP, on page 740
• Assign a phone to a new or existing end user by performing one of the following tasks:
743
SIP Profile Overview
Procedure
Step 2 From the Find and List Directory Numbers page, perform one of the following steps:
• To update a DN, click Find and select the Directory Number for which you want to display unique
identity.
• To create a new Directory Number, click Add New.
Step 3 In the External Presentation Information section, enter the name and number that you want to display on
the called device.
Note • External Presentation Number field accepts up to 32 digits and can contain the following
characters: [0-9, X, *, #, \, +].
• External Presentation Name field accepts up to 50 characters.
Step 4 (Optional), if you want to show configured External Presentation Number and External Presentation
Name as anonymous, check the Anonymous External Presentation check box.
Note • By default, the Anonymous External Presentation field is unchecked.
• If you check the Anonymous External Presentation field:
External Presentation Number and External Presentation Name fields are noneditable.
Also, the entries from these fields disappear.
Step 5 Complete the remaining fields on the Directory Number Configuration page. For help with the fields and
Step 6 Click Save.
SIP Profile Overview

A SIP profile is a template that comprises common SIP settings. You must assign a SIP profile for every SIP
trunk and SIP device in your network. When you configure a SIP profile and then assign that profile to a SIP
trunk, or a SIP device, the system applies the configured SIP settings to that trunk or device.
744
SIP Profile Configuration Tasks
SIP Profile Configuration Tasks

Procedure

Step 1 Configure SIP Profiles, on page 745 Use this procedure to configure a SIP profile.
Step 2 Configure External Presentation Information To configure the external presentation number
on SIP Profile, on page 745 and external presentation name for a SIP profile.
Configure SIP Profiles

Use this procedure to configure a SIP profile with common SIP settings that you can assign to SIP devices
and trunks that use this profile.
Procedure
• Click Find and select the SIP profile to edit an existing profile, .
• Click Add New to create a new profile.
Step 3 If you want your SIP phones and trunks to support IPv4 and IPv6 stacks, check the Enable ANAT check box.
Step 4 If you want to assign an SDP transparency profile to resolve SDP interoperability, from the SDP Transparency
Profile drop-down list.
Step 5 If you want to assign a normalization or transparency script to resolve SIP interoperability issues, from the
Normalization Script drop-down list, select the script.
Step 6 (Optional) Check the Send ILS Learned Destination Route String check box for Global Dial Plan Replication
deployments where you may need to route calls across a Cisco Unified Border Element.
Step 7 Complete the remaining fields in the SIP Profile Configuration window. For more information on the fields
Step 8 Click Save.
Configure External Presentation Information on SIP Profile

Use this procedure to configure the separate external presentation name and number on SIP Profile
Configuration page.
Before you begin

• Check the Enable External Presentation Name and Number check box on the SIP Profile
Configuration page.
• Set Display External Presentation Name and Number parameter value to True under Clusterwide
Parameters (Device-Phone) section on Service Parameter Configuration page.
745
SIP Trunk Overview
Procedure
• To edit an existing profile, click Find and select the SIP profile.
• To create a new profile, click Add New.
Step 3 In the External Presentation Information section, enter the name and number that you want to display on
the called device.
Note • External Presentation Number field accepts up to 32 digits and can contain the following
characters: [0-9, X, *, #, \, +].
• External Presentation Name field accepts a maximum of 50 characters.
Step 4 (Optional), if you want to show configured External Presentation Number and External Presentation
Name as anonymous, check the Anonymous External Presentation check box.
Note • By default, the Anonymous External Presentation field is unchecked.
External Presentation Number and External Presentation Name fields are noneditable.
Also, the entries from these fields disappear.
Step 5 Complete the remaining fields in the SIP Profile Configuration page. For more information on the fields
Step 6 Click Save.
SIP Trunk Overview

If you are deploying SIP for call control signaling, configure SIP trunks that connect Cisco Unified
Communications Manager to external devices such as SIP gateways, SIP Proxy Servers, Unified
Communications applications, remote clusters, or a Session Management Edition.
Within the Cisco Unified CM Administration, the SIP Trunk Configuration window contains the SIP signaling
configurations that Cisco Unified Communications Manager uses to manage SIP calls.
SIP Trunk supports the separate presentation name and number that is different from existing Caller ID DN
and Caller Name. A new check box Anonymous Presentation is provided to show the configured presentation
name and number as Anonymous on the called device.
746
Trunk Configuration Tasks
Trunk Configuration Tasks

Procedure

Step 1 Configure SIP Trunk Security Profile, on page Configure SIP trunk security profiles with any
747 security settings that you want to apply to your
SIP trunks. For example, you can configure
digest authentication, device security mode, and
TLS encryption for SIP signaling.
If you don't configure SIP trunk security
profiles, by default, Cisco Unified
Communications Manager applies a nonsecure
sip trunk security profile.
Step 2 Configure Common Device Configuration, on Set up a Common Device Configuration for the
page 748 trunk. For dual-stack trunks, configure the IP
addressing preference.
Step 3 Configure SIP Trunks, on page 749 Configure the SIP trunks in your network. In
the Trunk Configuration window, configure
the SIP settings for your trunks. Assign a SIP
profile, SIP trunk security profile, and a
Common Device Configuration to your SIP
trunk. In addition, assign any SIP normalization
or transparency scripts that your trunk
connection requires. For example, if your SIP
trunk connects to a Cisco TelePresence VCS,
you must assign the vcs-interop script to the
SIP trunk.
Step 4 Configure Presentation Information on SIP To configure the presentation name and number
Trunks, on page 750 on SIP Trunk page.
Configure SIP Trunk Security Profile

Configure a SIP Trunk Security Profile with security settings such as digest authentication or TLS signaling
encryption. When you assign the profile to a SIP trunk, the trunk takes on the settings of the security profile.
Note If you don't assign a SIP trunk security profile to your SIP trunks, Cisco Unified Communications Manager
assigns a nonsecure profile by default.
Procedure
Step 1 From Cisco Unified CM Administration, choose System > Security > SIP Trunk Security Profile.
747
Configure Common Device Configuration
Step 3 To enable SIP signaling encryption with TLS, perform the following:
a) From the Device Security Mode drop-down list, select Encrypted.
b) From the Incoming Transport Type and Outgoing Transport Type drop-down lists, choose TLS.
c) For device authentication, in the X.509 Subject Name field, enter the subject name of the X.509 certificate.
d) In the Incoming Port field, enter the port on which you want to receive TLS requests. The default for
TLS is 5061.
Step 4 To enable digest authentication, do the following
a) Check the Enable Digest Authentication check box
b) Enter a Nonce Validity Timer value to indicate the number of seconds that must pass before the system
generates a new nonce. The default is 600 (10 minutes).
c) To enable digest authentication for applications, check the Enable Application Level Authorization
check box.
Step 5 Complete the additional fields in the SIP Trunk Security Profile Configuration window.For more information
Step 6 Click Save.
Note You must assign the profile to a trunk in the Trunk Configuration window so that the trunk can
uses the settings.
Configure Common Device Configuration

A common device configuration comprises a set of optional set of user-specific feature attributes. If you are
deploying IPv6, you can use this configuration to assign IPv6 preferences for SIP trunks or SCCP phones.
Procedure
Step 3 For SIP trunks, SIP Phones or SCCP phones, choose a value for the IP Addressing Mode drop-down list:
• IPv4 Only—The device uses only an IPv4 address for media and signaling.
• IPv6 Only—The device uses only an IPv6 address for media and signaling.
• IPv4 and IPv6 (Default)—The device is a dual-stack device and uses whichever IP address type is
available. If both IP address types are configured on the device, for signaling the device uses the IP
Addressing Mode Preference for Signaling setting and for media the device uses the IP Addressing
Mode Preference for Media enterprise parameter setting.
Step 4 If you configure IPv6 in your previous step, then configure an IP addressing preference for the IP Addressing
Mode for Signaling drop-down list:
• IPv4—The dual stack device prefers IPv4 address for signaling.
• IPv6—The dual stack device prefers IPv6 address for signaling.
• Use System Default—The device uses the setting for the IP Addressing Mode Preference for Signaling
enterprise parameter.
Step 5 Configure the remaining fields in the Common Device Configuration window. For more information on the
748
Configure SIP Trunks
Step 6 Click Save.
Configure SIP Trunks

Use this procedure to configure a SIP trunk. You can assign up to 16 destination addresses for a SIP trunk.
Procedure

Step 4 From the Protocol Type drop-down list, choose the type of SIP trunk that matches your deployment and click
Next:
• None (Default)
• Call Control Discovery
• Extension Mobility Cross Cluster
• Cisco Intercompany Media Engine
• IP Multimedia System Service Control
Step 5 (Optional) If you want to apply a Common Device Configuration to this trunk, select the configuration from
the drop-down list.
Step 6 Check the SRTP Allowed check box if you want to allow encypted media over the trunk.
Step 7 Check the Run on All Active Unified CM Nodes check box if you want to enable the trunk for all cluster
nodes.
Step 8 Configure the destination address for the SIP trunk:
a) In the Destination Address text box, enter an IPv4 address, fully qualified domain name, or DNS SRV
record for the server or endpoint that you want to connect to the trunk.
b) If the trunk is a dual stack trunk, in the Destination Address IPv6 text box, enter an IPv6 address, fully
qualified domain name, or DNS SRV record for the server or endpoint that you want to connect to the
trunk.
c) If the destination is a DNS SRV record, check the Destination Address is an SRV check box.
d) To add additional destinations, click the (+).
Step 9 From the SIP Trunk Security Profile drop-down, assign a security profile. If you don't select this option, a
nonsecure profile will be assigned.
Step 10 From the SIP Profile drop-down list, assign a SIP profile.
Step 11 (Optional) If you want to assign a normalization script to this SIP trunk, from the Normalization Script
drop-down list, select the script that you want to assign.
Step 12 Configure any additional fields in the Trunk Configuration window. For more information on the fields and
Step 13 Click Save.
749
Configure Presentation Information on SIP Trunks
Configure Presentation Information on SIP Trunks

Use this procedure to configure the presentation name and number on SIP Trunk page.
Before you begin

• Check the Enable External Presentation Name and Number check box on the SIP Profile Configuration
page.
• Configure SIP Trunks, on page 749
Procedure

Step 4 From the Trunk Service Type drop-down list, choose the type of SIP trunk that you want to configure:
• None (Default): The trunk will not be used for Call Control Discovery, Extension Mobility Cross-Cluster,
Intercompany Media Engine, or IP Multimedia System Service Control.
• Call Control Discovery: The trunk supports the Call Control Discovery feature.
• Extension Mobility Cross Cluster: The trunk supports Extension Mobility Cross Cluster.
• Cisco Intercompany Media Engine: The trunk supports the Intercompany Media Engine (IME). Make
sure that the IME server is installed before you configure this type of trunk.
• IP Multimedia System Service Control: Choose this option to enable the trunk with support for IP
Multimedia System Service Control.
Step 5 Click Next.

Step 6 In the Presentation Information section, enter the name and number that you want to display on the called
device.
Note • Presentation Number field accepts up to 32 digits and can contain the following characters:
[0-9, X, *, #, \, +].
• Presentation Name field accepts a maximum of 50 characters.
Step 7 (Optional) If you want to show the presentation name and number as anonymous, check the Anonymous
Presentation check box.
Note • By default, the Anonymous Presentation field is unchecked.
Presentation Number and Presentation Name fields are noneditable. Also, the entries
from these fields disappear.
Step 8 (Optional) Check the Send Presentation Name and Number only in the FROM header and not in the
other identity headers check box, if you want to send presentation information that is configured on the SIP
Trunk only to FROM header.
750
Intercluster SME Call Flows
Step 9 Configure any additional fields in the Trunk Configuration window.For more information on the fields and
Step 10 Click Save.

The Cisco Unified Communications Manager Session Management Edition software is same as the Cisco
Unified Communications Manager used mainly for a call routing between clusters or various devices. With
this release, Cisco Unified Communications Manager supports the intercluster SME calls.
Incoming Calls
Let us assume a user from a PSTN network initiates a call with Enable External Presentation Name
and Number enabled in its SIP profile. If Display External Presentation Name and Number service
parameter is set to True, then Cisco Unified Communications Manager sends the presentation number
information to the X-Cisco-Presentation header and displays on the called device. The FROM and PAID
headers contains the identity of the user that is the user's DN or DDI.
If Display External Presentation Name and Number service parameter is set to False, then Cisco
Unified Communications Manager sends the presentation number information to the X-Cisco-Presentation
header. The FROM and PAID headers contains the user’s DN or DDI and displays on the called device.
Outgoing Calls
A user who is configured with External Presentation Name and External Presentation Number
initiates a call to a PSTN network through intercluster SIP trunks. If Enable External Presentation
Name and Number check box is disabled in its SIP profile, then, Cisco Unified Communications
Manager sends the original directory number information in the FROM and PAID headers and displays
on the called device and configured External Presentation Information in the X-Cisco-Presentation header.
Similarly, if Enable External Presentation Name and Number check box is enabled in its SIP profile,
Cisco Unified Communications Manager sends the configured External Presentation Information in the
FROM header and displays on the called device and original Directory Number in the PAID header.
751
752
CHAPTER 63
SIP OAuth Mode
• SIP OAuth Mode Overview, on page 753
• SIP OAuth Mode Prerequisites, on page 754
• SIP OAuth Mode Configuration Task Flow, on page 754
SIP OAuth Mode Overview

Secure registrations to Unified Communications Manager involves a process of updating CTL files, setting
up a mutual certificate trust store and so on. If devices are switching between on-premises and off-premises,
it is difficult to update LSCs and renew Certificate Authority Proxy Function (CAPF) enrolment each time
when a secure registration is completed.
SIP OAuth mode allows you to use OAuth refresh tokens for all devices authentication in secure environments.
This feature enhances the security ofUnified Communications Manager.
Unified Communications Manager verifies the token presented by the endpoints and serves the configuration
files only to authorized ones. OAuth token validation during SIP registration is completed when OAuth based
authorization is enabled on Unified Communications Manager cluster and other Cisco devices.
OAuth support for SIP registrations is extended for
• Cisco Jabber devices from Cisco Unified Communications Manager 12.5 release onwards
• SIP Phones from Cisco Unified Communications ManagerRelease 14 onwards
Note By default, TFTP is secure for SIP phones when SIP OAUth is enabled. TFTP file download happens through
secured channel, and only for authenticated phones. SIP OAuth provides end to end secure signaling and
media encryption without CAPF on-premises as well as over MRA.
The following are the Phone Security Profile Types that can be configured for OAuth.
• Cisco Dual Mode For iPhone (TCT device)
• Cisco Dual Mode For Android (BOT device)
• Cisco Unified Client Service Framework (CSF device)
• Cisco Jabber for Tablet (TAB device)
753
SIP OAuth Mode Prerequisites
• Universal Device Template

• Cisco 8811
• Cisco 8841
• Cisco 8851
• Cisco 8851NR
• Cisco 8861
• Cisco 7811
• Cisco 7821
• Cisco 7841
• Cisco 7861
• Cisco 8845
• Cisco 8865
• Cisco 8865NR
• Cisco 7832
• Cisco 8832
• Cisco 8832NR
SIP OAuth Mode Prerequisites

This feature assumes that you have already completed the following:
• Ensure Mobile and Remote Access is configured and the connection is established between Unified
Communication Manager and Expressway.
• Ensure Unified Communications Manager is registered to a Smart or Virtual account with allow
export-controlled functionality.
• Ensure client firmware supports SIP OAuth.
SIP OAuth Mode Configuration Task Flow

Complete the following tasks to configure SIP OAuth for your system.
Procedure

Step 1 Upload CA Certificate to the Phone Edge Trust Upload CA Certificate to the phone edge trust
to get the tokens. This step is not applicable for
Cisco Jabber device.
754
Upload CA Certificate to the Phone Edge Trust

Step 2 Configure Refresh Logins, on page 755 Enable oauth with refresh login flow on Unified
Communications Manager to register the device
via SIP OAuth.
Step 3 Configure OAuth Ports, on page 756 Assign the ports for OAuth for each node that
has OAuth registration.
Step 4 Configure OAuth Connection to Expressway-C, Configure a mutually authenticated TLS

on page 757 connection to Expressway-C.
Step 5 Enable SIP OAuth Mode, on page 757 Enable OAuth services using a CLI command
on the publisher node.
Step 6 Restart Cisco CallManager Service, on page 757 Restart this service on all nodes that have
OAuth registrations.
Step 7 Configure Device Security Mode in Phone Configure OAuth support within a Phone
Security Profile Security Profile if you are deploying encryption
for the endpoints.
Upload CA Certificate to the Phone Edge Trust

Use this procedure to upload the root certificate of Tomcat signed certificate to the Phone Edge Trust.
Note This procedure is performed only for Cisco Phones and not applicable for Cisco Jabber.
Procedure
Step 2 Click Upload Certificate/Certificate chain.
Step 3 In the Upload Certificate/Certificate chain window, from the Certificate Purpose drop-down list choose
Phone-Edge-Trust.
Step 4 In the Upload File field, click Browse and upload the certificate.
Configure Refresh Logins

Use this procedure to configure Refresh Logins with OAuth access tokens and refresh tokens for Cisco Jabber
clients.
755
Configure OAuth Ports
Procedure
Step 2 Under SSO and OAuth Configuration, set the OAuth with Refresh Login Flow parameter to Enabled.
Step 3 (Optional) Set any other parameters in the SSO and OAuth Configuration section. For parameter descriptions,
click on the parameter name.
Step 4 Click Save.
Configure OAuth Ports

Use this procedure to assign the ports that are used for SIP OAuth.
Procedure
Step 1 From Cisco Unified CM Administration, choose, System > Cisco Unified CM.
Step 2 Do the following for each server that uses SIP OAuth.
Step 3 Select the server.
Step 4 Under Cisco Unified Communications Manager TCP Port Settings, set the port values for the following
fields:
• SIP Phone OAuth Port
Default value is 5090. Acceptable configurable range is 1024–49151.
• SIP Mobile and Remote Access Port

Default value is 5091. Acceptable configurable range is 1024–49151.
Note Cisco Unified Communications Manager uses SIP Phone OAuth Port (5090) to listen for SIP
line registration from Jabber on-premises devices over TLS. However, Unified CM uses SIP
Mobile Remote Access Port (default 5091) to listen for SIP line registrations from Jabber over
Expressway through mTLS.
Both ports use the Cisco Tomcat certificate and Tomcat-trust for incoming TLS/mTLS connections.
Make sure that your Tomcat-trust store is able to verify the Expressway-C certificate for SIP
OAuth mode for Mobile and Remote Access to function accurately.
You must perform extra steps to upload the Expressway-C certificate into the Tomcat-Trust
certificate store of the Cisco Unified Communications Manager, when:
• Expressway-C certificate and Cisco Tomcat certificate is not signed by the same CA
certificate.
• Unified CM Cisco Tomcat certificate is not CA signed.
Step 5 Click Save.
756
Configure OAuth Connection to Expressway-C
Step 6 Repeat this procedure for each server that uses SIP OAuth.
Configure OAuth Connection to Expressway-C

Use this procedure to add the Expressway-C connection to Cisco Unified Communications Manager
Administration. You need this configuration for devices in Mobile and Remote Access mode with SIP OAuth.
Procedure
Step 1 From Cisco Unified CM Administration, choose Device > Expressway-C.

Step 2 (Optional) In the Find and List Expressway-C window, click Find to verify X.509 Subject Name/Subject
Alternate Name that is pushed from the Expressway-C to Unified Communications Manager.
Note If required, you can modify the values. Alternatively, if the entries are missing, add Expressway-C
information.
If the Expressway-C has a different domain than the Unified Communications Manager, then the
administrator needs to access the Cisco Unified CM Administration User Interface and add the
domain to the Expressway C in the Unified CM configuration.

Step 4 Enter an IP Address, Hostname or fully qualified domain name for the Expressway-C.
Step 5 Enter a Description.
Step 6 Enter the X.509 Subject Name/Subject Alternate Name of the Expressway-C from the Expressway-C certificate.
Step 7 Click Save.
Enable SIP OAuth Mode

Use the Command Line Interface to enable SIP OAuth mode. Enabling this feature on the publisher node also
enables the feature on all cluster nodes.
Procedure
Step 1 On the Unified Communications Manager publisher node, log in to the Command Line Interface.
Step 2 Run the utils sipOAuth-mode enable CLI command.
Restart Cisco CallManager Service

After enabling SIP OAuth through CLI, restart the Cisco CallManager service on all nodes where endpoints
register through SIP OAuth.
757
Configure Device Security Mode in Phone Security Profile
Procedure
Step 1 From Cisco Unified Serviceability, choose Tools > Control Center > Feature Services.
Step 2 From the Server drop-down list, select the server.
Step 3 Check the Cisco CallManager service and click Restart.
Configure Device Security Mode in Phone Security Profile

Use this procedure to configure the device security mode in the phone security profile and is required only if
you have set the Device Security Mode within that phone’s Phone Security Profile to Encrypted.
Procedure
Step 2 Perform either of the following:
• Search for an existing phone security profile
• Click Add New
Step 3 In the Phone Security Profile Information section, from the Device Security Mode drop-down list, choose
Encrypted.
Step 4 From the Transport Type drop-down list, choose TLS.
Step 5 Check the Enable OAuth Authentication check box.
Step 6 Click Save.
Step 7 Associate the Phone Security Profile to the phone. For more information on how to apply the phone security
phones, see "Apply Security Profiles to Phone" section in Security Guide for Cisco Unified Communications
Manager.
Note Reset your phone for the changes to take effect.
Note When SIP OAuth Mode is enabled, Enable Digest Authentication and TFTP Encrypted Config
options are not supported. Phones will download the TFTP config file securely over https(6971)
and use the token for authentication.
Configure SIP Oauth Registered Phones for MRA Mode

Use this procedure to configure SIP OAuth registered phones to MRA mode.
758
Before you begin
Important This section is applicable from Release 14 onwards.
Make sure your phones are configured to use Activation Codes. For more information see Set Registration
Method to use Activation Codes section in System Configuration Guide for Cisco Unified Communications
Manager.
Note When using SIP OAuth over MRA , user cannot use username / password for login but have to use activation
code based onboarding
Procedure

Step 2 Click Find and select the device which you want to configure for off-premises mode.
Step 3 In the Device Information section, do the following:
• Check Allow Activation Code via MRA check box.
• From the Activation Code MRA Service Domain drop-down list, choose the required MRA service
domain. For more information on how to configure the MRA service domain see, the MRA Service
Domain Configuration section in System Configuration Guide for Cisco Unified Communications
Manager.
Note For SIP OAuth over MRA mode, use only activation code and do not use username/password
based login.
Step 4 In the Protocol Specific Information section, choose the OAuth enabled SIP profile from the Device Security
Profile drop-down list. Make sure that the phone supports OAuth firmware. For more information, on how
to create a security profile, see Configure Phone Security Profile section in System Configuration Guide for
Step 5 Click Save and Apply Configuration.
Note The phone switches to MRA mode and initiates communication with the Expressway. If your
internal network does not allow communication with Expressway from on-premises, the phone
doesn't register but is ready to contact Expressway when it's powered up off-premises.
759
760
PA R T XV
QoS Management
• Configure QoS with APIC-EM Controller, on page 763
• Configure AS-SIP Endpoints, on page 769
• Configure Multilevel Precedence and Preemption, on page 781
CHAPTER 64
Configure QoS with APIC-EM Controller
• APIC-EM Controller Overview, on page 763
• APIC-EM Controller Prerequisites, on page 764
• APIC-EM Controller Configuration Task Flow, on page 764
APIC-EM Controller Overview

The APIC-EM Controller provides a centralized system for managing network traffic so that you always have
the bandwidth to maintain communications, even in congested networks. You can configure Cisco Unified
Communications Manager to use the APIC-EM Controller to manage SIP media flows thereby providing the
following benefits:
• Centralizes QoS management, thereby eliminating the need for endpoints to assign DSCP values.
• Applies differential QoS treatment for different media flows. For example, you can prioritize audio over
video to ensure that basic audio communication is always maintained, even when network bandwidth is
low.
• External QoS setting in the SIP Profile allows you to target which users will use the APIC-EM. For
example, you may have Cisco Jabber users use the APIC-EM to manage media flows, while Cisco Unified
IP Phone users use the DSCP settings in Cisco Unified Communications Manager
SIP Media Flow Management

For SIP calls that use APIC-EM, Cisco Unified Communications Manager sends the policy request to the
APIC-EM Controller at the call outset notifying the APIC-EM of the media flow that is being set up. The
policy request contains information about the call, including the IP address and ports for source and destination
devices, the media type for the flow and the protocol.
The APIC-EM notifies the switch at the beginning of the call flow of the DSCP values for the associated
media flows. The switch inserts those DSCP values into individual media packets, overwriting any values
that the endpoint inserts. If a gateway in the call flow experiences congestion, that gateway sends through the
packets with the higher DSCP values first. This ensures that high priority audio and video streams are not
blocked by lower-priority network traffic such as email, print jobs, or software downloads. When the call
ends, Cisco Unified Communications Manager notifies the APIC-EM and the APIC-EM notifies the switch
to delete the flow.
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QoS Management
APIC-EM Controller Prerequisites
External QoS Support

In order for Cisco Unified Communications Manager to use the APIC-EM to manage media flows, the External
QoS parameter must be enabled at both the system level, via a clusterwide service parameter, and at the device
level, via the SIP Profile.
APIC-EM Controller Prerequisites

Before using APIC-EM, you must do the following:
• Configure DSCP priority for different SIP media flows in Cisco Unified Communications Manager. For
details, see DSCP Settings Configuration Task Flow, on page 731.
• Configure the APIC-EM controller hardware within your network. For details, see the hardware
documentation that comes with the APIC-EM controller.
APIC-EM Controller Configuration Task Flow

Complete these tasks on Cisco Unified Communications Manager to enable APIC-EM Controller to manage
SIP media flows.
Procedure

Step 1 Configure the APIC-EM Controller, on page Configure Unified CM on the APIC-EM
765 Controller.
Step 2 Upload APIC-EM Controller Certificate, on Upload the APIC-EM certificate into Cisco
page 765 Unified OS Administration.
Step 3 Configure HTTPS Connection to APIC-EM Configure an HTTP Profile that points to the
Controller, on page 766 APIC-EM service.
Step 4 Enable External QoS Service for System, on Enable the External QoS Enable service
page 766 parameter to configure the system to use the
APIC-EM to manage media flows. The service
parameter must be enabled for devices to use
the APIC-EM for SIP media flow management.
Note You must also enable external
QoS within the SIP Profile for
devices that will use the
APIC-EM for SIP media flow
management.
Step 5 Configure External QoS Service at SIP Profile Enable external QoS within a SIP Profile. All
Level, on page 766 devices that use this SIP Profile will be able to
use the APIC-EM to manage SIP media flows
764
QoS Management
Configure the APIC-EM Controller

You can use the SIP Profile setting to configure
which devices and device types you want the
APIC-EM to manage media flows.
Step 6 Assign SIP Profile to Phones, on page 767 Associate the external QoS-enabled SIP Profile
to a phone.
Configure the APIC-EM Controller

Use this procedure on the APIC-EM Controller to add Cisco Unified Communications Manager as a user.
APIC-EM's role-based access control feature provides Cisco Unified Communications Manager with access
to APIC-EM resources.
Procedure
Step 1 On the APIC-EM Controller, choose Settings > Internal Users.

Step 2 Create a new user with the following role: ROLE_POLICY_ADMIN. Keep track of the username and
password that you enter because you must enter identical credentials in Cisco Unified Communications
Manager's HTTP Profile window.
Step 3 Go to the Discovery tab and add a discovery with CDP or the IP address range of the available devices.
Step 4 Select the Device Inventory tab and select the reachable devices.
Step 5 Click on Set Policy Tag.
Step 6 Create a policy tag and set it for the devices.
Step 7 On the EasyQoS tab, select the policy that you created and enable DynamicQoS.
Upload APIC-EM Controller Certificate

Use this procedure to upload the APIC-EM controller certificate into Cisco Unified Communications Manager.
Procedure
Step 2 Click Upload Certificate/Certificate Chain.
The Upload Certificate/Certificate Chain popup window appears.
Step 3 From the Certificate Purpose drop-down list, choose CallManager-trust.
Step 4 Enter a Description for the certificate.
Step 5 Click Browse to search for, and select, the certificate.
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QoS Management
Configure HTTPS Connection to APIC-EM Controller
Configure HTTPS Connection to APIC-EM Controller

Use this procedure to set up an HTTP Profile to connect Cisco Unified Communications Manager to the
APIC-EM Controller. In this connection, Cisco Unified Communications Manager acts as an HTTP user and
the APIC-EM acts as the HTTP server.
Procedure
Step 1 From Cisco Unified CM Administration, choose Call Routing > HTTP Profile.
Step 2 Enter a Name for the service.
Step 3 Enter the User Name and Password for this HTTP connection. The user name does not have to be a configured
end user in Cisco Unified Communications Manager, but the user name and password must match the values
that are configured in the APIC-EM Controller.
Step 4 In the Web Service Root URI text box, enter the IP address or fully qualified domain name of the APIC-EM
service.
Step 5 Configure any remaining fields in the HTTP Profile window. For help with the fields and their options, refer
to the online help.
Step 6 Click Save.
Enable External QoS Service for System

Enable External QoS Service for System
Use this procedure to configure Cisco Unified Communications Manager to use an external service for QoS
management. You must enable this service parameter in order to use an APIC-EM controller for QoS.
Procedure
Step 4 Set the value of the External QoS Enabled service parameter to True.
Step 5 Click Save.
Note To use the APIC-EM to manage call flows for devices, you must also enable external QoS within
the SIP Profile for the device.
Configure External QoS Service at SIP Profile Level

If you have enabled the External QoS Enabled clusterwide service parameter, use this procedure to enable
external QoS for SIP devices that use this SIP Profile.
766
QoS Management
Assign SIP Profile to Phones
Note External QoS must be enabled at both the system level and in the SIP Profile to use the APIC-EM to manage
QoS.
Procedure
Step 1 In Cisco Unified CM Administration, choose Device > Device Settings > SIP Profile.
Step 3 Check the Enable External QoS check box. This check box must be checked for phones that use this SIP
Profile to use the APIC-EM Controller to manage QoS.
Step 4 Complete the remaining fields in the SIP Profile Configuration window. For help with the fields and their
settings, see the online help.
Step 5 Click Save.

Use this procedure if you want to assign the external QoS-enabled SIP Profile that you created to a phone.
Tip Use the Bulk Administration Tool to update the SIP Profile for a large selection of phones in a single operation.
For details, see the Bulk Administration Guide for Cisco Unified Communications Manager.
Procedure

Step 2 Click Find and select an existing phone.
Step 3 From the SIP Profile drop-down list, select the SIP Profile that you updated for phones that will use the
APIC-EM Controller to manage traffic.
Step 4 Complete any remaining fields in the Phone Configuration window. For more information on the fields and
Step 5 Click Save.
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QoS Management
768
CHAPTER 65
Configure AS-SIP Endpoints
• AS-SIP Overview, on page 769
• AS-SIP Prerequisites, on page 771
• AS-SIP Enpdoint Configuration Task Flow, on page 772
AS-SIP Overview
Assured Services SIP (AS-SIP) endpoints are compliant with MLPP, DSCP, TLS/SRTP, and IPv6 requirements.
AS-SIP provides for multiple endpoint interfaces on the Unified Communications Manager.
Many Cisco IP phones support AS-SIP. In addition, the Third-Party AS-SIP Endpoint device type allows a
third-party AS-SIP compliant endpoint to be configured and used with Cisco Unified Communications
Manager. In addition, the Third-Party AS-SIP Endpoint device type allows a third-party AS-SIP-compliant
generic endpoint to be configured and used with Cisco Unified Communications Manager
AS-SIP Capabilities
The following capabilities are implemented or made available for AS-SIP endpoints:
• MLPP
• TLS
• SRTP
• DSCP for precedence levels
• Error responses
• V.150.1 MER
• Conference Factory flow support
• AS-SIP Line Early Offer
Third-Party AS-SIP Phones

Third-party phones can be provisioned in Cisoc Unified Communications Manager using the Third-Party
AS-SIP Endpoint device type.
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QoS Management
Third-Party AS-SIP Phones
Third-party phones that are running AS-SIP do not get configured through the Cisco Unified Communications
Manager TFTP server. The customer must configure them by using the native phone configuration mechanism
(usually a web page or TFTP file). The customer must keep the device and line configuration in the Cisco
Unified Communications Manager database synchronized with the native phone configuration (for example,
extension 1002 on the phone and 1002 in Cisco Unified Communications Manager). Also, if the directory
number of a line is changed, the customer must ensure that it gets changed in both Cisco Unified CM
Administration and in the native phone configuration mechanism.
Identification of Third-Party Phones

The third-party phones that are running SIP do not send a MAC address, they must identify themselves by
using username. The REGISTER message includes the following header:
Authorization: Digest
username=”swhite”,realm=”ccmsipline”,nonce=”GBauADss2qoWr6k9y3hGGVDAqnLfoLk5”,uri
=”sip:172.18.197.224”,
algorithm=MD5,response=”126c0643a4923359ab59d4f53494552e”
The username, swhite, must match a user that is configured in the End User Configuration window of
Cisco Unified CM Administration. The administrator configures the SIP third-party phone with the user; for
example, swhite, in the Digest User field of Phone Configuration window.
Note You can assign each user ID to only one third-party phone. If the same user ID is assigned as the Digest User
for multiple phones, the third-party phones to which they are assigned will not successfully register.
Configuration of Third Party AS-SIP Phones and Cisco IP Phones

The following table provides a comparison overview of the configuration differences between Cisco Unified
IP Phones and third-party phones that are running AS-SIP.
Table 77: Comparison of the Configuration Differences Between Cisco IP Phones and Third-Party Phones
Phone Integrated Sends MAC Downloads Downloads Supports Supports

Running with Address Softkey File Dial Plan File Unified Reset and
AS-SIP Centralized Communications Restart
TFTP Manager
Failover and
Fallback
Cisco IP Yes Yes Yes Yes Yes Yes
Phone
Third-party No No No No No No
AS-SIP
device
Note Not all Cisco IP Phones support AS-SIP. See the phone administration guide for your phone model for support
information
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QoS Management
AS-SIP Conferencing
Use Cisco Unified CM Administration to configure third-party phones that are running SIP (For more
information, see "Configure SIP Profile" topic in System Configuration Guide for Cisco Unified
the ). The administrator must perform configuration steps on the third-party phone that is running SIP; see
the following examples:
• Ensure that proxy address in the phone is the IP or Fully Qualified Domain Name (FQDN) of Cisco
• Ensure directory numbers in the phone match the directory numbers that are configured for the device
in Cisco Unified CM Administration.
• Ensure digest user ID (sometimes referred to as Authorization ID) in the phone matches the Digest User
ID in the Cisco Unified CM Administration.
For more information, refer to the documentation that came with the third-party phone.
AS-SIP Conferencing
MOH is applied to its target (a held party, transferee just before transfer, or conferee just before joining the
conference), if the feature invoker (holder, transferor, or conference initiator) supports Cisco-proprietary
feature signaling. If the feature invoker does not support Cisco-proprietary feature signaling, then MOH is
not applied to its target. Also, if an endpoint explicitly signals that it is a conference mixer, then MOH will
not be played to the target. There are two forms of AS-SIP Conferencing:
• Local mixing
• Conference Factory
Local mixing
To the Unified CM, the conference initiator simply appears to have established simultaneously active calls,
one to each of the other conference attendees. The initiator host the conference locally and the voices are
mixed there. The calls from the conference initiator have special signaling that prevent it from being connected
to an MOH source.
Conference Factory
The conference initiator calls a Conference Factory Server located off a SIP trunk. Through IVR signaling,
the conference initiator instructs the Conference Factory to reserve a conference bridge. The Conference
Factory gives the numeric address (a routable DN) to the conference initiator, who then establishes a
subscription with the bridge to receive conference list information to track the participants. The Conference
Factory sends special signaling that prevent it from being connected to an MOH Source.
AS-SIP Prerequisites
Determine whether sufficient Device License Units are available. For more information, see"Smart Software
Licensing" chapter from System Configuration Guide for Cisco Unified Communications Manager
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QoS Management
AS-SIP Enpdoint Configuration Task Flow
AS-SIP Enpdoint Configuration Task Flow

Complete the following tasks to configure an AS-SIP endpoint.
Procedure

Step 1 Configure a Digest User, on page 773 Configure the end user to use digest
authentication for SIP requests.
Step 2 Configure SIP Phone Secure Port, on page 773 Cisco Unified Communications Manager uses
this port to listen to SIP phones for SIP line
registrations over TLS.
Step 3 Restart Services, on page 773 After configuring the secure port, restart the
Cisco CallManager and Cisco CTL Provider
services.
Step 4 Configure SIP Profile for AS-SIP, on page 774 Configure a SIP profile with SIP settings for
your AS-SIP endpoints and for your SIP
trunks.
Note The phone-specific parameters
are not downloaded to a
third-party AS-SIP phone. They
are used only by Cisco Unified
Third-party phones must locally
configure the same settings.
Step 5 Configure Phone Security Profile for AS-SIP, You can use the phone security profile to
on page 775 assign security settings such as TLS, SRTP,
and digest authentication
Step 6 Configure AS-SIP Endpoint, on page 775 Configure a Cisco IP Phone or a third-party
endpoint with AS-SIP support.
Step 7 Associate Device with End User, on page 776 Associate the endpoint with a user.
Step 8 Configure SIP Trunk Security Profile for You can use the sip trunk security profile to
AS-SIP, on page 777 assign security features such as TLS or digest
authentication to a SIP trunk.
Step 9 Configure SIP Trunk for AS-SIP, on page 777 Configure a SIP trunk with AS-SIP support.
Step 10 Configure AS-SIP Features, on page 778 Configure additional AS-SIP features such as
MLPP, TLS, V.150 and IPv6.
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QoS Management
Configure a Digest User
Configure a Digest User

Use this procedure to configure an end user as a digest user whom uses digest authentication. Devices that
are associated to the user will be authenticated via the user's digest credentials.
Procedure
• Click Add New to create a new user.
• Click Find and select an existing user.
Step 3 Make sure the following mandatory fields are completed:

• User ID
• Last Name
Step 4 In the Digest Credentials field, enter a password. End users must authenticate themselves via this password
when using the endpoint.
Step 5 Complete any remaining fields. For help with the fields and their settings, see the online help.
Step 6 Click Save.
Configure SIP Phone Secure Port

Follow these steps to configure the SIP Phone Secure Port. Cisco Unified Communications Manager uses this
port to listen to SIP phones for SIP line registrations over TLS.
Procedure
Step 1 From Cisco Unified CM Administration, choose System > Cisco Unified CM.
Step 2 In the Cisco Unified Communications Manager TCP Port Settings for this Server section, specify a port
number in the SIP Phone Secure Port field, or leave the field set to default. The default value is 5061.
Step 3 Click Save.
Step 5 Click Ok.
Restart Services
Follow these steps to restart Cisco CallManager and Cisco CTL Provider services.
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QoS Management
Configure SIP Profile for AS-SIP
Procedure
Step 1 From the Cisco Unified Serviceability interface, choose Tools > Control Center - Feature Services.
Step 2 Choose the Cisco Unified Communications Manager server from the Servers drop-down list.
In the CM Services area, Cisco CallManager displays in the Service Name column.
Step 3 Click the radio button that corresponds to the Cisco CallManager service.
The service restarts and displays the message, Service Successfully Restarted.
Step 5 Repeat step 3 and step 4 to restart Cisco CTL Provider service.
Configure SIP Profile for AS-SIP

Use this procedure to configure SIP profile with SIP settings for your AS-SIP endpoints and for your SIP
trunks.
Procedure
Step 1 In Cisco Unified CM Administration, choose Device > Device Settings > SIP Profile.
Step 3 Enter a Name and Description for the SIP Profile.

Step 4 Check the Assured Services SIP conformance check box.
Note This checkbox must be checked for SIP trunks and for third-party AS-SIP phones. It's not
mandatory for Cisco IP Phones that support AS-SIP.
Step 5 In the Parameters used in Phone section, configure DSCP precedence values for the types of calls that you
expect to make.
Note You can also configure DSCP values via clusterwide service parameters. However, the DSCP
values within a SIP Profile override the clusterwide settings for all devices that use the SIP Profile.
Step 6 From the Early Offer support for voice and video calls drop-down list, select one of the following options
to configure Early Offer support for SIP trunks that use this profile:
• Disabled
• Best Effort (no MTP Inserted)
• Mandatory (insert MTP if needed)
Step 7 Complete the remaining fields in the SIP Profile Configuration window. For more information on the fields
Step 8 Click Save.
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QoS Management
Configure Phone Security Profile for AS-SIP
Configure Phone Security Profile for AS-SIP

Use this procedure to configure a phone security profile for AS-SIP endpoints. You can use the security profile
to assign security settings such as TLS and SRTP.
Procedure
• Click Add New to create a new phone security profile.
• Click Find to edit an existing profile.
Step 3 For new profiles, select an option from the Phone Security Profile drop-down, choose the phon
emodelThird-party AS-SIP Endpoint and click Next.
• For Cisco IP phones, select the phone model and click Next.
• For third-party AS-SIP endpoints, select Third-party AS-SIP Endpoint and click Next.
Step 4 For the protocol, select SIP and click Next.

Step 5 Enter a Name and Description for the protocol.
Step 6 Assign the Device Security Mode, to one of the following settings:
• Authenticated—Cisco Unified Communications Manager uses TLS signaling, providing integrity and
authentication for the phone.
• Encrypted—Cisco Unified Communications Manager uses TLS signaling, providing integrity and
authentication for the phone. In addition, SRTP encrypts the media streams.
Step 7 Check the Enable Digest Authentication check box.

Step 8 Configure the remaining fields in the Phone Security Profile Configuration window. For help with the fields
and their settings, see the online help.
Step 9 Click Save.
Configure AS-SIP Endpoint

Use this procedure to configure an AS-SIP endpoint. Many Cisco IP Phones support AS-SIP. In addition, you
can configure AS-SIP for third-party endpoints.
Procedure

Step 3 From the Phone Type drop-down list, select a Cisco IP Phone that supports AS-SIP. Otherwise, select
Third-Party AS-SIP Endpoint.
Step 4 Click Next.
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Associate Device with End User
Step 5 Configure the following mandatory fields.For more information on the fields and their configuration options,
see Online Help.
• Device Trust Mode—For third-party AS-SIP endpoints only. Select Trusted or Not Trusted.
• MAC Address
• Device Pool
• Phone Button Template
• Owner User ID
• Device Security Profile—Select the phone security profile that you set up for AS-SIP.
• SIP Profile—Select the AS-SIP-enabled SIP Profile that you configured.
• Digest User—Select the user ID that you configure as a digest user. The user must be enabled for digest
authentication
• Require DTMF Reception—Check this check box to allow the endpoint to accept DTMF digits.
• Early Offer support for voice and video calls—Check this check box to enable early offer support. This
field appears for third-party phones only.
Step 6 Configure the fields in the MLPP and Confidential Access Level Information section.
Step 7 Click Save.
Step 8 Add a Directory Number:
a) In the left navigation bar, click Add a new DN. The Directory Number Configuration window opens.
b) Add a Directory Number.
c) Complete any remaining fields in the Directory Number Configuration window
d) Click Save.
Step 9 From Related Links, select Configure Device and click Go.
Associate Device with End User

Use this procedure to associate an end user to the AS-SIP endpoint.
Procedure
Step 2 Click Find and select the user whom you want to associate to the device.
Step 3 In the Device Information section, click Device Association.
The User Device Association window appears.
Step 4 Click Find to view a list of available devices.
Step 5 Select the device that you want to associate, and click Save Selected/Changes.
Step 6 From Related Links, choose Back to User, and click Go.
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Configure SIP Trunk Security Profile for AS-SIP
Configure SIP Trunk Security Profile for AS-SIP

Use this procedure to configure a security profile for a SIP trunk that supports AS-SIP
Procedure
Step 1 From Cisco Unified CM Administration, choose System > Security > SIP Trunk Security Profile.
Step 3 Enter a Name for the security profile.
Step 4 From the Device Security Mode drop-down list, choose Authenticated or Encrypted.
Step 5 The Incoming Transport Type and Outgoing Transport Type fields change to TLS automatically.
Step 6 Check the Enable Digest Authentication check box.
Step 7 If you are deploying V.150, configure a value for the SIP V.150 Outbound SDP Offer Filtering drop-down
list.
Step 8 Complete the remaining fields in the SIP Trunk Security Profile Configuration window.For more information
Step 9 Click Save.
Configure SIP Trunk for AS-SIP

Use this procedure to set up a SIP trunk that supports AS-SIP.
Procedure

• Click Add New to create a new trunk.
Step 3 For new trunks, from the Trunk Type drop-down list, select SIP Trunk.
Step 4 From the Trunk Service Type drop-down list, select None (Default) and click Next.
Step 5 Enter a Device Name for the trunk.
Step 6 From the Device Pool drop-down list, select a device pool.
Step 7 In the Destination Address field, enter the address of the server to which you are connecting the trunk.
Step 8 From the SIP Trunk Security Profile drop-down list, select the profile that you created for AS-SIP.
Step 9 From the SIP Profile drop-down list, select the SIP Profile that you set up for AS-SIP.
Step 10 Complete any remaining fields in the Trunk Configuration window. For more information on the fields and
Step 11 Click Save.
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Configure AS-SIP Features

The procedures in the preceding task flow describe how to configure AS-SIP support on endpoints and trunk.
The following table outlines the AS-SIP features that you can deploy and provides configuration reference
for each.
AS-SIP Feature Configuration Description
Early Offer SIP Early Offer allows your endpoints to negotiate media during the INVITE
request and the 200OK response. There are two modes for Early Offer:
• Best Effort Early Offer (no MTP Inserted)
• Mandatory Early Offer (insert MTP if needed)
Configure Early Offer support via the fields in the following configuration
windows. Refer to the online help for detailed field descriptions:
SIP Profile Configuration window
• Early Offer support for voice and video calls—Configure this field to enable
Early Offer support on a SIP trunk
• SDP Session-level Bandwidth Modifier for Early Offer and Re-invite
• Send send-receive SDP in mid-call INVITE
Phone Configuration window (only if the Third Pary AS-SIP Endpoint device
type is used)
• Early Offer support for voice and video calls - check this check box to enable
early offer support
Conference Factory Specify the URI that an IMS client uses to set up a conference:
1. From Cisco Unified CM Administration, choose System > Service
Parameters.
2. From the Server drop-down list select your Cisco Unified Communications
Manager server.
3. From the Service, select Cisco CallManager.
4. Under Clusterwide Paramters (Feature - Conference) assign an IMS
Conference Factory URI.
5. Click Save.
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AS-SIP Feature Configuration Description
DSCP Markings DSCP settings allow you to manage QoS and bandwidth within your network.
DSCP settings are used to assign a prioritized Traffic Class Label to calls on a
per-call basis.
You can configure clusterwide DSCP settings via service parameters and you
can use the SIP Profile to assign a customized QoS policy for users whom use
that profile. For example, you could assign higher priority for the calls of an
executive (for example, a CEO) or a sales team to ensure that their calls are not
dropped if network bandwidth issues arise.
To configure DSCP, see DSCP Settings Configuration Task Flow, on page 731.
IPv6 By default, Cisco Unified Communications Manager is configured to use IPv4

addressing. However, you can configure the system to support the IPv6 stack
thereby allowing you to deploy a SIP network with IPv6-only endpoints.
For more information to configure IPv6, see "Dual Stack IPv6 Configuration
Task Flow" chapter in the System Configuration Guide for Cisco Unified
Multilevel Precedence and The Multilevel Precedence and Preemption (MLPP) service allows placement of
Preemption (MLPP) priority calls. This capability assures high-ranking personnel of communication
to critical organizations and personnel during network stress situations, such as
a national emergency or degraded network situations.
To configure MLPP, see Multilevel Precendence and Preemption Task Flow, on
page 781.
Secure Real-Time The Secure Real-time Transport Protocol (SRTP) can be used to provide
Transport Protocol encryption and authentication to media streams in your calls.
(SRTP)
SRTP can be configured for phones within the Phone Security Profile
Configuration that the phone uses. You must set the Device Security Mode
field to Encrypted.
Transport Layer Transport Layer Security (TLS) provides secure and reliable signaling and data
Signalling (TLS) transfer between two systems or devices, by using secure ports and certificate
exchange.
For more information to configure TLS, see the "TLS Setup" chapter in the
Security Guide for Cisco Unified Communications Manager.
V.150 The V.150 Minimum Essential Requirements feature allows you to make secure
calls in a modem over IP network. The feature uses a dialup modem for large
installed bases of modems and telephony devices operating on a traditional public
switched telephone network (PSTN).
For more information to configure V.150, see the "Cisco V.150 Minimum
Essential Requirements (MER)" chapter in the Security Guide for Cisco Unified
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CHAPTER 66
Configure Multilevel Precedence and Preemption
• Multilevel Precedence and Preemption Overview, on page 781
• Multilevel Precedence and Preemption Prerequisites, on page 781
• Multilevel Precendence and Preemption Task Flow, on page 781
• Multilevel Precedence and Preemption Interactions, on page 796
• Multilevel Precedence and Preemption Restrictions, on page 797
Multilevel Precedence and Preemption Overview

The Multilevel Precedence and Preemption (MLPP) service allows placement of priority calls. Properly
validated users can preempt lower priority phone calls with higher priority calls. An authenticated user can
preempt calls either to targeted stations or through fully subscribed TDM trunks. This capability assures
high-ranking personnel of communication to critical organizations and personnel during network stress
situations, such as a national emergency or degraded network situations.
Multilevel Precedence and Preemption Prerequisites

Supported SCCP or SIP phones. See the Cisco IP Phone Administration Guide and Cisco IP Phone User
Guide for your phones for feature support and more information.
Multilevel Precendence and Preemption Task Flow

Before you begin
Procedure

Step 1 To Configure Domains and Domain Lists, on Configure an MLPP domain to specify the
page 783, perform the following subtasks: devices and resources that are associated with
an MLPP subscriber.
• Configure a Multilevel Precedence and
Preemption Domain, on page 784
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Multilevel Precendence and Preemption Task Flow

• Configure a Resource Priority Namespace
Network Domain, on page 784
• Configure a Resource Priority Namespace
Network Domain List, on page 785
Step 2 Configure a Common Device Configuration A common device configuration includes

for Multilevel Precedence and Preemption, on MLPP-related information that can be applied
page 785 to multiple users and their devices. Ensure that
each device is associated with a common
device configuration. These settings override
the enterprise parameter settings.
Step 3 Configure the Enterprise Parameters for Set enterprise parameters to enable MLPP
Multilevel Precedence and Preemption, on indication and preemption. If individual
page 786 devices and devices in common device
configurations have MLPP settings of Default,
the MLLP-related enterprise parameters apply
to these devices and common device
configurations.
Step 4 Configure a Partition for Multilevel Configure a partition to create a logical

Precedence and Preemption, on page 787 grouping of directory numbers (DNs) and route
patterns with similar reachability
characteristics. Devices that are typically
placed in partitions include DNs and route
patterns. These entities associate with DNs that
users dial. For simplicity, partition names
usually reflect their characteristics.
Step 5 Configure a Calling Search Space for A calling search space is an ordered list of
Multilevel Precedence and Preemption, on partitions. Calling search spaces determine the
page 788 partitions that calling devices, including IP
phones, softphones, and gateways, can search
when attempting to complete a call.
Step 6 Configure a Route Pattern for Multilevel Configure route patterns to route or block both
Precedence and Preemption, on page 789 internal and external calls.
Step 7 Configure a Translation Pattern for Multilevel Configure translation patterns to specify how
Precedence and Preemption, on page 790 to route a call after it is placed. Configuring
translation patterns allows your system to
manipulate calling and called digits as needed.
When the system identifies that a pattern match
occurred, your system uses the calling search
space that is configured for the translation
pattern to perform the subsequent match.
Step 8 Configure Multilevel Precedence and Configure Cisco Unified Communications

Preemption for Gateways, on page 791 Manager to communicate with non-IP
telecommunications devices.
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Configure Domains and Domain Lists

Step 9 Configure Multilevel Precedence and
Preemption for Phones, on page 792
Step 10 Configure a Directory Number to Place After you configure a device, you can add a
Multilevel Precedence and Preemption Calls, line (directory number) from the updated
on page 794 Device Configuration window.
Step 11 Configure a User Device Profile for Multilevel When a user profile is assigned to a phone, the
Precedence and Preemption, on page 794 phone inherits the configuration of the assigned
user, including any CSS that is associated with
the user. The phone CSS can, however,
override the user profile. Cisco Unified
Communications Manager assigns the
precedence level that is associated with the
dialed pattern to the call when a pattern match
occurs. The system sets the call request as a
precedence call with the assigned precedence
level.
Step 12 Configure the Default Device Profile for Use the default device profile for whenever a
Multilevel Precedence and Preemption, on user logs on to a phone model for which no
page 795 user device profile exists. A default device
profile comprises the set of services and
features that are associated with a particular
device.
Configure Domains and Domain Lists

Configure an MLPP domain to specify the devices and resources that are associated with an MLPP subscriber.
Procedure

Step 1 Configure a Multilevel Precedence and Associate devices and resources with an MLPP
Preemption Domain, on page 784 subscriber. When an MLPP subscriber that
belongs to a particular domain places a
precedence call to another MLPP subscriber
that belongs to the same domain, the MLPP
service can preempt the existing call that the
called MLPP subscriber is on for a higher
precedence call. MLPP service availability does
not span across different domains.
The MLPP domain subscription of the
originating user determines the domain of the
call and its connections. Only higher precedence
calls in one domain can preempt connections
that calls in the same domain are using.
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Configure a Multilevel Precedence and Preemption Domain

Step 2 Configure a Resource Priority Namespace Configure namespace domains for a Voice over
Network Domain, on page 784 Secured IP (VoSIP) network that uses SIP
trunks. Your system prioritizes the SIP-signaled
resources so that those resources can be used
most effectively during emergencies and
congestion of telephone circuits, IP bandwidth,
and gateways. Endpoints receive the precedence
and preemption information.
Step 3 Configure a Resource Priority Namespace Configure a list of acceptable network domains.
Network Domain List, on page 785 Incoming calls are compared to the list and
processed, if an acceptable network domain is
in the list.
Configure a Multilevel Precedence and Preemption Domain

Associate devices and resources with an MLPP subscriber. When an MLPP subscriber that belongs to a
particular domain places a precedence call to another MLPP subscriber that belongs to the same domain, the
MLPP service can preempt the existing call that the called MLPP subscriber is on for a higher precedence
call. MLPP service availability does not span across different domains.
The MLPP domain subscription of the originating user determines the domain of the call and its connections.
Only higher precedence calls in one domain can preempt connections that calls in the same domain are using.
Procedure
Step 1 From Cisco Unified CM Administration, choose System > MLPP > Domain > MLPP Domain.
Step 3 In the Domain Name field, enter the name that you want to assign to the new MLPP domain.
You can enter up to 50 alphanumeric characters, and any combination of spaces, periods (.), hyphens (-), and
underscore characters (_).
Step 4 In the Domain ID field, enter a unique six-character hexadecimal MLPP domain ID.
Domain IDs must fall in the range between 000001 and FFFFFF. (000000 is reserved for the default MLPP
domain ID.)
Step 5 Click Save.
Configure a Resource Priority Namespace Network Domain

Configure namespace domains for a Voice over Secured IP (VoSIP) network that uses SIP trunks. Your system
prioritizes the SIP-signaled resources so that those resources can be used most effectively during emergencies
and congestion of telephone circuits, IP bandwidth, and gateways. Endpoints receive the precedence and
preemption information.
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Configure a Resource Priority Namespace Network Domain List
Procedure
Step 1 From Cisco Unified CM Administration, System > MLPP > Namespace > Resource Priority Namespace
Network Domain.
Step 2 Enter the name for the Resource Priority Namespace Network Domain in the information section. The maximum
number of domain names is 100.
Step 3 Enter a description for the domain name.
percentage sign (%), ampersand (&), or angle brackets (<>).
Step 4 Check the Make this the Default Resource Priority Namespace Network Domain check box if you want
the domain name to be the default.
Step 5 Click Save.
Configure a Resource Priority Namespace Network Domain List

Configure a list of acceptable network domains. Incoming calls are compared to the list and processed, if an
acceptable network domain is in the list.
Procedure
Step 1 From Cisco Unified CM Administration, choose System > MLPP > Namespace > Resource Priority
Namespace List.
Step 2 Enter the name for the Resource Priority Namespace List. The maximum number of characters is 50.
Step 3 Enter a description for the list. The description can include up to 50 characters in any language, but it cannot
include double-quotes ("), percentage sign (%), ampersand (&), back-slash (\), or angle brackets (<>).
Step 4 Use the Up and Down Arrows to move a Resource Priority Namespace Network Domain to the Selected
Resource Priority Namespaces field.
Step 5 Click Save.
Configure a Common Device Configuration for Multilevel Precedence and

Preemption
A common device configuration includes MLPP-related information that can be applied to multiple users and
their devices. Ensure that each device is associated with a common device configuration. These settings
override the enterprise parameter settings.
Procedure
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Configure the Enterprise Parameters for Multilevel Precedence and Preemption
• Click Find to modify an existing common device configuration and choose a common device configuration
• Click Add New to add a new common device configuration.
Step 3 Configure the fields on the Common Device Configuration window. For more information on the fields and
Step 4 Click Save.
ConfiguretheEnterpriseParametersforMultilevelPrecedenceandPreemption
Set enterprise parameters to enable MLPP indication and preemption. If individual devices and devices in
common device configurations have MLPP settings of Default, the MLLP-related enterprise parameters apply
to these devices and common device configurations.
Procedure
Step 1 Choose System > Enterprise Parameters.

Step 2 Configure the MLPP enterprise parameters on the Enterprise Parameters Configuration window. See the
Related Topics section for more information about the parameters and their configuration options.
Step 3 Click Save.
Enterprise Parameters for Multilevel Precedence and Preemption

Table 78: Enterprise Parameters for Multilevel Precedence and Preemption
MLPP Domain Identifier Set this parameter to define a domain. Because MLPP service applies to a
domain, Cisco Unified Communications Manager marks only connections and
resources that belong to calls from MLPP users in a given domain with a
precedence level. Cisco Unified Communications Manager can preempt only
lower precedence calls from MLPP users in the same domain.
The default is 000000.
MLPP Indication Status This parameter specifies whether devices use MLPP tones and special displays
to indicate MLPP precedence calls. To enable MLPP indication across the
enterprise, set this parameter to MLPP Indication turned on.
The default is MLPP Indication turned off.
MLPP Preemption Setting This parameter determines whether devices should apply preemption and
preemption signaling (such as preemption tones) to accommodate higher
precedence calls. To enable MLPP preemption across the enterprise, set this
parameter to Forceful Preemption.
The default is No preemption allowed.
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Configure a Partition for Multilevel Precedence and Preemption
Precedence Alternate Party In a precedence call, if the called party subscribes to alternate party diversion,
Timeout this timer indicates the seconds after which Cisco Unified Communications
Manager will divert the call to the alternate party if the called party does not
acknowledge preemption or does not answer a precedence call.
The default is 30 seconds.
Use Standard VM Handling This parameter determines whether a precedence call will forward to the
For Precedence Calls voice-messaging system.
If the parameter is set to False, precedence calls do not forward to the
voice-messaging system. If the parameter is set to True, precedence calls forward
to the voice-messaging system.
For MLPP, the recommended setting for this parameter is False, as users, not
the voice-messaging system, should always answer precedence calls.
The default is False.
Configure a Partition for Multilevel Precedence and Preemption

Configure a partition to create a logical grouping of directory numbers (DNs) and route patterns with similar
reachability characteristics. Devices that are typically placed in partitions include DNs and route patterns.
These entities associate with DNs that users dial. For simplicity, partition names usually reflect their
characteristics.
Procedure
name in this field.
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Partition Naming Guidelines
incoming call.
Step 8 Click Save.
Partition Naming Guidelines

2 characters 340
3 characters 256
4 characters 204
5 characters 172
... ...
10 characters 92
15 characters 64
Configure a Calling Search Space for Multilevel Precedence and Preemption

A calling search space is an ordered list of partitions. Calling search spaces determine the partitions that calling
devices, including IP phones, softphones, and gateways, can search when attempting to complete a call.
Procedure
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Configure a Route Pattern for Multilevel Precedence and Preemption
Partitions box.
Step 8 Click Save.
Configure a Route Pattern for Multilevel Precedence and Preemption

Configure route patterns to route or block both internal and external calls.
Procedure
• To modify the settings for an existing route pattern, enter search criteria, click Find, and then choose an
existing route pattern from the resulting list.
• To add a new route pattern, click Add New.
Step 3 Configure the fields on the Route Pattern Configuration window. See the Related Topics section for more
Step 4 Click Save.
Route Pattern Configuration Fields for Multilevel Precedence and Preemption

Table 80: Route Pattern Configuration Fields for Multilevel Precedence and Preemption
Field Description
Route Pattern Enter the route pattern, including numbers and wildcards, without spaces. For
example, for NANP, enter 9.@ for typical local access or 8XXX for a typical private
network numbering plan. Valid characters include the uppercase characters A, B, C,
and D and \+, which represents the international escape character +.
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Configure a Translation Pattern for Multilevel Precedence and Preemption
Field Description
MLPP Precedence Choose an MLPP precedence setting for this route pattern from the drop-down list:
• Executive Override—Highest precedence setting for MLPP calls.
• Flash Override—Second highest precedence setting for MLPP calls.
• Flash—Third highest precedence setting for MLPP calls.
• Immediate—Fourth highest precedence setting for MLPP calls.
• Priority—Fifth highest precedence setting for MLPP calls.
• Routine—Lowest precedence setting for MLPP calls.
• Default-—Does not override the incoming precedence level but rather lets it
pass unchanged.
Apply Call Blocking Check this check box to enable the Destination Code Control (DCC) feature. By
Percentage enabling DCC, all calls other than flash and higher precedence calls made to the
destination are filtered and allowed or disallowed based on the Call Blocking
Percentage quota set for the destination. Flash and higher precedence calls are allowed
at all times. DCC is disabled by default.
The Apply Call Blocking Percentage field is enabled only if the MLPP level is
immediate, priority, routine or default.
Call Blocking Enter the percentage of calls to be blocked for this destination in numerals. This
Percentage (%) value specifies the percentage of lower precedence calls made to this destination that
get blocked by the route pattern. This percentage limits the lower precedence calls
only; the flash and higher precedence calls made to this destination are allowed at
all times
The Call Blocking Percentage (%) field is enabled only if the Apply Call Blocking
Percentage check box is checked.
Resource Priority Choose a Resource Priority Namespace Network Domain from the drop-down list.
Namespace Network To configure the Resource Priority Namespace Network Domains, choose System
Domain > MLPP > Namespace > Resource Priority Namespace Network Domain.
Configure a Translation Pattern for Multilevel Precedence and Preemption

Configure translation patterns to specify how to route a call after it is placed. Configuring translation patterns
allows your system to manipulate calling and called digits as needed. When the system identifies that a pattern
match occurred, your system uses the calling search space that is configured for the translation pattern to
perform the subsequent match.
Procedure
Step 1 In Cisco Unified CM Administration, choose Call Routing > Translation Pattern.
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Configure Multilevel Precedence and Preemption for Gateways
• To modify the settings for an existing translation pattern, enter search criteria, click Find, and choose
an existing Translation Pattern from the resulting list.
• To add a new translation pattern, click Add New.
Step 3 From the MLPP Precedence drop-down list, choose one of the following settings for this translation pattern:
• Executive Override—Highest precedence setting for MLPP calls.
• Flash Override—Second highest precedence setting for MLPP calls.
• Flash—Third highest precedence setting for MLPP calls.
• Immediate—Fourth highest precedence setting for MLPP calls.
• Priority—Fifth highest precedence setting for MLPP calls.
• Routine—Lowest precedence setting for MLPP calls.
• Default—Does not override the incoming precedence level but rather lets it pass unchanged.
Step 4 From the Resource-Priority Namespace Network Domain drop-down list, choose a resource priority
namespace network domain that you configured.
Step 5 From the Calling Search Space drop-down list, choose the calling search space that you configured.
Step 6 Click Save.
Configure Multilevel Precedence and Preemption for Gateways

Configure Cisco Unified Communications Manager to communicate with non-IP telecommunications devices.
Before you begin

• Configure one of the following gateways:
• Cisco Catalyst 6000 24 port FXS Gateway
• Cisco Catalyst 6000 E1 VoIP Gateway
• Cisco Catalyst 6000 T1 VoIP Gateway
• Cisco DE-30+ Gateway
• Cisco DT-24+ Gateway
• H.323 Gateway
Procedure
Step 1 From Cisco Unified CM Administration, choose Device > Gateway

• To modify the settings for an existing gateway, enter search criteria, click Find, and choose a gateway
• To add a new gateway:
a. Click Add New.
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Configure Multilevel Precedence and Preemption for Phones
b. From the Gateway Type drop-down list, choose one of the supported gateway models.
c. Click Next.
Step 3 Configure the MLPP fields on the Gateway Configuration window. See the Related Topics section for more
Step 4 Click Save.
Configure Multilevel Precedence and Preemption for Phones
Caution Do not configure a device with the following combination of settings: MLPP Indication is set to Off or Default
(when default is Off) while MLPP Preemption is set to Forceful.
Procedure

Step 2 Enter search criteria.
Step 3 Click Find and choose a phone from the resulting list.
Step 4 Configure the MLPP fields on the Phone Configuration window. See the Related Topics section for more
Multilevel Precedence and Preemption Settings for Phones

Table 81: Multilevel Precedence and Preemption Settings for Phones
MLPP Settings for Phones Description

Field
Common Device Configuration Choose the common device configuration that you configured. The common
device configuration includes the attributes (services or features) that are
associated with a particular user.
Calling Search Space From the drop-down list, choose a calling search space (CSS) that you
configured . A calling search space comprises a collection of partitions
that are searched to determine how a dialed number should be routed. The
calling search space for the device and the calling search space for the
directory number are used together. The directory number CSS takes
precedence over the device CSS.
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Multilevel Precedence and Preemption Settings for Phones
MLPP Settings for Phones Description

Field
MLPP Domain Choose an MLPP domain from the drop-down list for the MLPP domain
that is associated with this device. If you leave the None value, this device
inherits its MLPP domain from the value that was set in the common device
configuration. If the common device configuration does not have an MLPP
domain setting, this device inherits its MLPP domain from the value that
was set for the MLPP Domain Identifier enterprise parameter.
MLPP Indication If available, this setting specifies whether a device that can play precedence
tones will use the capability when it places an MLPP precedence call.
From the drop-down list, choose a setting to assign to this device from the
following options:
• Default—This device inherits its MLPP indication setting from the
common device configuration.
• Off—This device does not handle nor process indication of an MLPP
precedence call.
• On—This device handles and processes indication of an MLPP
precedence call.
Note Do not configure a device with the following combination of

settings: MLPP Indication is set to Off or Default (when
default is Off) while MLPP Preemption is set to Forceful.
Turning on MLPP Indication (at the enterprise parameter or
device level) disables normal Ring Setting behavior for the
lines on a device, unless MLPP Indication is turned off
(overridden) for the device.
MLPP Preemption Be aware that this setting is not available on all devices. If available, this
setting specifies whether a device that can preempt calls in progress will
use the capability when it places an MLPP precedence call.
From the drop-down list, choose a setting to assign to this device from the
following options:
• Default
—This device inherits its MLPP preemption setting from the common
device configuration.
• Disabled—This device does not allow preemption of lower precedence
calls to take place when necessary for completion of higher precedence
calls.
• Forceful—This device allows preemption of lower precedence calls
to take place when necessary for completion of higher precedence
calls.
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Configure a Directory Number to Place Multilevel Precedence and Preemption Calls
Configure a Directory Number to Place Multilevel Precedence and Preemption

Calls
After you configure a device, you can add a line (directory number) from the updated Device Configuration
window.
Procedure
Step 1 From Cisco Unified CM Administration in the Device Configuration window, click Add a new DN for the
appropriate line.
Step 2 In the Target (Destination) field, enter the number to which MLPP precedence calls should be diverted if
this directory number receives a precedence call and neither this number nor its call forward destination
answers the precedence call.
Values can include numeric characters, octothorpe (#), and asterisk (*).
Step 3 From the MLPP Calling Search Space drop-down list, choose the calling search space to associate with the
MLPP alternate party target (destination) number.
Step 4 In the MLPP No Answer Ring Duration (seconds), enter the number of seconds (between 4 and 60) after
which an MLPP precedence call is directed to this directory number alternate party if this directory number
and its call-forwarding destination have not answered the precedence call.
Leave this setting blank to use the value that is set in the Precedence Alternate Party Timeout enterprise
parameter.
Step 5 Click Save.
Configure a User Device Profile for Multilevel Precedence and Preemption

When a user profile is assigned to a phone, the phone inherits the configuration of the assigned user, including
any CSS that is associated with the user. The phone CSS can, however, override the user profile. Cisco Unified
Communications Manager assigns the precedence level that is associated with the dialed pattern to the call
when a pattern match occurs. The system sets the call request as a precedence call with the assigned precedence
level.
Procedure
• To modify the settings for an existing device profile, enter search criteria, click Find, and then choose
an existing device profile from the resulting list.
• To add a new device profile:
• Click Add New.
• From the Device Profile Type drop-down list, choose a profile type.
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• Click Next.
• From the Device Protocol drop-down list, choose either SIP or SCCP.
Step 3 Click Next.

Step 4 From the MLPP Domain drop-down list, choose an MLLP domain that you configured.
Step 5 From the MLPP Indication drop-down list, choose one of the following settings to specify whether a device
that is capable of playing precedence tones will use the capability when it places an MLPP precedence call:
• • Default—This device inherits its MLPP indication setting from its device pool.
• Off—This device does not handle nor process indication of an MLPP precedence call.
• On—This device does handle and process indication of an MLPP precedence call.
Step 6 From the MLPP Preemption drop-down list, choose one of the following settings to specify whether a device
that is capable of preempting calls in progress will use the capability when it places an MLPP precedence
call:
• • Default—This device inherits its MLPP preemption setting from its device pool.
• Disabled—This device does not allow preemption of lower precedence calls to take place when
necessary for completion of higher precedence calls.
• Forceful—This device allows preemption of lower precedence calls to take place when necessary
for completion of higher precedence calls.
Step 7 Click Save.
ConfiguretheDefaultDeviceProfileforMultilevelPrecedenceandPreemption
Use the default device profile for whenever a user logs on to a phone model for which no user device profile
exists. A default device profile comprises the set of services and features that are associated with a particular
device.
Caution Do not configure a default device profile with the following combination of settings: MLPP Indication is set
to Off or Default (when default is Off) while MLPP Preemption is set to Forceful.
Procedure
Step 1 In Cisco Unified CM Administration, choose Device > Device Settings > Default Device Profile.
• To modify the settings for an existing default device profile, choose an existing default device profile
from the Device Profile Defaults section.
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Multilevel Precedence and Preemption Interactions
• To add a new default device profile, choose a device profile type from the drop-down list, click Next,
choose a device protocol, and then click Next.
Step 3 From the MLPP Domain drop-down list, choose an MLPP domain that you configured to associate to the
device.
Step 4 From the MLPP Indication drop-down list, choose one of the following settings to specify whether a device
that is capable of playing precedence tones will use the capability when it places an MLPP precedence call:
• Default—This device inherits its MLPP indication setting from its device pool.
• Off—This device does not handle nor process indication of an MLPP precedence call.
• On—This device does handle and process indication of an MLPP precedence call.
Step 5 From the MLPP Preemption drop-down list, choose one of the following settings to specify whether a device
that is capable of preempting calls in progress will use the capability when it places an MLPP precedence
call:
• Default—This device inherits its MLPP preemption setting from its device pool.
• Disabled—This device does not allow preemption of lower precedence calls to take place when necessary
for completion of higher precedence calls.
• Forceful—This device allows preemption of lower precedence calls to take place when necessary for
completion of higher precedence calls.
Step 6 Click Save.
Multilevel Precedence and Preemption Interactions

Table 82: Multilevel Precedence and Preemption Interactions
Feature Interaction
729 Annex A 729 Annex A is supported.
Cisco Extension Mobility The MLPP service domain remains associated with a user device profile
when a user logs in to a device by using extension mobility. The MLPP
Indication and Preemption settings also propagate with extension mobility.
If either the device or the device profile do not support MLPP, these settings
do not propagate.
Cisco Unified MLPP interacts with Cisco Unified Communications Manager Assistant as
Communications Manager follows:
Assistant
• When Cisco Unified Communications Manager Assistant handles an
MLPP precedence call, Cisco Unified Communications Manager
Assistant preserves call precedence.
• Cisco Unified Communications Manager Assistant filters MLPP
precedence calls in the same manner as it filters all other calls. The
precedence of a call does not affect whether the call is filtered.
• Because Cisco Unified Communications Manager Assistant does not
register the precedence of a call, it does not provide any additional
indication of the precedence of a call on the assistant console.
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Multilevel Precedence and Preemption Restrictions
Feature Interaction
Immediate Divert Immediate Divert diverts calls to voice-messaging mail boxes regardless of
the type of call (for example, a precedence call). When Alternate Party
Diversion (call precedence) is activated, Call Forward No Answer (CFNA)
is also deactivated.
Resource Reservation Protocol RSVP supports MLPP inherently. The Cisco Unified Communications
(RSVP) Manager System Guide explains how MLPP functions when RSVP is
activated.
Supplementary Services MLPP interacts with multiple line appearances, call transfer, call forwarding,
three-way calling, call pickup, and hunt pilots as documented in the and the
subsections that describe the interaction with each service.

Table 83: Multilevel Precedence and Preemption Restrictions
Bandwidth Cisco Unified Communications Manager preempts lower precedence calls

when adjusting video bandwidth for high priority calls. If the bandwidth
is not sufficient to preempt, Cisco Unified Communications Manager
instructs endpoints to use previously reserved lower video bandwidth.
When Cisco Unified Communications Manager preempts a video call,
the preempted party receives a preemption tone and the call gets cleared.
Call Detail Records For the DRSN, CDRs represent precedence levels with values 0, 1, 2, 3,
and 4 where 0 specifies Executive Override and 4 specifies Routine, as
used in DSN. CDRs thus do not use the DRSN format.
Common Network Facility Common Network Facility Preemption support exists only for T1-CAS
Preemption and T1-PRI (North American) interfaces on targeted Voice over IP
gateways that Cisco Unified Communications Manager controls by using
MGCP protocol and that have been configured as MLPP Preemption
Enabled.
Intercluster trunks Intercluster trunk MLPP carries precedence information through dialed
digits. Domain information does not get preserved and must be configured
per trunk for incoming calls.
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QoS Management
Line Groups MLPP-enabled devices are not supported in line groups. We recommend
the following guidelines:
• MLPP-enabled devices should not be configured in a line group.
Route groups, however, are supported. Both trunk selection and
hunting methods are supported.
• If an MLPP-enabled device is configured in a line group or route
group, in the event of preemption, if the route list does not lock onto
the device, the preempted call may be rerouted to other devices in
the route/hunt list and preemption indication may be returned only
after no devices are able to receive the call.
• Route lists can be configured to support either of two algorithms of
trunk selection and hunting for precedence calls. In method 1,
perform a preemptive search directly. In method 2, first perform a
friendly search. If this search is not successful, perform a preemptive
search. Method 2 requires two iterations through devices in a route
list. If route lists are configured for method 2, in certain scenarios
involving line groups, route lists may seem to iterate through the
devices twice for precedence calls.
Look Ahead For Busy Cisco Unified Communications Manager does not support the Look Ahead
for Busy (LFB) option.
MLPP Notification Only MLPP Indication Enabled devices generate MLPP-related

notifications, such as tones and ringers. If a precedence call terminates at
a device that is not MLPP Indication Enabled, no precedence ringer gets
applied. If a precedence call originates from a device that is not MLPP
Indication Enabled, no precedence ringback tone gets applied. If a device
that is not MLPP Indication Enabled is involved in a call that is preempted
(that is, the other side of the call initiated preemption), no preemption
tone gets applied to the device.
Phones and trunks For phones, devices that are MLPP indication disabled (that is, MLPP
Indication is set to Off) cannot be preempted. For trunks, MLPP indication
and preemption function independently.
Ring Setting Behavior Turning on MLPP Indication (at the enterprise parameter, common device
configuration, or device level) disables normal Ring Setting behavior for
the lines on a device, unless MLPP Indication is turned off (overridden)
for the device.
SCCP IOS gateways support the SCCP interface to Cisco Unified
Communications Manager. They support BRI and analog phones which
appear on Cisco Unified Communications Manager as supported phone
models. SCCP phones support the MLPP feature, and so do some phones
with specific SIP loads. See the relevant phone administration and user
guides for Cisco IP phone support information.
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QoS Management
Supplementary Services MLPP support for supplementary services specifies the following
restrictions:
• MLPP addresses only the basic Call Pickup feature and Group Call
Pickup feature, not Other Group Pickup.
• Call Forward All (CFA) support for inbound MLPP calls always
forwards the call to the MLPP Alternate Party (MAP) target of the
called party, if the MAP target is configured. In the event of an
incorrect configuration (that is, if no MAP target is specified), the
call is rejected, and the calling party receives a reorder tone.
• Call Forward No Answer (CFNA) support for inbound MLPP calls
forwards the call once to a CFNA target. After the first hop, if the
call is unanswered, the call is sent to the MAP target of the original
called party, if the MAP target has been configured. In the event of
an incorrect configuration (that is, if no MAP target is specified),
the call gets rejected, and the calling party receives reorder tone.
• Call Forward Busy (CFB) support for inbound MLPP calls forwards
the call up to the maximum number that has been configured for
forwarding hops. If the maximum hop count gets reached, the call
gets sent to the MAP target of the original called party, if the MAP
target has been configured. In the event of an incorrect configuration
(that is, no MAP target is specified), the call gets rejected, and the
calling party receives reorder tone.
• For hunt pilot support, the hunt group algorithm must specify Longest
Idle Time, Top Down, or Circular. Ensure the hunt group options
for busy treatment, no answer treatment, and unregistered treatment
are set to Try next member, but do not go to next group. Preemption
only occurs across a single hunt group.
User Access Channel User Access Channel support exists only for the following Cisco Unified
IP Phone models, which must be configured as MLPP Preemption
Enabled:
• Cisco Unified IP Phone 7960, 7962, 7965
• Cisco Unified IP Phone 7940, 7942, 7945
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QoS Management
800
PA R T XVI
SIP Interoperability
• Configure SIP Normalization and Transparency, on page 803
• Configure SDP Transparency Profiles, on page 809
• Configure Presentation Sharing using BFCP, on page 811
• Video Telephony, on page 815
CHAPTER 67
Configure SIP Normalization and Transparency
• SIP Normalization and Transparency Overview, on page 803
• SIP Normalization and Transparency Prerequisites, on page 804
• SIP Normalization and Transparency Configuration Task Flow, on page 804
SIP Normalization and Transparency Overview

SIP normalization and transparency is an optional feature that handles SIP interoperability issues between
Unified Communications Manager and endpoints, service providers, PBXs, or gateways that implement SIP
differently. To configure SIP normalization and transparency, apply a customized LUA script to a SIP trunk
or SIP line. Unified Communications Manager applies the script to the SIP messaging that passes through the
SIP trunks or SIP lines.
Upon installation, Unified Communications Manager contains default normalization and transparency scripts
that you can assign to the SIP trunks and SIP profiles in your system. You can also create and import your
own customized scripts.
SIP Normalization
SIP normalization scripts modify incoming and outgoing SIP messages. For example, if you are interoperating
Unified Communications Manager with a Cisco TelePresence Video Communications Server, apply the
vcs-interop script to the SIP trunk that connects the two. The script resolves the differences in the SIP messaging
so that the two products can communicate.
You can apply a normalization script to any SIP trunk connection, regardless of which protocol is being used
by the endpoint that connects to that SIP trunk.
SIP Transparency
SIP transparency scripts enable Unified Communications Manager to transparently pass SIP information,
such as proprietary headers, from one call leg to the other. For transparency to work, both call legs must be
SIP.
Another feature of SIP transparency is REFER transparency, which allows Unified Communications Manager
to pass on REFER requests without acting on them. You can use REFER transparency in call center
environments where a centralized agent may answer a call and then transfer the call to an agent who resides
in the same geographical area as the caller. REFER transparency allows the centralized Unified Communications
Manager to drop the call and shift call control to the new agent.
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Default Scripts for SIP Normalization and Transparency
Default Scripts for SIP Normalization and Transparency

Upon installation, Cisco Unified Communications Manager contains the following default scripts for SIP
Normalization and Transparency. You can apply these scripts to a SIP trunk or SIP profile, but you cannot
edit these scripts. If none of these scripts meet your needs, you can create your own scripts:
• cisco-meeting-server-interop—Provides interoperability between Cisco Unified Communications
Manager and Cisco Meeting Server (CMS).
• cisco-telepresence-conductor-interop—Provides interoperability for endpoints that are registered to
TelePresence Conductor.
• cisco-telepresence-mcu-ts-direct-interop—Provides interoperability between Cisco Unified
Communications Manager and either Cisco TelePresence MCU or Cisco TelePresence Server.
• diversion-counter—Provides capability to adjust the diversion counter.
• HCS-PCV-PAI passthrough—Provides Cisco HCS platform integration with Enterprise IMS.
• refer-passthrough—Removes Cisco Unified Communications Manager from the call due to a blind
transfer between SIP trunks.
• vcs-interop—Provides interoperability for endpoints that are registered to the Cisco TelePresence Video
Communications Server.
SIP Normalization and Transparency Prerequisites

• Cisco Unified Communications Manager provides default scripts for SIP Normalization and Transparency.
Make sure to review the existing scripts and system settings to verify whether they meet your needs. For
information on the available default scripts, see Default Scripts for SIP Normalization and Transparency,
on page 804.
• Make sure that you understand your deployment's SIP requirements in addition to the SIP requirements
for any third-party products. For information on Cisco Unified Communications Manager's implementation
of SIP, review the SIP Line Messaging Guide for Cisco Unified Communications Manager (Standard
Edition) at http://www.cisco.com/c/en/us/support/unified-communications/
unified-communications-manager-callmanager/products-programming-reference-guides-list.html.
• If you plan to develop customized SIP Normalization scripts, review the Developer Guide for SIP
Normalization and Transparency at http://www.cisco.com/c/en/us/support/unified-communications/
unified-communications-manager-callmanager/products-programming-reference-guides-list.html.
SIP Normalization and Transparency Configuration Task Flow

Procedure

Step 1 Create New SIP Normalization and Optional. If none of the preinstalled scripts meet
Transparency Scripts, on page 805 your needs, use this procedure to configure a
customized script. You can create your new
804
Create New SIP Normalization and Transparency Scripts

script in the SIP Normalization Script
Configuration window or you can import a
customized script.
Step 2 Apply Normalization or Transparency Script In the Trunk Configuration window, apply a
to SIP Trunk, on page 806 script directly to a SIP trunk. Cisco Unified
Communications Manager applies the script to
all the SIP messaging that passes through the
trunk
Step 3 Apply Normalization or Transparency to SIP If you want to apply a normalization or

Devices, on page 806 transparency script to a SIP line, apply a script
to the SIP profile that is associated to that SIP
line. Cisco Unified Communications Manager
applies the script to all SIP messaging that uses
that SIP profile.
Create New SIP Normalization and Transparency Scripts

If the default normalization and transparency scripts do not meet your needs, use this procedure to create a
new LUA script. You can either write the new script in Cisco Unified Communications Manager or import a
file into the system.
Tip If the script that you want to create closely resembles a default script, open the default script in the SIP
Normalization Script Configuration window and copy the Contents text box. Create a new script and paste
the contents into the Contents text box. You can then edit the content in the new script.
Note The memory utilization of the SIP Normalization Script is based on each trunk and not on each script.
Procedure
Step 1 In Cisco Unified CM Administration, choose Device > Device Settings > SIP Normalization Script.
The SIP Normalization Script Configuration window appears.
Step 3 Enter a Name and Description for your script.
Step 4 If you are writing a new script, edit the script in the Contents text box.
Step 5 Optional. If you have an external file that you want to import, do the following
a) Click Import File.
b) Browse to locate the file and select the file.
c) Click Import File.
The SIP Normalization Script Configuration window displays the contents of the imported file in the
Contents text box.
805
Apply Normalization or Transparency Script to SIP Trunk
Step 6 Complete the fields in the SIP Normalization Script Configuration window. For help with the fields and
their contents, refer to the online help.
Step 7 Click Save.
What to do next
Assign the script to a SIP profile or SIP trunk:
• Apply Normalization or Transparency to SIP Devices, on page 806
• Apply Normalization or Transparency Script to SIP Trunk, on page 806
Apply Normalization or Transparency Script to SIP Trunk

Use this procedure to apply a SIP normalization or transparency script to a SIP trunk. Cisco Unified
Communications Manager applies the script to all SIP messaging that passes through the trunk.
Procedure

Step 2 Click Find and select the trunk to which you want to apply a script.
Step 3 From the Normalization Script drop-down list, choose the script that you want to apply to the trunk.
Step 4 (Optional) If you want to normalize specific parameters within the SIP messaging, do the following:
a) Enter the Parameter Name that you want to normalize, and the Parameter Value for the value that you
want to apply to the parameter. For example, you could enter a Location parameter and North
Carolina as the value.
b) To add additional parameters, click the (+) to create additional lines where you can enter additional
parameters and values.
Step 5 (Optional) If you want to produce SDI traces against the script, check the Enable Trace check box.
Note Cisco recommends that you enable tracing while debugging your scripts.
Step 6 Click Save.
Apply Normalization or Transparency to SIP Devices

You can apply a customized SIP Normalization and Transparency script, or a customized SDP Transparency
Profile to a SIP phone by applying the script to the SIP Profile that is used by that device.
Procedure
Step 2 Click Find and select the SIP profile to which you want to apply a script.
806
Step 3 In the SDP Information area, from the SDP Transparency Profile drop-down list, choose a profile.
Step 4 From the Normalization Script drop-down list, choose the script that you want to apply to the trunk.
Step 5 (Optional) If you want to normalize specific parameters within the SIP messaging, do the following:
a) Enter the Parameter Name that you want to normalize, and the Parameter Value for the value that you
want to apply to the parameter. For example, you could enter a Location parameter and North
Carolina as the value.
b) To add additional parameters, click the (+) to create additional lines where you can enter additional
parameters and values.
Step 6 (Optional) If you want to produce SDI traces against the script, check the Enable Trace check box.
Note Cisco recommends that you enable tracing while debugging your scripts.
Step 7 Click Save.
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808
CHAPTER 68
Configure SDP Transparency Profiles
• SDP Transparency Profile Overview, on page 809
• SDP Transparency Profile Restrictions, on page 809
• SDP Transparency Profile Prerequisites, on page 810
• Configure SDP Transparency Profile, on page 810
SDP Transparency Profile Overview

SDP Transparency Profiles contain a set of rules for declarative SDP attributes that allow the system to pass
through declarative attributes that are not natively supported by Unified Communications Manager from the
ingress to the egress call leg. Without an SDP transparency profile, Unified Communications Manager drops
non-supported SDP attributes.
You can configure SDP transparency profiles with multiple rules and apply them to SIP devices via the SIP
profile. In order for the SDP transparency profile to be applied, both call legs must be SIP. You can configure
the following types of rules for SDP attributes:
• Property—If a rule is configured for a property attribute, Unified Communications Manager passes
through the SDP attribute unless the attribute has a value.
• Any Value—If a rule is configured for any value, the SDP attribute gets passed through so long as it has
a value that consists of at least one non-white space character.
• Value From List—If a rule is configured using this option, the SDP attribute gets passed through so long
as it matches one of the specified values. You can configure up to five possible values
SDP Transparency Profile Restrictions

The following restrictions apply to SDP transparency profiles. If any of these situations occur on the egress
call leg, Cisco Unified Communications Manager will not pass through the declarative SDP attribute:
• One or more Media Termination Points (MTPs) or Trusted Relay Points (TRPs) that do not support
passthrough are allocated
• The Media Termination Point Required check box is checked for the SIP trunk
• A transcoder is being used
• RSVP is being used
809
SDP Transparency Profile Prerequisites
• The ingress call leg is using Delayed Offer while the egress call leg is using Early Offer
• The media line has been rejected (port=0)
• Either call leg is using a protocol other than SIP
SDP Transparency Profile Prerequisites

If you plan to deploy any third-party SIP products, make sure that you understand how the products implement
the Session Description Protocol (SDP).
Configure SDP Transparency Profile

Configure a customized SDP Transparency Profile with a set of rules for declarative SDP attributes that are
not natively supported by Cisco Unified Communications Manager.
Procedure
Step 1 From Cisco Unified CM Administration, choose Device > Device Settings > SDP Transparency Profile.
Step 3 Enter a Name and Description.
Step 4 In the Attribute Information pane, create the rules for the SDP attributes that you want to pass through:
• To pass through a property attribute, enter the attribute in the Name text box (for example, a=recvonly)
and from the Type drop-down list, select Property.
• To pass through a value attribute, enter the attribute in the Name text box (for example, a=rtpmap), and
select Any Value from the Type drop-down list box.
• To pass through a value attribute with any of up to five values, enter the attribute in the Name field (for
example, a=rtpmap) and select Any Value from the Type drop-down list. In the resulting Value text
box, enter the value of the attribute. You can click + to add up to five possible values for this attribute.
Step 5 Click the (+) to create new lines where you can enter additional SDP attributes for this transparency profile.
Step 6 Click Save.
Note You must apply this profile to a SIP Profile so that the devices that use the SIP Profile can use
the SDP Transparency Profile.
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CHAPTER 69
Configure Presentation Sharing using BFCP
• Binary Floor Control Protocol Overview, on page 811
• Presentation Sharing using BFCP Prerequisites, on page 812
• Presentation Sharing using BFCP Configuration Task Flow, on page 813
Binary Floor Control Protocol Overview

Unified Communications Manager supports presentation sharing using the Binary Floor Control Protocol
(BFCP) for supported Cisco endpoints and third-party video endpoints. This feature lets users share a
presentation within ongoing audio or video conversation.
The following example describes how presentation sharing works using BFCP:
• An ongoing video conversation exists between two video phones. User A decides to share content with
User B during the conversation. User A has the option to share the entire screen or share the specific
application.
• The BFCP stream allows User B to view User A's shared content.
An audio-video call with content share requires at least four channels: audio, main video, the second video,
and BFCP control channel, to achieve video conferencing and sharing presentations in the second video
channel. If the call parties are capable of Far-End Camera Control (FECC), a fifth channel must be established.
BFCP Architecture
Presentation sharing using BFCP is supported only on BFCP enabled SIP networks. The entire network,
including the endpoint devices and trunks, must be SIP.
Unified Communications Manager aids in the negotiation of the BFCP stream by relaying SIP messages
between two endpoints.
This negotiation involves establishing a floor, which is a temporary permission to access shared resources.
The BFCP stream is a point-to-point stream between the endpoints. Unified Communications Manager is
never a target of the BFCP stream.
The following figure provides an example of a complex video network with multiple Unified Communications
Manager clusters. BFCP must be enabled on all the trunks and lines connecting the devices. For this network,
BFCP must be enabled on the four SIP trunks and two SIP lines that connect the endpoints.
811
BFCP Limitations
Figure 14: Video Network with Multiple Cisco Unified Communications Manager Clusters
BFCP Limitations
Unified Communications Manager rejects the BFCP stream in the following scenarios:
• The Allow Presentation Sharing using BFCP check box on the SIP Profile page is unchecked for one
of the SIP lines or trunks in the network.
• One endpoint offers BFCP, but the other does not.
• When the SIP line or SIP trunk uses MTP (non pass-through mode) or Transcoder.
• For SIP line endpoints, the BFCP share does not work when,
• Two endpoints are in Receive Only mode for the main video endpoint with BFCP support.
• TRP is allocated during the call.
Note BFCP control channel is always unencrypted. However, the presentation channel is encrypted if both phones
are encrypted.
Presentation Sharing using BFCP Prerequisites

• Make sure all endpoints and trunks in the call flow are running SIP profile.
• Check Phone Support procedure and generate a report for the feature BFCP Support to obtain a list of
Cisco endpoints that support Presentation Sharing using BFCP. For these endpoints, BFCP support is
enabled by default. You need not perform any additional configuration for the phone to support BFCP.
For more information, see Generate a Phone Feature List, on page 1.
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Presentation Sharing using BFCP Configuration Task Flow
Presentation Sharing using BFCP Configuration Task Flow

Complete the following tasks to enable Presentation sharing using the Binary Floor Control Protocol (BFCP).
Procedure

Step 1 Enable BFCP Support for SIP Trunks, on page Enables BFCP support on all SIP trunks in the
813 call flow.
Step 2 Enable Presentation Sharing using BFCP for Enables BFCP support in the third-party phone
Third-Party Phones, on page 814 configuration, if you are using third-party SIP
endpoints.
Enable BFCP Support for SIP Trunks

If you are using Presentation sharing with BFCP, the feature must be enabled in the SIP Profile that is used
by all trunks in the messaging or call flow. The BFCP stream will be rejected by any trunk that does not
support presentation sharing.
Procedure
Step 1 Enable BFCP support within the SIP Profile that is used by the SIP trunk:
a) From the Cisco Unified CM Administration, choose Device > Device Settings > SIP Profile.
b) Perform either of the following steps:
• Click Find to select an existing SIP profile.
• Click Add New to create a new SIP profile.
c) In the SDP Information section, check the Allow Presentation Sharing using BFCP check box to
enable BFCP in the Unified Communications Manager.
By default, the check box is unchecked. For presentation sharing, BFCP must be enabled for all SIP trunks
between the Unified CM clusters.
d) Complete any other fields in the SIP Profile Configuration window. For more information on the fields
e) Click Save.
Step 2 Associate the BFCP enabled SIP Profile to your SIP trunks:
a) From the Cisco Unified CM Administration, choose Device > Trunks.
b) Click Find and select an existing SIP trunk.
c) In the SIP Information section, choose the SIP Profile for which you enabled BFCP to share the
presentation in the intercluster call from the SIP Profile drop-down list.
d) Click Save.
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Enable Presentation Sharing using BFCP for Third-Party Phones
e) Repeat this step for all SIP trunks that will be in the call flow of a BFCP session.
Enable Presentation Sharing using BFCP for Third-Party Phones

If you want to use Presentation sharing using BFCP with third-party SIP Phones, you must make sure that
the feature is enabled for the endpoint. This feature is supported by the following third-party phone types:
• Third-party SIP Device (Advanced)
• Third-party AS-SIP Endpoint
Procedure

Step 2 Click Find and select an existing third-party SIP phone.
Step 3 Check the Allow Presentation Sharing using BFCP check box.
Step 4 Click Save.
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CHAPTER 70
Video Telephony
• Video Telephony Overview, on page 815
• Video Telephony Support, on page 815
• Video Network, on page 818
• Video Telephony Configuration Task Flow, on page 820
• H.323 Video, on page 820
• Video Support, on page 825
• Video Features , on page 828
• QoS for Video Networks , on page 830
Video Telephony Overview

Unified Communications Manager supports video telephony and thus unifies the world of voice and video
calls. Video endpoints use Unified CM call-handling features and access a unified voice and video solution
for dialing and connecting video calls.
The Unified Communications Manager video telephony solution offers these features:
• Supports video and video-related features, such as Far-End Camera Control (FECC)
• Supports multiple logical channels that are needed to allow the transmission of video streams
• Transmits midcall, media-related messages that are needed for video (that is, transmits commands or
indications that are needed for video calls)
• Supports H.323, Skinny Client Control Protocol (SCCP), and Session Initiation Protocol (SIP)
• Enhances locations and regions to provide Bandwidth Management
• Provides serviceability information, such as Call Detail Records (CDRs), about video calls
Video Telephony Support

The following sections discuss the details of video telephony in the Unified Communications Manager
environment.
815
Video Calls
Video Calls
The typical video call includes two or three Real-Time Protocol (RTP) streams in each direction (that is, four
or six streams). The call can include the following stream types:
• Video (H.261, H.263, H.263+, H.264-SVC, X-H.264UC, H.264-AVC, H.265, AV1 and VT Camera
wideband video codecs)
• Far-End Camera Control (FECC) - Optional
• Binary Floor Control Protocols (BFCP)
Note Call control for video calls operates the same way as the call control that governs all other calls. For more
information, see the Configure Media Resources chapter in the System Configuration Guide. You can also
see, Configure Conference Bridges chapter in the System Configuration Guide for details on how Unified
Communications Manager can allocate a video conference bridge automatically.
Real-Time Transport Control Protocol Pass-Through in MTP Topologies

An IOS Media Terminate Point (MTP) before 15.2(2)T, cannot pass-through Real-Time Transport Control
Protocol (RTCP) packets and therefore cannot exchange Real-Time Protocol (RTP) feedback data to enhance
the RTP transmission. The primary function of RTCP is to provide feedback on media distribution by
periodically sending statistics to participants in a streaming multimedia session. RTCP gathers statistics for
a media connection and information such as transmitted octet and packet counts, lost packet counts, jitter, and
round-trip delay time. An application may use this information to control the quality of service parameters,
perhaps by limiting flow or using a different codec.
IOS MTP Version 15.2(2)T and later supports RTCP pass-through capability so that the endpoint in a call
with an MTP present can still provide feedback and status on an RTP transmission. RTCP pass-through
capability applies to media channels.
The RTCP pass-through feature is not limited to a specific call signaling protocol. For example, it can be
SIP-SIP, SIP-nonSIP, or nonSIP-nonSIP.
For Unified CM to allocate an RTCP pass-through capable MTP specifically, the call needs to fulfill the
following conditions:
• An MTP is requested for a feature that requires the MTP to be in media pass-through mode. For example,
TRP, DTMF translation, IP address V4/V6 translation, and so on. RTCP pass-through is only applicable
when the media is in pass-through mode.
• The RTCP pass-through MTP needs to be included in the Media Resource Group Lists (MRGL) of the
endpoint that sponsors the MTP. MTP can be inserted by RSVP, TRP, DTMF mismatch reasons.
• When the call is capable of establishing video channels, Unified CM attempts to search for an RTCP
pass-through capable MTP. For example, Unified CM picks an RTCP pass-through capable MTP from
other non-capable ones in the MRGL. If an RTCP pass-through capable MTP is not available, then
Unified CM stills allocate an MTP for the call.
• When the call is capable of establishing an audio channel only, Unified CM does not intentionally request
an RTCPpass-through capable MTP for the non-video calls. However, if the MRGL only contains
RTCPpass-through capable MTP(s), then Unified CM inserts one of those into the audio call.
816
Video Codecs
• The call also needs to fulfill the current CAC bandwidth for video calls to have an RTCP pass-through
capable MTP.
Note If a call initially establishes with a non-RTCP pass-through capable MTP (before version 15.2(2)T) present
in the call, and the call escalates into a video-capable call, Unified CM does not reallocate to an RTCP
pass-through capable MTP. In that case, even though the call has been escalated to a video call, the existing
MTP does not allow RTCP packets to be passed through.
Video Codecs
Common video codecs include H.261, an older video codec, H.263, a newer codec that gets used to provide
internet protocol (IP) video, and H.264, a high-quality codec. The system supports H.264 for calls that use
the Skinny Client Control Protocol (SCCP), H.323, and SIP on originating and terminating endpoints only.
The system also supports regions and locations.
Unified Communications Manager maintains the offerer's video codec ordering preference when making the
answer, if possible. H.265 is the preferred video codec is available on the endpoints, otherwise, Unified
Communications Manager follows the following codec preference preference order:
Preference Order Codecs Description
1 H.265 (HEVC) Provides higher quality video using lower bandwidth.
2 H.264 (SVC) Allows rendering of variable quality video from the same media
stream, by disregarding a subset of the packets received.
Note H. 264 SVC is a new annex to the H.264-AVC
video compression standard; meaning it is an
enhancement on top of H.264-AVC. It provides the
ability to encapsulate multiple video streams at
various frame-rates and resolutions in one container.
3 X-H.264UC (Lync) Microsoft-Proprietary Variant
4 H.264 (AVC) Advanced Video Coding
817
Video Network
Preference Order Codecs Description
5 H.263 H.263 and H.261 codecs exhibit the following parameters and
typical values:
6 H.261
• Bit rates range from 64 kb/s to a few mb/s. These bit rates
can exist in any multiple of 100 b/s. H.261 and H.263 can
function with bit rates lower than 64 kb/s, but video quality
suffers in such cases.
• One-quarter Common Interchange Format (QCIF)
(Resolution equals 176x144.)
• Common Interchange Format (CIF) (Resolution equals
352x288.)
• 4CIF (Resolution equals 704x576.)
• Sub QCIF (SQCIF) (Resolution equals 128x96.)
• 16CIF (Resolution equals 1408x1152.)
• Custom Picture Format
• Resolution:
• Frame Rate: 15 frames per second (fps), 30 fps
• Annexes: F, D, I, J, K, L, P, T, N
The bandwidth of video calls equals the sum of the audio bandwidth and the video bandwidth. The total
bandwidth does not include overhead.
A 384-kb/s video call may comprise G.711 at 64 kb/s (for audio) plus 320 kb/s (for video). This sum
does not include overhead. If the audio codec for a video call is G.729 (at 24 kb/s), the video rate
increases to maintain a total bandwidth of 384 kb/s. If the call involves an H.323 endpoint, the H.323
endpoint may use less than the total available video bandwidth. Regardless of protocol, the endpoint
may always choose to send at less than the max bit rate for the call.
Video Network
The following illustration provides an example of a video network that uses a single Unified Communications
Manager cluster. In a successful video network, any endpoint can call any other endpoint. Video availability
only exists if both endpoints are video-enabled. Video capabilities extend across trunks.
818
Video Network
The Cisco video conference portfolio comprises the following video bridges:
• Cisco TelePresence MCU series
• Webex Meeting Server
The Cisco UC Endpoints portfolio comprises the following endpoints that support video:
For more information on Cisco UC Endpoint portfolio that support vedio, see Compatibility Matrix.
Note Third-party SIP video endpoints are capable of connecting to Cisco Unified Communications Manager as a
line-side device or as a trunk-side device. For more information, see Third-Party SIP Endpoints.
819
Video Telephony Configuration Task Flow
Video Telephony Configuration Task Flow

To configure video telephony in Cisco Unified Communications Manager Administration follow these steps.
Procedure
Step 1 Configure regions for video call bandwidth if you use regions for call admission control.
Note All devices include a default region, which defaults to 384 kb/s for video. You can set the
bandwidth setting in Region configuration high enough for the desired resolution (For example,
increase to 2Mb/s for high-definition video call).
Step 2 Configure locations for video call bandwidth if you use locations for call admission control.
Step 3 (Optional) Configure the RSVP service parameters, or set the RSVP policy in the Location Configuration
window if using RSVP for bandwidth management of SIP video calls.
Step 4 Configure the appropriate conference bridge for your network to use a Cisco video conference bridge.
Step 5 Configure the media resource groups and media resource group lists for the user accordingly to configure a
user to use the video conference bridge instead of using other conference bridges.
Step 6 Configure the H.323 gateways in your system to retry video calls as audio calls (default behavior) or configure
AAR groups and route/hunt lists to use alternate routing for video calls that do not connect.
Step 7 Configure the H.323 phones in your system to retry video calls as audio calls (default behavior) or configure
AAR groups and route/hunt lists to use alternate routing for video calls that do not connect. Choose Enabled
for Video Capabilities.
Step 8 Configure the H.323 trunks in your system to retry video calls as audio calls (default behavior) or configure
AAR groups and route/hunt lists to use alternate routing for video calls that do not connect.
Step 9 Configure the Cisco Unified IP Phones that will support video.
Step 10 Configure the third-party SIP endpoints that will support video.
Step 11 Configure the SIP trunks in your system to retry video calls as audio calls (default behavior).
H.323 Video
H.323 video exhibits the following characteristics:
• H.323 endpoints can be configured as H.323 phones, H.323 gateways, or H.323 trunks.
• Call forwarding, dial plan, and other call-routing-related features work with H.323 endpoints.
• H.323 video endpoints cannot initiate hold, resume, transfer, park, and other similar features.
• If an H.323 endpoint supports the empty capability set (ECS), the endpoint can be held, parked, and so
forth.
• Some vendors implement call setup in such a way that they cannot increase the bandwidth of a call when
the call gets transferred or redirected. In such cases, if the initial call is audio, users may not receive
video when they are transferred to a video endpoint.
820
H.239-Extended Video Channels in H.323 Call
• No video media termination point (MTP) nor video transcoder currently exists. If an audio transcoder
or MTP is inserted into a call, that call will be audio only. This is true when the IPVC audio transcoding
capabilities is not being used. When the IPVC transcoders are used, you can transcode the audio and
send/receive video.
• For H.323 video calls, users must specify video call bandwidth.
H.239-Extended Video Channels in H.323 Call

The extended video channels feature works via H.239 protocol and enables multiple video channel support.
Cisco Unified Communications Manager supports negotiating an extended video channel using the H.239
protocol in direct point-to-point H.323 calls. This also includes calls across the H.323 intercluster trunk.
Cisco Unified Communications Manager supports all H.239 associated support signals and commands that
are specified in the H.239 recommendation.
The following sections describes characteristics which apply to the extended video channels feature.
Support for Third-Party H.323 Devices

The extended video channel feature supports H.239 interoperability among third-party video endpoints and
Cisco Unified Voice Conferencing. Cisco Unified Communications Manager allows an extended video channel
to be used for presentation and live meeting transmission. This feature focuses on multiple video channel
support via H.245 signaling. The following presentation applications provide basis for this multichannel
support:
• Natural Presenter package by the third-party vendor video endpoint
• People + Content by the third-party vendor Polycom
Both Natural Presenter package and People + Content use the H.239 protocol to negotiate capabilities and
define the roles of the additional video channels.
Note Natural Presenter package by video endpoint and People + Content by Polycom only support H.239 for the
presentation mode.
Be aware that the presentation applications that are offered by video endpoint and Polycom are optional
features. You must have one of these options and H.239 enabled in both caller and callee endpoints to negotiate
second video channels or the call will be limited to a single video channel.
H.323 Devices Invoke Presentation Feature

Cisco and Polycom video endpoints allow the user to share presentation materials from various components
(for example, VCR, Projector, PC, and so on). The components can physically connect with the endpoints,
and the PC can also run presentation applications that are provided by the vendor to transmit the presentation
image. The source of presentation and the component connection with the video endpoint are irrelevant to the
mechanism of establishing video channels by using H.239.
Note For details on setting up presentation sources, see the video endpoint user guide.
821
Opening Second Video Channels
When two H.239-enabled endpoints attempt to establish a video call, they declare their video capabilities for
the main video channel for meeting participants and their extended video capabilities (H.239 capabilities) for
the second video channel. The following contents comprise H.239 capability signals:
1. The endpoints send signals to indicate that the devices support H.239. They also send associated commands
and indication signals for managing the second video channel. This enables both the endpoints to be aware
that the call is capable of opening multiple video channels.
2. The endpoints sends out one or more extended video codec capabilities to express video codec capabilities
for second channels. The endpoint must specify the role of the second video channel. The defined role
labels can be
• Live-video-This channel gets processed normally and is suitable for live video of people
• Presentation-This channel relays a token-managed presentation that is distributed to the devices
After the capabilities have been exchanged, both endpoints immediately open two-way audio channels and
the main video channels as in the traditional video calls.
Opening Second Video Channels

Depending on the third-party endpoint implementation, the second video channel is handled differently among
vendors.
Natural Presenter Package by Tandberg

Video endpoints initiates the second video channel on demand. A video endpoint device does not open the
second video channel immediately after the main video channel is established. The second channel gets opened
when one of the callers (the presenter) specifies the source of the presentation and invokes a command to start
the presentation.
When a video endpoint user decides to start sharing the presentation, video endpoint requests the other call
party to open an extended video channel for receiving the presentation image; therefore, a video endpoint-video
endpoint call has only one-way second video channel.
People + Content by Polycom

Unlike video endpoint, a Polycom video endpoint initiates the second video endpoint immediately as a part
of the default mechanism, after both parties have confirmed that additional video channels can be supported.
822
Call Admission Control (CAC) on Second Video Channels
Note The channel established gets automatically if both parties support H.239 and have the extended video channel
feature enabled. However, the additional channel does not show anything until one of the parties start to share
presentation.
Polycom initiates a request for the second video channel to the other call party regardless of the usage of the
second video channel; therefore, in a Polycom-Polycom call, two-way video channels get opened between
the devices even if only one of them sends out presentation image/video.
This implementation ensures that both call parties have the second video channel ready for transmission when
the call parties decide to take the token to present something. Although one of the two video channels remains
idle (not sending anything), the Polycom device controls bandwidth to ensure load efficiency.
This difference in handling second video channels does not affect the implementation of H.239. Unified
Communications Manager does not initiate any receiving channel request in an H.323-H.323 call. Unified
Communications Manager simply relays all channel requests from one terminal to another.
Unified Communications Manager does not enforce two-way transmission for the second set of video channels
because this does not represent a requirement in the H.239 protocol.
Call Admission Control (CAC) on Second Video Channels

The following call admission control policies of Cisco Unified Communications Manager get applied to the
second video channels:
Cisco Unified Communications Manager restricts the bandwidth usage by the second video channels on the
basis of location configuration. When the second video channel is being established, Cisco Unified
Communications Manager makes sure that enough video bandwidth stays available within the location pool
and reserves bandwidth accordingly. If the required bandwidth is not available, Cisco Unified Communications
Manager instructs the channel to reduce the available bandwidth to zero.
No change occurs in the region configuration or policies to support the second video channels.
Traditionally, Cisco Unified Communications Manager region policy has only supported a call with a single
video channel and the total bandwidth usage of this call never gets larger than what the region configuration
specifies.
If the administrator sets a finite region video bandwidth restriction for an H.239 call, Cisco Unified
Communications Manager will violate the region policy because the region value will get used against the
bandwidth that is requested for each video channel independently.
Example
If the region video bandwidth is set to 384Kbps and the audio channel uses 64Kb/s,
the maximum allowed bandwidth for each video channel will be (384Kb/s - 64Kb/s)=
320Kb/s.
i.e. the maximum bandwidth to be used by the H.239 call will be (audio bw + 2*(384
- audio
bw)) = 704Kb/s, which goes beyond the 384Kb/s bandwidth that the region specifies.
823
Number of Video Channels Allowed
Note You should consider relaxing both region and location bandwidth restrictions for H.239 calls, so the H.239
devices are allowed to readjust and balance load for both the video channels without Cisco Unified
Communications Manager intervention.
Number of Video Channels Allowed

Unified Communications Manager supports only a maximum of two video channels due to the following
reasons:
• Both Cisco and Polycom only support two video channels, one of which is for main video, and the other
is for presentation.
• H.239 only defines an Additional Media Channel (AMC) for H.320-based system to partition the traditional
H.320 video channel for the purpose of presentation.
H.239 Commands and Indication Messages

Command and Indication (C&I) messages get used for H.239 to manage tokens for the Presentation and Live
roles and to permit devices to request release of video flow control to enable the operation of additional media
channels. Cisco Unified Communications Manager supports all the C & I messages. Whenever Cisco Unified
Communications Manager receives C&I messages, it relays them to the call party accordingly.
Be aware that the flow control release request and response messages can be used to request that the far end
release flow control, so it allows an endpoint to send the indicated channel at the indicated bit rate.
Note Be aware that the call party may or may not honor the request as is indicated by flow control release response.
The Presentation role token messages allow an H.239 device to acquire the token for presentation. The other
call party may accept or reject the request. The presenter device sends out a token release message when it is
no longer needed.
Topology and Protocol Interoperability Limitation

Cisco Unified Communications Manager supports only H.239 in H.323 to H.323 calls. Cisco Unified
Communications Manager allows H.239 calls to be established across H.323 intercluster trunk or multiple
nodes. If an H.239-enabled device attempts to make a call with a non-H323 end, the H.239 capabilities will
get ignored and the call will get conducted like the traditional video calls that are supported by Cisco Unified
Cisco Unified Communications Manager does not support a second video channel when a media termination
point or transcoder is inserted into the call. If it happens, the call will fall back to normal video calls.
Midcall Feature Limitation

Cisco Unified Communications Manager supports opening second video channels only in direct H.323 to
H.323 calls.
824
Video Support
Caution Do not attempt to invoke any midcall features such as call transfer or hold/resume operations. Doing so can
lead to problems and the second video channel can get disconnected.
Video Support
Unified Communications Manager supports video over H.323, SCCP and SIP protocols.
Skinny Client Control Protocol Video

Skinny Client Control Protocol video exhibits the following characteristics:
• If a phone that is running Skinny Client Control Protocol reports video capabilities, Cisco Unified
Communications Manager automatically opens a video channel if the other end supports video.
• For Skinny Client Control Protocol video calls, system administration determines video call bandwidth
by using regions. The system does not ask users for bit rate.
SIP Video
SIP video supports the following video calls by using the SIP Signaling Interface (SSI):
• SIP to SIP
• SIP to H.323
• SIP to SCCP
• SIP intercluster trunk
• H.323 trunk
• Combination of SIP and H.323 trunk
SIP video calls also provide media control functions for video conferencing.
Unified Communications Manager video supports SIP on both SIP trunks and lines support video signaling.
SIP supports the H.261, H.263, H.263+, H.264 (AVC), H.264 (SVC), X-H.264UC (Lync), and H.265 video
codecs (it does not support the wideband video codec that the VTA uses).
Configuring SIP Devices for Video Calls

Perform the following steps to enable video calls on SIP devices:
SIP Trunks
• On the Trunk Configuration window in Unified Communications Manager, check the Retry Video Call
as Audio check box if you want the call to use audio when the video connection is not available.
• Reset the trunk.
825
Cisco Video Conference Bridges
Third-Party SIP Endpoints

• On the Phone Configuration window in Cisco Unified Communications Manager Administration, check
the Retry Video Call as Audio check box if you want the call to use audio when the video connection
is not available.
• Reset the endpoint.
Cisco Video Conference Bridges

Unified Communications Manager supports a variety of solutions for video conferencing. The following video
conference bridges support ad hoc and meet-me video conferencing:
• Cisco TelePresence MCU
• Cisco TelePresence Conductor
• Cisco Meeting Server
Cisco TelePresence MCU Video Conference Bridge

Cisco TelePresence MCU is a set of hardware conference bridges for Cisco Unified Communications Manager.
The Cisco TelePresence MCU is a high-definition (HD) multipoint video conferencing bridge. It delivers up
to 1080p at 30 frames per second, full continuous presence for all conferences, full transcoding, and is ideal
for mixed HD endpoint environments. The Cisco TelePresence MCU supports SIP as the signaling call control
protocol. It has a built in Web Server that allows for complete configuration, control, and monitoring of the
system and conferences. The Cisco TelePresence MCU provides XML management API over HTTP.
Cisco TelePresence MCU allows both ad hoc and meet-me voice and video conferencing. Each conference
bridge can host several simultaneous, multiparty conferences. Cisco TelePresence MCU must be configured
in Port Reservation mode.
Cisco TelePresence Conductor Video Conference Bridge

Cisco TelePresence Conductor provides intelligent conference administrative controls and is scalable, supporting
device clustering for load balancing across MCUs and multiple device availability. Administrators can
implement the Cisco TelePresence Conductor as either an appliance or a virtualized application on VMware
with support for Cisco Unified Computing System (Cisco UCS) platforms or third-party-based platforms.
Multiway conferencing, that allows for dynamic two-way to three-way conferencing, is also supported.
Cisco TelePresence Conductor supports both ad hoc and meet-me voice and video conferencing. Cisco
TelePresence Conductor dynamically selects the most appropriate Cisco TelePresence resource for each new
conference. Ad hoc, “MeetMe” and scheduled voice and video conferences can dynamically grow and exceed
the capacity of individual MCUs. One Cisco TelePresence Conductor appliance or Cisco TelePresence
Conductor cluster has a system capacity of 30 MCUs or 2400 MCU ports. Up to three Cisco TelePresence
Conductor appliances or virtualized applications may be clustered to provide greater resilience.
Cisco Meeting Server

The Cisco Meeting Server conference bridge solution allows Ad Hoc, Meet-Me, Conference Now, and
Rendezvous conferences. This conference bridge offers premises-based audio, video, and web conferencing,
and works with third-party on-premises infrastructure. It scales for small or large deployments. You can add
capacity incrementally as needed, to ensure that you can support the current and future needs of your
826
Video Encryption
organization. This conference bridge provides advanced interoperability. Any number of participants can
create and join meetings from:
• Cisco or third-party room or desktop video systems
• Cisco Jabber Client
• Cisco Meeting App (can be native or with a WebRTC compatible browser)
• Skype for Business
A minimum release of Cisco Meeting Server 2.0 is required to use the Cisco Meeting Server conference
bridge.
The Cisco Meeting Server supports SIP as the signaling call control protocol. It has a built in Web Server that
allows for complete configuration, control, and monitoring of the system and conferences. The Cisco Meeting
Server provides XML management API over HTTP.
Note Cisco Meeting Server does not support H.265 video codec and Far End Camera Control (FECC).
Video Encryption
Unified Communications Manager supports encryption of audio, video, and other media streams so long as
the individual endpoints involved in the communication also support encryption. Unified CM uses the Secure
Real-Time Transport Protocol (SRTP) to encrypt the media streams. Some of the features include:
• Support for SIP and H.323 endpoints
• Support for encryption of main audio and video line while operating in Media Termination Point (MTP)
passthru mode
• Support for multiple encryption methods
• Support for Session Description Protocol (SDP) crypto-suite session parameters in accordance with RFC
4568
To provide encrypted communications, encryption keys are exchanged between the endpoints and Unified
Communications Manager during the SIP call setup. For this reason, the SIP signaling should be encrypted
using TLS. During the initial call setup, the video endpoints exchange a list of encryption methods they
support, select an encryption suite supported by both endpoints, and exchange encryption keys. If the endpoints
cannot agree on a common encryption suite, the media streams are unencrypted and transported using the
Real-Time Transport Protocol (RTP).
Note If the individual endpoints do not support encryption, the communication will take place using RTP.
Configure Interop with VCS

Perform the following steps on the SIP trunk that connects Unified Communications Manager to Cisco VCS
to enable Unified CM to interoperate with a Cisco VCS.
827
Video Features
Procedure

Step 2 Perform one of the following:
• Click Add New to configure a new trunk.
Step 3 In the Trunk Configuration window, choose the Trunk Type, Device Protocol, Trunk Service Type that
connects Unified Communications Manager to the Cisco VCS and click Next.
Step 4 In the SIP Profile drop-down list, choose Standard SIP Profile for VCS.
Step 5 In the Normalization Script drop-down list, choose vcs-interop.
Step 6 In the Normalization Script area, leave the Parameter Name and Parameter Value fields empty. If these
fields are populated with values, delete the contents of the field.
Step 7 Click Save.
Video Features
The following video-related features are supported in SIP video networks:
• Binary Floor Control Protocol (BFCP)
• Encrypted iX Channel
• Far End Camera Control (FECC)
Endpoint Support for the Binary Floor Control Protocol

Unified Communications Manager provides support for the Binary Floor Control Protocol (BFCP) for specific
Cisco and third-party video endpoints. BFCP allows users to share a presentation within an ongoing video
conversation.
For more information, see chapter Configure Presentation Sharing using BFCP in Feature Configuration Guide
Encrypted iX Channel
Unified Communications Manager supports an encrypted iX channel. The iX channel provides a reliable
channel for multiplexing application media between SIP phones in a video conference. Encrypted iX Channel
uses DTLS to add security to your deployment and ensures that the application media is sent over the iX
Channel is private and cannot be viewed by intermediate parties who attempt to intercept media.
IOS MTP and RSVP agents in pass through mode also support encrypted iX Channel.
Configuration
To enable an encrypted iX Channel on Unified Communications Manager, you must:
828
Encryption Modes
• Check the Allow iX Application Media check box in the SIP Profile Configuration that is used by any
intermediate SIP trunks. This setting turns on the iX channel negotiation.
• Configure the Secure Call Icon Display Policy service parameter to enable a secure lock icon. By
default, the setting is All media except BFCP and iX transports must be encrypted.
Encryption Modes
There are two types of Session Description Protocol (SDP) offers thatUnified Communications Manager
supports for iX Channel encryption for encrypted phones. This encryption type is driven by what the endpoints
support and is not a configurable item in the Unified Communications Manager.
• Best Effort Encryption—The SDP offer is for an encrypted iX Channel, but falls back to a non-encrypted
iX Channel if the SIP peers do not support it. This approach can be used if encryption is not mandatory
in the solution.
For example, encryption is mandatory within the cloud, and not in a single enterprise.
Best-Effort iX Encryption
m=application 12345 UDP/UDT/IX *
a=setup:actpass
a=fingerprint: SHA-1 <key>
• Forced Encryption—The SDP offer is for an encrypted iX Channel only. This offer is rejected if the
SIP peers do not support iX Channel encryption. This approach can be used in deployments where
encryption is mandatory between endpoints.
For example, encryption is mandatory between the two SIP devices.
Forced iX Encryption
m=application 12345 UDP/DTLS/UDT/IX *
a=setup:actpass
a=fingerprint: SHA-1 <key>
By default, all Cisco IP Phones are set to offer Best Effort iX Encryption. However, you can reset this to
Forced Encryption by setting the Encryption Mode to On within the Product-Specific Configuration of Cisco
TelePresence endpoints, or by reconfiguring settings on the Cisco Meeting Server.
Non-Encrypted Modes
Unified Communications Managerenables negotiation of secure active control messages in media path from
endpoints in a meeting when the endpoint may not be deployed in a fully secure mode. For example, if the
endpoint is Off-Net and is registered with Unified CM in Mobile and Remote Access mode.
Prerequisite
Before you start using this feature, make sure that:
• System adheres to the export compliance requirement
• SIP trunk to the conference bridge is secure
829
Far End Camera Control Protocol Support
Unified CM can negotiate the DTLS information in secure active control messages for non-secure endpoints
or softphones and receive messages in the following ways:
• Best Effort Encryption iX to On-Premise registered endpoints or softphones
• Forced iX Encryption to Off-Premise registered endpoints or softphones
Far End Camera Control Protocol Support

The Far End Camera Control (FECC) protocol allows you to control a remote camera. Within video calls,
FECC allows the party at one end of the call to control the camera at the far end. This control can include
panning the camera from one side to the other, tilting the camera, or zooming in and out. For video conferences
that use multiple cameras, FECC can be used to switch from one camera to another.
Unified Communications Manager supports the FECC protocol for video endpoints that are FECC-capable.
Cisco Unified Communications Manager supports FECC for SIP-SIP calls or H.323-H.323 calls, but does
not support FECC for SIP-H.323 calls. To support FECC, Unified Communications Manager sets the application
media channel through SIP or H.323 signaling. After the media channel is established, the individual endpoints
can communicate the FECC signaling.
QoS for Video Networks

Cisco Unified Communications Manager contains a number of administrative tools for managing Quality of
Service (QoS) for video networks:
• Bandwidth Management—Mange bandwidth allocations for specific Regions and Locations
• Enhanced Locations Call Admission Control
• Session Level Bandwidth Modifiers
• Flexible DSCP Markings
• Alternate Routing
Bandwidth Management
Bandwidth allocations for audio and video calls are managed through regions and locations that you configure
in Cisco Unified Communications Manager Administration.
The amount of bandwidth available for a specific call must be able to manage the combination of all media
streams that are associated with the session, including voice, video, signaling, and any extra media, such as
a BFCP presentation. Cisco Unified Communications Manager contains features able to manage bandwidth.
Enhanced Locations Call Admission Control

Enhanced Locations Call Admission Control (CAC) enables you to control the audio and video quality of
calls over a wide-area (IP WAN) link by limiting the number of calls that are allowed on that link at the same
time. For example, you can use call admission control to regulate the voice quality on a 56-kb/s frame relay
line that connects your main campus and a remote site.
830
Session Level Bandwidth Modifiers
CAC verifies whether there is sufficient bandwidth available to complete a call. CAC can reject calls due to
insufficient bandwidth.
In Unified Communications Manager, locations-based call admission control works in conjunction with
regions to define the characteristics of a network link. Regions and locations work in the following manner:
• Regions allow the bandwidth of video calls to be set. The audio limit on the region may result in filtering
out codecs with higher bit rates. However, for video calls, the video limit constrains the quality (resolution
and transmission rate) of the video.
• Locations define the amount of total bandwidth available for all calls on that link. When a call is made
on a link the regional value for that call must be subtracted from the total bandwidth allowed for that
link.
For more information about call admission control, see chapter 'Configure Enhanced Locations Call Admission
Control' in System Configuration Guide for Cisco Unified Communications Manager
.
Session Level Bandwidth Modifiers

Unified Communications Manager provides location call admission control support for handling session level
bandwidth modifiers. Session level bandwidth modifiers are communicated as part of the parameters in the
SDP portion of the initial SIP signaling. These parameters indicate the maximum amount of bandwidth each
endpoint will support for that type of call. These parameters are used, along with regions and locations settings,
to set the bandwidth for each call.
During the initial call setup, both parties communicate to Unified Communications Manager their maximum
allowed bandwidth for the call. Unified Communications Manager passes this communication to the other
endpoint, but if the bandwidth that is specified by the endpoint is greater than the region setting, Unified
Communications Manager replaces the value with the region bandwidth value.
Unified Communications Manager uses the following rules to determine the amount of bandwidth to allocate
to a specific call:
• When Unified Communications Manager receives an Offer or Answer from an endpoint, it checks whether
there is a session level bandwidth modifier in the SDP:
• If there is a session level bandwidth modifier, Unified Communications Manager retrieves the
bandwidth value from the modifier. If there is more than one modifier type, it retrieves the modifier
in the following order of preference: Transport Independent Application Specific (TIAS), Application
Specific (AS), Conference Total (CT).
• If there is no session level bandwidth modifier, Unified Communications Manager retrieves the
bandwidth value from the sum of the media level bandwidth modifiers.
• The allocated bandwidth is the maximum of what the two endpoints support up to the maximum value
of the Region setting. The allocated bandwidth cannot exceed the region setting.
Unified Communications Manager uses the following logic when communicating with endpoints:
• When generating an Answer, Early Offer or Re-Invite Offer to an endpoint that contains more than one
session level bandwidth modifier type (TIAS, AS, CT), Unified Communications Manager uses the same
bandwidth value for each.
831
Video Resolution Support for SIP Phones
• When generating an answer, Unified Communications Manager uses the same session level bandwidth
modifier type (TIAS, CT, AS) that was received in the initial offer
• For backward compatibility, the older Unified Communications Manager suppresses the Session Level
Bandwidth Modifier when a video call is put on hold and music on hold (MOH) is inserted.
Video Resolution Support for SIP Phones

Cisco Unified Communications Manager supports the imageattr line in the SDP portion of the SIP header for
higher resolution video calls. Cisco SIP phones that support w360p (640 x 360), such as the 9951, 9971, and
8961, automatically select the best resolution for video calls depending on the following criteria:
• If the session level bandwidth is greater than 800Kb/s and the imageattr[640 x 480] line in the SDP exists,
then VGA is used.
• If the session level bandwidth is greater than 800Kb/s and the imageattr[640 x 480] line in the SDP does
not exist, then w360p is used.
• If the session level bandwidth is less than 800Kb/s but greater than 480 bits per second and the
imageattr[640 x 480] line exists, then VGA 15 frames per second is used.
Note If you currently have a Cisco IP Phone model 9951, 9971, or 8961 that supports w360p (640 x 360) video
resolution and are upgrading to Cisco Unified Communications Manager release 8.5(1) or later, you may
notice changes in the resolution of video calls. The w360p resolution was introduced at phone load 9.2(1).
The following video call flow is between two 9951 phones (Phone A and Phone B) without imageattr line
support (for example, using Cisco Unified Communications Manager releases 8.0(1) and earlier):
1. Phone A sends a SIP message with an imageattr line in the SDP.
2. Cisco Unified Communications Manager deletes the imageattr line in the SDP and then sends the modified
SIP message to Phone B.
3. Phone B attempts to send video with the w360p resolution because there is no imageattr line in the SDP
portion of the SIP header.
The following video call flow is between two 9951 phones (Phone A and Phone B) with imageattr line support
(for example, using Cisco Unified Communications Manager releases 8.5(1) and later):
1. Phone A sends a SIP message with the imageattr line in the SDP.
2. Cisco Unified Communications Manager does not delete the imageattr line and sends the SIP message to
Phone B unchanged.
3. Phone B attempts to send video with the VGA resolution.
Alternate Routing
If an endpoint cannot obtain the bandwidth that it needs for a video call, a video call retries as an audio call
for the default behavior. To use route/hunt lists or Automated Alternate Routing (AAR) groups to try different
832
Flexible DSCP Markings
paths for such video calls, uncheck the Retry Video Call as Audio setting in the configuration settings for
applicable gateways, trunks, and phones.
For more information, see Configure AAR Group section under Configure Call Routing chapter in the System
Flexible DSCP Markings

Differentiated Services Code Point (DSCP) packet marking, is used to specify the class of service for each
packet. DSCP marking lets you prioritize certain types of calls or media over other types. For example, you
can prioritize audio over video so that, even if you experience network bandwidth issues, audio calls should
not experience bandwidth issues.
You can customize DSCP markings in either of these ways:
• Configure cluster wide service parameters to set the default DSCP settings for the cluster
• (Optional) For a subset of the DSCP categories, you can assign customized DSCP settings to devices via
the SIP Profile. For the devices that use the profile, the customized settings override the service parameter
defaults.
For more information on how to configure DSCP markings, see 'Configure Flexible DSCP Marking and Video
Promotion' chapter in the Feature Configuration Guide for Cisco Unified Communications Manager.
Phone Configuration for Video Calls

The following setting for video-enabled devices affects video calls:
• Retry Video Call as Audio-By default, this check box remains checked. Thus, if an endpoint (phone,
gateway, trunk) cannot obtain the bandwidth that it needs for a video call, call control retries the call as
an audio call. This setting applies to the destination devices of video calls.
• Video Capabilities Enabled/disabled-This drop-down list box turns video capabilities on and off.
Conference Control for Video Conferencing

Unified Communications Manager supports the following conference controls capabilities:
• Roster/Attendee List
• Drop Participant
• Terminate Conference
• Show Conference Chairperson/Controller
• Continuous Presence
Unified Communications Manager also supports the following video conference capabilities for Skinny Client
Control Protocol (SCCP) phones:
• Display controls for video conferences. The SCCP phones can choose to use the continuous presence or
voice-activated mode to view the video conference. When a mode is chosen, a message gets sent to the
833
Video Telephony and Cisco Unified Serviceability
bridge to indicate which mode to use on the video channel. Switching between modes does not require
renegotiation of media.
• Display participant information such as the user name in the video stream. The system can use the
participant information for other conferencing features such as roster.
For more information, see 'Encrypted iX Channel' chapter in the Security Guide for Cisco Unified
Video Telephony and Cisco Unified Serviceability

Cisco Unified Serviceability tracks video calls and conferences by updating performance monitoring counters,
video bridge counters, and call detail records (CDRs).
Performance Counters
Video telephony events cause updates to the following Cisco Unified Serviceability performance monitoring
counters:
Cisco CallManager
• VCBConferenceActive
• VCBConferenceCompleted
• VCBConferenceTotal
• VCBOutOfConferences
• VCBOutOfResources
• VCBResourceActive
• VCBResourceAvailable
• VideoCallsActive
• VideoCallsCompleted
• VideoOutOfResources
Gatekeeper
CiscoH.323
Cisco Locations
• RSVP VideoCallsFailed
834
Video Bridge Counters
• RSVP VideoReservationErrorCounts
• VideoBandwidthAvailable
• VideoBandwidthMaximum
Cisco SIP
Cisco Vedio Conference Bridge

• ConferencesActive
• ConferencesAvailable
• ConferencesCompleted
• ConferencesTotal
• OutOfConferences
• OutOfResources
• ResourceActive
• ResourceAvailable
• ResourceTotal
Video Bridge Counters

Video conference events cause updates to these Cisco video conference bridge performance monitoring
counters:
• ConferencesActive
• ConferencesAvailable
• ConferencesCompleted
• ConferencesTotal
• OutOfConferences
• OutOfResources
• ResourceActive
• ResourceAvailable
• ResourceTotal
These counters also display in the Cisco Unified Communications Manager object with the VCB prefix.
835
Call Detail Records (CDRs)
Call Detail Records (CDRs)

Video telephony events cause updates to Call Detail Records (CDRs) in Cisco Unified Serviceability. These
CDRs include the following information:
• origVideoCap_Codec
• origVideoCap_Bandwidth
• origVideoCap_Resolution
• origVideoTransportAddress_IP
• origVideoTransportAddress_Port
• destVideoCap_Codec
• destVideoCap_Bandwidth
• destVideoCap_Resolution
• destVideoTransportAddress_IP
• destVideoTransportAddress_Port
• origRSVPStat
• destRSVPVideoStat
• origVideoCap_Codec_Channel2
• origVideoCap_Bandwidth_Channel2
• origVideoCap_Resolution_Channel2
• origVideoTransportAddress_IP_Channel2
• origVideoTransportAddress_Port_Channel2
• origVideoChannel_Role_Channel2
• destVideoCap_Codec_Channel2
• destVideoCap_Bandwidth_Channel2
• destVideoCap_Resolution_Channel2
• destVideoTransportAddress_IP_Channel2
• destVideoTransportAddress_Port_Channel2
• destVideoChannel_Role_Channel2
Call Management Records (CMRs)

Video telephony events cause updates to Call Management Records (CMRs) in Cisco Unified Serviceability.
These CMRs include the following information:
• videoContentType Text String
• videoDuration Integer
836
• numberVideoPacketsSent Integer
• numberVideoOctetsSent Integer
• numberVideoPacketsReceived Integer
• numberVideoOctetsReceived Integer
• numberVideoPacketsLost Integer
• videoAverageJitter Integer
• videoRoundTripTime
• videoOneWayDelay
• videoTransmissionMetrics
837
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Feature Configuration Guide for Cisco Unified Communications

Manager, Release 12.5(1)SU1

PART I Getting Started 43

CHAPTER 1 Feature Configuration Overview 1

About the Feature Configuration Guide 1

CHAPTER 2 Configuration Tools 3

About the Feature Configuration Guide 3

Cisco Unified Communications Manager Serviceability 4

PART II Remote Worker Features 7

CHAPTER 3 Cisco Unified Mobility 9

Cisco Unified Mobility Overview 9

Configure Softkey Template for Mobility 16

CHAPTER 4 Device Mobility 43

Device Mobility Overview 43

Device Pool Assignment 45

CHAPTER 5 Extend and Connect 55

CHAPTER 6 Remote Worker Emergency Calling 65

Remote Worker Emergency Calling Overview 65

Specify Alternate Routing for Emergency Calling 67

CHAPTER 7 Configure Mobile and Remote Access 69

Mobile and Remote Access Overview 69

PART III Remote Network Access 79

CHAPTER 8 Wireless LAN 81

Wireless LAN Overview 81

CHAPTER 9 VPN Client 85

VPN Client Overview 85

Complete Cisco IOS Prerequisites 87

Complete ASA Prerequisites for AnyConnect 89

CHAPTER 10 Licensing 101

PART V Monitoring and Recording 117

CHAPTER 11 Silent Monitoring 119

Silent Monitoring Overview 119

Assign Security Profile to Phone 124

CHAPTER 12 Recording 127

Recording Overview 127

Enable Built in Bridge for a Phone 134

Configure a Record Feature Button 136

PART VI Call Center Features 143

CHAPTER 13 Agent Greeting 145

Agent Greeting Overview 145

CHAPTER 14 Auto-Attendant 149

Auto-Attendant Overview 149

Cisco Unity Express Auto-Attendant Troubleshooting 156

CHAPTER 15 Manager Assistant 157

Cisco Unified Communications Manager Assistant Overview 157

Configure CTI Route Point 178

Configure Assistant Line Appearances for Shared Line 184

Calling Party Gets Reorder Tone 190

PART VII Voice Messaging Features 199

CHAPTER 16 Audible Message Waiting Indicator 201

Audible Message Waiting Indicator Overview 201

CHAPTER 17 Immediate Divert 205

Immediate Divert Overview 205

Immediate Divert Configuration Task Flow 206

PART VIII Conferencing Features 215

CHAPTER 18 Ad Hoc Conferencing 217

Ad Hoc Conferencing Overview 217

CHAPTER 19 Meet-Me Conferencing 229

Meet-Me Conferencing Overview 229