LTRT 65437 MP 11x and MP 124 Sip Users Manual Ver 66

User's Manual
AudioCodes Analog Media Gateways
MediaPack™ Series
MP-11x & MP-124 Analog VoIP Gateways
Version 6.6
User's Manual Contents
Table of Contents
1 Overview ............................................................................................................ 19
1.1 MediaPack Models ................................................................................................. 20
1.2 SIP Overview ......................................................................................................... 20
Getting Started with Initial Connectivity ................................................................25

2 Assigning the OAMP IP Address ..................................................................... 27
2.1 Web Interface ......................................................................................................... 27
2.2 BootP/TFTP Server ................................................................................................ 29
2.3 CLI.......................................................................................................................... 30
2.4 FXS Voice Menu Guidance .................................................................................... 32
Management Tools ..................................................................................................35

3 Introduction ....................................................................................................... 37
4 Web-Based Management .................................................................................. 39
4.1 Getting Acquainted with the Web Interface ............................................................ 39
4.1.1 Computer Requirements..........................................................................................39
4.1.2 Accessing the Web Interface ...................................................................................40
4.1.3 Areas of the GUI ......................................................................................................41
4.1.4 Toolbar Description..................................................................................................42
4.1.5 Navigation Tree .......................................................................................................43
4.1.5.1 Displaying Navigation Tree in Basic and Full View ..................................44
4.1.5.2 Showing / Hiding the Navigation Pane .....................................................45
4.1.6 Working with Configuration Pages ..........................................................................45
4.1.6.1 Accessing Pages ......................................................................................45
4.1.6.2 Viewing Parameters .................................................................................46
4.1.6.3 Modifying and Saving Parameters ...........................................................47
4.1.6.4 Working with Tables .................................................................................48
4.1.7 Searching for Configuration Parameters .................................................................51
4.1.8 Working with Scenarios ...........................................................................................52
4.1.8.1 Creating a Scenario..................................................................................53
4.1.8.2 Accessing a Scenario ...............................................................................55
4.1.8.3 Editing a Scenario ....................................................................................56
4.1.8.4 Saving a Scenario to a PC .......................................................................57
4.1.8.5 Loading a Scenario to the Device ............................................................57
4.1.8.6 Deleting a Scenario ..................................................................................58
4.1.8.7 Quitting Scenario Mode ............................................................................59
4.1.9 Creating a Login Welcome Message.......................................................................60
4.1.10 Getting Help .............................................................................................................61
4.1.11 Logging Off the Web Interface .................................................................................62
4.2 Viewing the Home Page......................................................................................... 63
4.2.1 Assigning a Port Name ............................................................................................65
4.3 Configuring Web User Accounts ............................................................................ 66
4.3.1 Basic User Accounts Configuration .........................................................................67
4.3.2 Advanced User Accounts Configuration ..................................................................69
4.4 Displaying Login Information upon Login ............................................................... 73
4.5 Configuring Web Security Settings ........................................................................ 73
4.6 Web Login Authentication using Smart Cards ....................................................... 74
Version 6.6 3 MP-11x & MP-124

MP-11x & MP-124
4.7 Configuring Web and Telnet Access List ............................................................... 75

4.8 Configuring RADIUS Settings ................................................................................ 76
5 CLI-Based Management.................................................................................... 77
5.1 Enabling CLI using Telnet ...................................................................................... 77
5.2 Enabling CLI using SSH and RSA Public Key ....................................................... 78
5.3 Establishing a CLI Session .................................................................................... 80
5.4 Command Shell...................................................................................................... 81
5.4.1 Getting Familiar with the Command Shell ...............................................................81
5.4.1.1 Basic Command Shell Commands ..........................................................81
5.4.1.2 Accessing Subdirectories .........................................................................81
5.4.2 Status Commands ...................................................................................................82
5.5 Ping Command ...................................................................................................... 84
5.6 Management Commands ....................................................................................... 84
5.7 Configuration Commands ...................................................................................... 85
6 SNMP-Based Management ............................................................................... 87
6.1 Configuring SNMP Community Strings .................................................................. 87
6.2 Configuring SNMP Trap Destinations .................................................................... 89
6.3 Configuring SNMP Trusted Managers ................................................................... 90
6.4 Configuring SNMP V3 Users.................................................................................. 91
7 EMS-Based Management .................................................................................. 93
8 INI File-Based Management .............................................................................. 95
8.1 INI File Format ....................................................................................................... 95
8.1.1 Configuring Individual ini File Parameters ...............................................................95
8.1.2 Configuring Table ini File Parameters .....................................................................96
8.1.3 General ini File Formatting Rules ............................................................................97
8.2 Loading an ini File .................................................................................................. 98
8.3 Modifying an ini File ............................................................................................... 99
8.4 Secured Encoded ini File ....................................................................................... 99
8.5 Configuring Password Display in ini File ................................................................ 99
9 TR-069 Based Management ............................................................................ 101
9.1 TR-069 ................................................................................................................. 101
9.2 TR-104 ................................................................................................................. 106
9.3 Configuring TR-069 .............................................................................................. 107
General System Settings ......................................................................................109

10 Configuring Certificates ................................................................................. 111
10.1 Replacing the Device's Certificate ....................................................................... 111
10.2 Loading a Private Key .......................................................................................... 113
10.3 Mutual TLS Authentication ................................................................................... 114
10.4 Self-Signed Certificates........................................................................................ 115
10.5 TLS Server Certificate Expiry Check .................................................................... 115
10.6 Configuring Certificate Revocation Checking (OCSP) ......................................... 116
10.7 Loading Certificate Chain for Trusted Root .......................................................... 117
User's Manual 4 Document #: LTRT-65437

11 Date and Time .................................................................................................. 119

11.1 Configuring Date and Time Manually ................................................................... 119
11.2 Automatic Date and Time through SNTP Server ................................................. 119
General VoIP Configuration ..................................................................................121

12 Network ............................................................................................................ 123
12.1 Ethernet Interface Configuration .......................................................................... 123
12.2 Configuring IP Network Interfaces ....................................................................... 124
12.2.1 Assigning NTP Services to Application Types ......................................................129
12.2.2 Multiple Interface Table Configuration Rules.........................................................129
12.2.3 Troubleshooting the Multiple Interface Table ........................................................130
12.2.4 Networking Configuration Examples .....................................................................131
12.2.4.1 One VoIP Interface for All Applications ................................................. 131
12.2.4.2 VoIP Interface per Application Type...................................................... 132
12.2.4.3 VoIP Interfaces for Combined Application Types ................................. 133
12.2.4.4 VoIP Interfaces with Multiple Default Gateways ................................... 134
12.3 Configuring the IP Routing Table ......................................................................... 135
12.3.1 Interface Column ...................................................................................................137
12.3.2 Routing Table Configuration Summary and Guidelines ........................................137
12.3.3 Troubleshooting the Routing Table .......................................................................138
12.4 Configuring Quality of Service.............................................................................. 139
12.5 Disabling ICMP Redirect Messages..................................................................... 141
12.6 DNS...................................................................................................................... 142
12.6.1 Configuring the Internal DNS Table.......................................................................142
12.6.2 Configuring the Internal SRV Table .......................................................................143
12.7 Configuring NFS Settings..................................................................................... 145
12.8 Network Address Translation Support ................................................................. 147
12.8.1 Device Located behind NAT ..................................................................................147
12.8.1.1 Configuring STUN ................................................................................. 148
12.8.1.2 Configuring a Static NAT IP Address for All Interfaces ......................... 149
12.8.2 Remote UA behind NAT ........................................................................................149
12.8.2.1 First Incoming Packet Mechanism ........................................................ 150
12.8.2.2 No-Op Packets ...................................................................................... 150
12.9 Robust Receipt of Media Streams ....................................................................... 151
12.10 Multiple Routers Support...................................................................................... 151
12.11 IP Multicasting ...................................................................................................... 151
13 Security ............................................................................................................ 153
13.1 Configuring Firewall Settings ............................................................................... 153
13.2 Configuring 802.1x Settings ................................................................................. 157
13.3 Configuring General Security Settings ................................................................. 158
13.4 IPSec and Internet Key Exchange ....................................................................... 159
13.4.1 Enabling IPSec ......................................................................................................160
13.4.2 Configuring IP Security Proposal Table.................................................................160
13.4.3 Configuring IP Security Associations Table...........................................................161
14 Media ................................................................................................................ 165
14.1 Configuring Voice Settings ................................................................................... 165
14.1.1 Configuring Voice Gain (Volume) Control .............................................................165

MP-11x & MP-124
14.1.2 Echo Cancellation ..................................................................................................166

14.2 Fax and Modem Capabilities................................................................................ 168
14.2.1 Fax/Modem Transport Modes ...............................................................................169
14.2.1.1 T.38 Fax Relay Mode ............................................................................ 169
14.2.1.2 G.711 Fax / Modem Transport Mode .................................................... 170
14.2.1.3 Fax Fallback .......................................................................................... 171
14.2.1.4 Fax/Modem Bypass Mode .................................................................... 171
14.2.1.5 Fax / Modem NSE Mode ....................................................................... 173
14.2.1.6 Fax / Modem Transparent with Events Mode ....................................... 174
14.2.1.7 Fax / Modem Transparent Mode ........................................................... 174
14.2.1.8 RFC 2833 ANS Report upon Fax/Modem Detection ............................ 175
14.2.2 V.34 Fax Support ...................................................................................................175
14.2.2.1 Bypass Mechanism for V.34 Fax Transmission .................................... 176
14.2.2.2 Relay Mode for T.30 and V.34 Faxes ................................................... 176
14.2.3 V.152 Support ........................................................................................................177
14.2.4 Fax Transmission behind NAT ..............................................................................178
14.3 Configuring RTP/RTCP Settings .......................................................................... 179
14.3.1 Configuring the Dynamic Jitter Buffer ....................................................................179
14.3.2 Comfort Noise Generation .....................................................................................180
14.3.3 Dual-Tone Multi-Frequency Signaling ...................................................................181
14.3.3.1 Configuring DTMF Transport Types...................................................... 181
14.3.3.2 Configuring RFC 2833 Payload ............................................................ 182
14.3.4 Configuring RTP Base UDP Port ...........................................................................183
14.4 Configuring Analog Settings................................................................................. 184
14.5 Configuring DSP Templates................................................................................. 185
14.6 Configuring Media Security .................................................................................. 186
14.7 Configuring Media Realms ................................................................................... 188
14.7.1 Configuring Quality of Experience per Media Realm ............................................190
14.8 Quality of Experience ........................................................................................... 193
14.8.1 Reporting Voice Quality of Experience to SEM .....................................................193
14.8.1.1 Configuring the SEM Server ................................................................. 193
14.8.1.2 Configuring Clock Synchronization between Device and SEM ............ 194
14.8.1.3 Enabling RTCP XR Reporting to SEM .................................................. 194
15 Services ........................................................................................................... 195
15.1 Least Cost Routing............................................................................................... 195
15.1.1 Overview ................................................................................................................195
15.1.2 Configuring LCR ....................................................................................................197
15.1.2.1 Enabling the LCR Feature ..................................................................... 197
15.1.2.2 Configuring Cost Groups ....................................................................... 199
15.1.2.3 Configuring Time Bands for Cost Groups ............................................. 200
15.1.2.4 Assigning Cost Groups to Routing Rules .............................................. 201
16 Enabling Applications..................................................................................... 203
17 Control Network .............................................................................................. 205
17.1 Configuring IP Groups.......................................................................................... 205
17.2 Configuring Proxy Sets Table .............................................................................. 208
18 SIP Definitions ................................................................................................. 213
18.1 Configuring SIP Parameters ................................................................................ 213
18.2 Configuring Account Table ................................................................................... 213
18.3 Configuring Proxy and Registration Parameters .................................................. 216
18.3.1 SIP Message Authentication Example ..................................................................217

19 Coders and Profiles ........................................................................................ 219

19.1 Configuring Coders .............................................................................................. 219
19.2 Configuring Coder Groups ................................................................................... 222
19.3 Configuring Tel Profile.......................................................................................... 223
19.4 Configuring IP Profiles ......................................................................................... 225
Gateway Application .............................................................................................231

20 Introduction ..................................................................................................... 233
21 Hunt Group ...................................................................................................... 235
21.1 Configuring Endpoint Phone Numbers................................................................. 235
21.2 Configuring Hunt Group Settings ......................................................................... 237
22 Manipulation .................................................................................................... 241
22.1 Configuring General Settings ............................................................................... 241
22.2 Configuring Source/Destination Number Manipulation Rules .............................. 241
22.3 Manipulating Number Prefix ................................................................................. 246
22.4 SIP Calling Name Manipulations.......................................................................... 248
22.5 Configuring Redirect Number IP to Tel ................................................................ 251
22.6 Mapping NPI/TON to SIP Phone-Context ............................................................ 253
23 Routing............................................................................................................. 255
23.1 Configuring General Routing Parameters ............................................................ 255
23.2 Configuring Tel to IP Routing ............................................................................... 256
23.3 Configuring IP to Hunt Group Routing Table ....................................................... 263
23.4 IP Destinations Connectivity Feature ................................................................... 266
23.5 Alternative Routing for Tel-to-IP Calls .................................................................. 268
23.5.1 Alternative Routing Based on IP Connectivity .......................................................268
23.5.2 Alternative Routing Based on SIP Responses ......................................................269
23.6 Alternative Routing for IP-to-Tel Calls .................................................................. 271
23.6.1 Alternative Routing to Trunk upon Q.931 Call Release Cause Code ...................271
23.6.2 Alternative Routing to an IP Destination upon a Busy Trunk ................................272
24 Configuring DTMF and Dialing ....................................................................... 275
24.1 Dialing Plan Features ........................................................................................... 275
24.1.1 Digit Mapping .........................................................................................................275
24.1.2 External Dial Plan File ...........................................................................................277
25 Configuring Supplementary Services ........................................................... 279
25.1 Call Hold and Retrieve ......................................................................................... 281
25.2 Call Pickup ........................................................................................................... 283
25.3 Consultation Feature ............................................................................................ 283
25.4 Call Transfer......................................................................................................... 284
25.4.1 Consultation Call Transfer .....................................................................................284
25.4.2 Blind Call Transfer .................................................................................................284
25.5 Call Forward ......................................................................................................... 284
25.5.1 Call Forward Reminder Ring .................................................................................285
25.5.2 Call Forward Reminder (Off-Hook) Special Dial Tone ..........................................286

MP-11x & MP-124
25.5.3 Call Forward Reminder Dial Tone (Off-Hook) upon Spanish SIP Alert-Info..........286
25.6 Call Waiting .......................................................................................................... 287
25.7 Message Waiting Indication ................................................................................. 287
25.8 Caller ID ............................................................................................................... 288
25.8.1 Caller ID Detection / Generation on the Tel Side ..................................................288
25.8.2 Debugging a Caller ID Detection on FXO..............................................................289
25.8.3 Caller ID on the IP Side .........................................................................................289
25.9 Three-Way Conferencing ..................................................................................... 290
25.10 Emergency E911 Phone Number Services.......................................................... 293
25.10.1 Pre-empting Existing Calls for E911 IP-to-Tel Calls ..............................................293
25.11 Multilevel Precedence and Preemption................................................................ 293
25.11.1 MLPP Preemption Events in SIP Reason Header ................................................294
25.11.2 Precedence Ring Tone ..........................................................................................295
25.12 Denial of Collect Calls .......................................................................................... 296
25.13 Configuring Voice Mail ......................................................................................... 297
25.14 Out-of-Band Digit Notifications According to KPML ............................................. 298
26 Analog Gateway .............................................................................................. 299
26.1 Configuring Keypad Features .............................................................................. 299
26.2 Configuring Metering Tones ................................................................................. 301
26.3 Configuring Charge Codes................................................................................... 302
26.4 Configuring FXO Settings .................................................................................... 303
26.5 Configuring Authentication ................................................................................... 304
26.6 Configuring Automatic Dialing .............................................................................. 305
26.7 Configuring Caller Display Information................................................................. 307
26.8 Configuring Call Forward ..................................................................................... 309
26.9 Configuring Caller ID Permissions ....................................................................... 310
26.10 Configuring Call Waiting....................................................................................... 311
26.11 Rejecting Anonymous Calls ................................................................................. 312
26.12 Configuring FXS Distinctive Ringing and Call Waiting Tones per
Source/Destination Number .......................................................................................... 312
26.13 FXS/FXO Coefficient Types ................................................................................. 314
26.14 FXO Operating Modes ......................................................................................... 315
26.14.1 FXO Operations for IP-to-Tel Calls ........................................................................315
26.14.1.1 One-Stage Dialing ................................................................................. 315
26.14.1.2 Two-Stage Dialing ................................................................................. 316
26.14.1.3 DID Wink ............................................................................................... 317
26.14.2 FXO Operations for Tel-to-IP Calls........................................................................317
26.14.2.1 Automatic Dialing .................................................................................. 317
26.14.2.2 Collecting Digits Mode........................................................................... 319
26.14.2.3 FXO Supplementary Services ............................................................... 319
26.14.3 Call Termination on FXO Devices .........................................................................320
26.14.3.1 Calls Termination by PBX ..................................................................... 320
26.14.3.2 Call Termination before Call Establishment .......................................... 321
26.14.3.3 Ring Detection Timeout ......................................................................... 321
26.15 Remote PBX Extension between FXO and FXS Devices .................................... 321
26.15.1 Dialing from Remote Extension (Phone at FXS) ...................................................322
26.15.2 Dialing from PBX Line or PSTN .............................................................................323
26.15.3 Message Waiting Indication for Remote Extensions .............................................323
26.15.4 Call Waiting for Remote Extensions ......................................................................323
26.15.5 FXS Gateway Configuration ..................................................................................324
26.15.6 FXO Gateway Configuration ..................................................................................325

Stand-Alone Survivability Application .................................................................327

27 SAS Overview .................................................................................................. 329
27.1 SAS Operating Modes ......................................................................................... 329
27.1.1 SAS Outbound Mode .............................................................................................329
27.1.1.1 Normal State ......................................................................................... 330
27.1.1.2 Emergency State ................................................................................... 330
27.1.2 SAS Redundant Mode ...........................................................................................331
27.1.2.1 Normal State ......................................................................................... 332
27.1.2.2 Emergency State ................................................................................... 332
27.1.2.3 Exiting Emergency and Returning to Normal State .............................. 332
27.2 SAS Routing......................................................................................................... 333
27.2.1 SAS Routing in Normal State ................................................................................333
27.2.2 SAS Routing in Emergency State ..........................................................................335
28 SAS Configuration .......................................................................................... 337
28.1 General SAS Configuration .................................................................................. 337
28.1.1 Enabling the SAS Application ................................................................................337
28.1.2 Configuring Common SAS Parameters .................................................................337
28.2 Configuring SAS Outbound Mode ........................................................................ 340
28.3 Configuring SAS Redundant Mode ...................................................................... 340
28.4 Configuring Gateway Application with SAS ......................................................... 341
28.4.1 Gateway with SAS Outbound Mode ......................................................................341
28.4.2 Gateway with SAS Redundant Mode ....................................................................343
28.5 Advanced SAS Configuration............................................................................... 345
28.5.1 Manipulating URI user part of Incoming REGISTER and/or INVITE .....................345
28.5.2 Manipulating Destination Number of Incoming INVITE .........................................346
28.5.3 SAS Routing Based on IP-to-IP Routing Table .....................................................349
28.5.4 Blocking Calls from Unregistered SAS Users........................................................354
28.5.5 Configuring SAS Emergency Calls ........................................................................354
28.5.6 Adding SIP Record-Route Header to SIP INVITE .................................................355
28.5.7 Re-using TCP Connections ...................................................................................356
28.5.8 Replacing Contact Header for SIP Messages .......................................................356
28.6 Viewing Registered SAS Users............................................................................ 357
29 SAS Cascading ................................................................................................ 359
Maintenance ...........................................................................................................361
30 Basic Maintenance .......................................................................................... 363
30.1 Resetting the Device ............................................................................................ 363
30.2 Remotely Resetting Device using SIP NOTIFY ................................................... 364
30.3 Locking and Unlocking the Device ....................................................................... 365
30.4 Saving Configuration ............................................................................................ 366
31 Resetting an Analog Channel ........................................................................ 367
32 Software Upgrade............................................................................................ 369
32.1 Loading Auxiliary Files ......................................................................................... 369
32.1.1 Call Progress Tones File .......................................................................................371
32.1.1.1 Distinctive Ringing ................................................................................. 373

MP-11x & MP-124
32.1.2 Prerecorded Tones File .........................................................................................375

32.1.3 Dial Plan File ..........................................................................................................376
32.1.3.1 Creating a Dial Plan File........................................................................ 376
32.1.3.2 Dialing Plans for Digit Collection ........................................................... 376
32.1.3.3 Obtaining IP Destination from Dial Plan File ......................................... 378
32.1.4 User Information File .............................................................................................379
32.1.4.1 User Information File for PBX Extensions and "Global" Numbers ........ 379
32.1.4.2 Enabling the User Info Table ................................................................. 381
32.2 Software License Key .......................................................................................... 381
32.2.1 Obtaining the Software License Key File...............................................................382
32.2.2 Installing the Software License Key.......................................................................383
32.2.2.1 Installing Software License Key using Web Interface ........................... 383
32.2.2.2 Installing Software License Key using BootP/TFTP.............................. 384
32.3 Software Upgrade Wizard .................................................................................... 385
32.4 Backing Up and Loading Configuration File ......................................................... 388
33 Automatic Update............................................................................................ 389
33.1 Automatic Configuration Methods ........................................................................ 389
33.1.1 BootP Request and DHCP Discovery upon Device Initialization ..........................389
33.1.2 VLAN ID Discovery using LLDP ............................................................................391
33.1.3 Local Configuration Server with BootP/TFTP ........................................................391
33.1.4 DHCP-based Provisioning .....................................................................................392
33.1.4.1 Provisioning from HTTP Server using DHCP Option 67 ....................... 393
33.1.4.2 Provisioning from TFTP Server using DHCP Option 66 ....................... 394
33.1.4.3 Provisioning the Device using DHCP Option 160 ................................. 394
33.1.5 HTTP-based Provisioning ......................................................................................395
33.1.5.1 Loading Files Securely by Disabling TFTP ........................................... 396
33.1.6 FTP- or NFS-based Provisioning...........................................................................397
33.1.7 Provisioning using AudioCodes EMS ....................................................................397
33.2 HTTP/S-Based Provisioning using the Automatic Update Feature ...................... 397
33.2.1 Files Provisioned by Automatic Update .................................................................398
33.2.2 File Location for Automatic Update .......................................................................398
33.2.3 Triggers for Automatic Update ...............................................................................398
33.2.4 Access Authentication with HTTP Server ..............................................................399
33.2.5 Querying Provisioning Server for Updated Files ...................................................399
33.2.6 File Download Sequence .......................................................................................401
33.2.7 Cyclic Redundancy Check on Downloaded Configuration Files ...........................402
33.2.8 MAC Address Automatically Inserted in Configuration File Name ........................403
33.2.9 Automatic Update Configuration Examples ...........................................................403
33.2.9.1 Automatic Update for Single Device ..................................................... 403
33.2.9.2 Automatic Update from NFS, FTP and HTTP Servers ......................... 404
33.2.9.3 Automatic Update for Mass Deployment............................................... 405
34 Restoring Factory Defaults ............................................................................ 407
34.1 Restoring Defaults using CLI ............................................................................... 407
34.2 Restoring Defaults using Hardware Reset Button................................................ 407
34.3 Restoring Defaults using an ini File...................................................................... 408
Status, Performance Monitoring and Reporting .................................................409

35 System Status ................................................................................................. 411
35.1 Viewing Device Information.................................................................................. 411
35.2 Viewing Ethernet Port Information ....................................................................... 411

36 Carrier-Grade Alarms ...................................................................................... 413

36.1 Viewing Active Alarms.......................................................................................... 413
36.2 Viewing Alarm History .......................................................................................... 413
37 VoIP Status ...................................................................................................... 415
37.1 Viewing Analog Port Information .......................................................................... 415
37.2 Viewing Active IP Interfaces................................................................................. 415
37.3 Viewing Performance Statistics............................................................................ 416
37.4 Viewing Call Counters .......................................................................................... 416
37.5 Viewing Registered Users .................................................................................... 418
37.6 Viewing Registration Status ................................................................................. 419
37.7 Viewing Call Routing Status ................................................................................. 420
37.8 Viewing IP Connectivity........................................................................................ 421
38 Reporting Information to External Party ....................................................... 423
38.1 RTP Control Protocol Extended Reports (RTCP XR) .......................................... 423
38.2 Generating Call Detail Records............................................................................ 428
38.2.1 Configuring CDR Reporting ...................................................................................428
38.2.2 CDR Field Description ...........................................................................................429
38.2.2.1 CDR Fields for Gateway/IP-to-IP Application ....................................... 429
38.2.2.2 Release Reasons in CDR ..................................................................... 432
38.3 Configuring RADIUS Accounting ......................................................................... 435
38.4 Event Notification using X-Detect Header ............................................................ 438
38.5 Querying Device Channel Resources using SIP OPTIONS ................................ 440
Diagnostics ............................................................................................................441
39 Syslog and Debug Recordings ...................................................................... 443
39.1 Syslog Message Format ...................................................................................... 443
39.1.1 Event Representation in Syslog Messages ...........................................................444
39.1.2 Identifying AudioCodes Syslog Messages using Facility Levels ...........................446
39.1.3 SNMP Alarms in Syslog Messages .......................................................................447
39.2 Configuring Syslog Settings ................................................................................. 448
39.3 Configuring Debug Recording .............................................................................. 449
39.4 Filtering Syslog Messages and Debug Recordings ............................................. 449
39.4.1 Filtering IP Network Traces ...................................................................................451
39.5 Viewing Syslog Messages ................................................................................... 453
39.6 Collecting Debug Recording Messages ............................................................... 454
40 Self-Testing ...................................................................................................... 457
41 Line Testing ..................................................................................................... 459
41.1 FXS Line Testing.................................................................................................. 459
41.2 FXO Line Testing ................................................................................................. 460
42 Testing SIP Signaling Calls ............................................................................ 461
42.1 Configuring Test Call Endpoints........................................................................... 461
42.1.1 Starting, Stopping and Restarting Test Calls.........................................................464
42.1.2 Viewing Test Call Statistics....................................................................................465

MP-11x & MP-124
42.2 Configuring DTMF Tones for Test Calls............................................................... 466

42.3 Configuring Basic Test Call .................................................................................. 467
42.4 Test Call Configuration Examples ........................................................................ 468
Appendix ................................................................................................................471
43 Dialing Plan Notation for Routing and Manipulation.................................... 473
44 Configuration Parameters Reference ............................................................ 475
44.1 Networking Parameters........................................................................................ 475
44.1.1 Ethernet Parameters..............................................................................................475
44.1.2 Multiple VoIP Network Interfaces and VLAN Parameters .....................................475
44.1.3 Routing Parameters ...............................................................................................477
44.1.4 Quality of Service Parameters ...............................................................................478
44.1.5 NAT and STUN Parameters ..................................................................................479
44.1.6 NFS Parameters ....................................................................................................482
44.1.7 DNS Parameters....................................................................................................482
44.1.8 DHCP and LLDP Parameters ................................................................................483
44.1.9 NTP and Daylight Saving Time Parameters ..........................................................484
44.2 Management Parameters..................................................................................... 486
44.2.1 General Parameters ..............................................................................................486
44.2.2 Web Parameters ....................................................................................................486
44.2.3 Telnet Parameters .................................................................................................489
44.2.4 SNMP Parameters .................................................................................................490
44.2.5 TR-069 Parameters ...............................................................................................493
44.2.6 Serial Parameters ..................................................................................................495
44.3 Debugging and Diagnostics Parameters.............................................................. 495
44.3.2 SIP Test Call Parameters ......................................................................................498
44.3.3 Syslog, CDR and Debug Parameters ....................................................................498
44.3.4 Resource Allocation Indication Parameters...........................................................502
44.3.5 BootP Parameters .................................................................................................503
44.4 Security Parameters............................................................................................. 504
44.4.2 HTTPS Parameters ...............................................................................................505
44.4.3 SRTP Parameters..................................................................................................507
44.4.4 TLS Parameters.....................................................................................................509
44.4.5 SSH Parameters ....................................................................................................511
44.4.6 IPSec Parameters..................................................................................................512
44.4.7 802.1X Parameters ................................................................................................514
44.4.8 OCSP Parameters .................................................................................................515
44.5 RADIUS Parameters ............................................................................................ 516
44.6 SIP Media Realm Parameters.............................................................................. 518
44.7 Control Network Parameters ................................................................................ 519
44.7.1 IP Group, Proxy, Registration and Authentication Parameters .............................519
44.8 General SIP Parameters ...................................................................................... 530
44.9 Coders and Profile Parameters ............................................................................ 552
44.10 Channel Parameters ............................................................................................ 556
44.10.1 Voice Parameters ..................................................................................................556
44.10.2 Coder Parameters .................................................................................................557
44.10.3 DTMF Parameters .................................................................................................558
44.10.4 RTP, RTCP and T.38 Parameters .........................................................................560
44.11 Gateway and IP-to-IP Parameters ....................................................................... 566
44.11.1 Fax and Modem Parameters .................................................................................566

44.11.2 DTMF and Hook-Flash Parameters.......................................................................572

44.11.3 Digit Collection and Dial Plan Parameters.............................................................576
44.11.4 Voice Mail Parameters...........................................................................................578
44.11.5 Supplementary Services Parameters ....................................................................582
44.11.5.1 Caller ID Parameters ............................................................................. 582
44.11.5.2 Call Waiting Parameters........................................................................ 587
44.11.5.3 Call Forwarding Parameters ................................................................. 589
44.11.5.4 Message Waiting Indication Parameters............................................... 590
44.11.5.5 Call Hold Parameters ............................................................................ 594
44.11.5.6 Call Transfer Parameters ...................................................................... 595
44.11.5.7 Three-Way Conferencing Parameters .................................................. 597
44.11.5.8 MLPP and Emergency Call Parameters ............................................... 599
44.11.5.9 Call Cut-Through Parameters ............................................................... 602
44.11.5.10 Automatic Dialing Parameters ......................................................... 603
44.11.5.11 Direct Inward Dialing Parameters .................................................... 604
44.11.6 Answer and Disconnect Supervision Parameters .................................................605
44.11.7 Tone Parameters ...................................................................................................611
44.11.7.1 Telephony Tone Parameters ................................................................. 611
44.11.7.2 Tone Detection Parameters .................................................................. 614
44.11.7.3 Metering Tone Parameters ................................................................... 615
44.11.8 Telephone Keypad Sequence Parameters............................................................616
44.11.9 General FXO Parameters ......................................................................................620
44.11.10 Hunt Groups and Routing Parameters .............................................................624
44.11.11 IP Connectivity Parameters ..............................................................................629
44.11.12 Alternative Routing Parameters .......................................................................631
44.11.13 Number Manipulation Parameters....................................................................633
44.12 Least Cost Routing Parameters ........................................................................... 638
44.13 Standalone Survivability Parameters ................................................................... 639
44.14 Auxiliary and Configuration File Name Parameters ............................................. 644
44.15 Automatic Update Parameters ............................................................................. 646
45 DSP Templates ................................................................................................ 649
46 Selected Technical Specifications ................................................................. 651

MP-11x & MP-124
This page is intentionally left blank.

User's Manual Notices
Notice
Information contained in this document is believed to be accurate and reliable at the time of
printing. However, due to ongoing product improvements and revisions, AudioCodes cannot
guarantee accuracy of printed material after the Date Published nor can it accept responsibility
for errors or omissions. Before consulting this document, check the corresponding Release
Notes regarding feature preconditions and/or specific support in this release. In cases where
there are discrepancies between this document and the Release Notes, the information in the
Release Notes supersedes that in this document. Updates to this document and other
documents as well as software files can be downloaded by registered customers at
http://www.audiocodes.com/downloads.
This document is subject to change without notice.
Date Published: October-03-2017
WEEE EU Directive
Pursuant to the WEEE EU Directive, electronic and electrical waste must not be disposed
of with unsorted waste. Please contact your local recycling authority for disposal of this
product.
Customer Support
Customer technical support and services are provided by AudioCodes or by an authorized
AudioCodes Service Partner. For more information on how to buy technical support for
AudioCodes products and for contact information, please visit our Web site at
www.audiocodes.com/support.
Abbreviations and Terminology

Each abbreviation, unless widely used, is spelled out in full when first used.
Throughout this manual, unless otherwise specified, the following naming conventions are
used:
 The term device refers to the MediaPack series gateways.
 The term MediaPack refers to MP-124, MP-118, MP-114, and MP-112.
 The term MP-11x refers to MP-118, MP-114, and MP-112.

MP-11x & MP-124
Regulatory Information
The Regulatory Information can be viewed at http://www.audiocodes.com/downloads.
Related Documentation
Manual Name
SIP CPE Release Notes

MP-11x & MP-124 SIP Installation Manual
MP-11x SIP Fast Track Guide
MP-124 AC SIP Fast Track Guide
MP-124 DC SIP Fast Track Guide
CPE Configuration Guide for IP Voice Mail
DConvert User's Guide
SNMP User's Guide
Notes and Warnings
Note: The scope of this document does not fully cover security aspects for deploying
the device in your environment. Security measures should be done in accordance
with your organization’s security policies. For basic security guidelines, you should
refer to AudioCodes Recommended Security Guidelines document.
Note: Before configuring the device, ensure that it is installed correctly as instructed
in the Hardware Installation Manual.
Note: This device supports the SAS and/or Gateway / IP-to-IP applications; not the
SBC application.
Legal Notice:
• This device includes software developed by the OpenSSL Project for use in the
OpenSSL Toolkit (http://www.openssl.org/).
• This device includes cryptographic software written by Eric Young
([email protected]).

User's Manual Notices
Document Revision Record
LTRT Description
65423 Warning bulletin regarding indoor installation and cabling; CLI commands in procedure
for assigning OAMP IP address; Command Shell commands; VDC for lighting lamp for
MWI; irrelevant parameters removed (GWInboundManipulationSet,
GWOutboundManipulationSet); command correction for FXS line testing.
65424 TR-069; maximum resolved IP addresses per DNS query; three-way conference
example; CPTWizard removed; maximum number of User Info file rules.
65425 Automatic update chapter revised.
65425 New parameter added – EnableLLDP.
65427 SAS feature key removed.
65428 FXS Line Testing updated to include MP-118 for Command Shell commands and
updated notes.
65429 MP-124 Rev. E added.
65430 Maximum channel capacity updated; Out-of-Band Digit Notifications According to KPML
(AdditionalOutOfBandDtmfFormat); Note added for User Information File syntax re
spaces; PublicationIPGroupID (new); RTCPXRReportMode (new option [3]);
GwSDPConnectionMode (new); description of CutThrough updated.
65431 TLSVersion parameter updated; note added to LifeLineType parameter description.
65432  New sections: Configuring Media Realms; Configuring Quality of Experience per
Media Realm; Quality of Experience; Reporting Voice Quality of Experience to SEM;
Configuring the SEM Server; Configuring Clock Synchronization between Device
and SEM; Enabling RTCP XR Reporting to SEM; SIP Media Realm Parameters
 New parameters: IgnoreAuthorizationStale; CpMediaRealm; QOEServerIP;
QOEInterfaceName; QOERules
65433  Updated sections: FXS Voice Menu Guidance; Alternative Routing Based on IP
Connectivity (typo)
 Updated parameters: PSTNPrefix_SourceAddress; ResetWebPassword;
SSHMaxLoginAttempts; SecureCallsFromIP
65434  New sections: Configuring Password Display in ini File
 Updated sections: Advanced User Accounts Configuration (ini file parameters); SIP
Calling Name Manipulations (max. rows); Selected Technical Specifications (MWI)
 Updated parameters: WebUsers_Password (note); FaxBypassPayloadType
 New parameters: INIPasswordsDisplayType
65435  Updated sections: Configuring Voice Settings (silence suppression removed);
Silence Suppression (removed); Fax / Modem Transparent Mode (silence
suppression removed); Configuring Coders (silence suppression removed);
Message Waiting Indication (lamp voltage); Viewing Active Alarms (note)
 New parameters: ActiveAlarmTableMaxSize; NoAlarmForDisabledPort;
EnableLowVoltageMwiGeneration; LedMwiOnDurationTime;
LedMwiOffDurationTime; NeonMwiOnDurationTime; NeonMwiOffDurationTime
 Updated parameters: IsCiscoSCEMode; EnableSilenceCompression (removed);
EnableSilenceDisconnect; UseDisplayNameAsSourceNumber

MP-11x & MP-124
LTRT Description
65436  Updated with patch version 6.60A.340.001

 Updated sections: Manipulating URI user part of Incoming REGISTER
 Updated parameters: NoAlarmForDisabledPort (Web); TLSVersion;
SASRegistrationManipulation (SASRegistrationManipulation_RuleApplyTo);
EnableLowVoltageMwiGeneration; LedMwiOnDurationTime;
LedMwiOffDurationTime; NeonMwiOnDurationTime; NeonMwiOffDurationTime
65437  Updated with patch version 6.60A.342.003
 Updated sections: Multiple Interface Table Configuration Rules (VLANs); VoIP
Interfaces for Combined Application Types (VLANs); Configuring Proxy Sets Table;
Configuring Account Table (max.); SAS Configuration (all subsections – path to
pages); Manipulating URI user part of Incoming REGISTER and/or INVITE (new
SAS Registration Table)
 New sections: Provisioning the Device using DHCP Option 160
 New parameters: ProxySet_HomingSuccessDetectionRetries;
DhcpOption160Support
Documentation Feedback
AudioCodes continually strives to produce high quality documentation. If you have any
comments (suggestions or errors) regarding this document, please fill out the
Documentation Feedback form on our Web site at http://online.audiocodes.com/doc-
feedback.

User's Manual 1. Overview
1 Overview
The MediaPack series analog Voice-over-IP (VoIP) Session Initiation Protocol (SIP) media
gateways (hereafter referred to as device) are cost-effective, cutting edge technology
products. These stand-alone analog VoIP devices provide superior voice technology for
connecting legacy telephones, fax machines and Private Branch Exchange (PBX) systems
to IP-based telephony networks, as well as for integration with new IP-based PBX
architectures. These devices are designed and tested to be fully interoperable with leading
softswitches and SIP servers.
The device is best suited for small and medium-sized enterprises (SME), branch offices, or
residential media gateway solutions. The device enables users to make local or
international telephone and / or fax calls over the Internet between distributed company
offices, using their existing telephones and fax. These calls are routed over the existing
network ensuring that voice traffic uses minimum bandwidth. The device also provides SIP
trunking capabilities for Enterprises operating with multiple Internet Telephony Service
Providers (ITSP) for VoIP services.
The device supports the SIP protocol, enabling the deployment of VoIP solutions in
environments where each enterprise or residential location is provided with a simple media
gateway. This provides the enterprise with a telephone connection (i.e., RJ-11 connector)
and the capability to transmit voice and telephony signals over a packet network.
The device provides FXO and/or FXS analog ports for direct connection to an enterprise's
PBX (FXO), and / or to phones, fax machines, and modems (FXS). Depending on model,
the device can support up to 24 simultaneous VoIP calls. The device is also equipped with
a 10/100Base-TX Ethernet port for connection to the IP network. The device provides
LEDs for indicating operating status of the various interfaces.
The device is a compact unit that can be easily mounted on a desktop, wall, or in a 19-inch
rack.
The device provides a variety of management and provisioning tools, including an HTTP-
based embedded Web server, Telnet, Element Management System (EMS), and Simple
Network Management Protocol (SNMP). The user-friendly, Web interface provides remote
configuration using any standard Web browser (such as Microsoft™ Internet Explorer™).
The figure below illustrates a typical MediaPack VoIP application.
Figure 1-1: Typical MediaPack VoIP Application

MP-11x & MP-124
1.1 MediaPack Models

The analog MediaPack 1xx models and their corresponding supported configurations are
listed in the table below:
Table 1-1: MediaPack 1xx Models and Configurations
Combined FXS / Number of

MediaPack Model FXS FXO
FXO Channels
MP-124 Yes No No 24
MP-118 Yes Yes 4+4 8
MP-114 Yes Yes 2+2 4
MP-112* Yes No No 2
* The MP-112 differs from the MP-114 and MP-118 in that its configuration excludes the
RS-232 connector and Lifeline option.
1.2 SIP Overview

Session Initiation Protocol (SIP) is an application-layer control (signaling) protocol used on
the gateway for creating, modifying, and terminating sessions with one or more
participants. These sessions can include Internet telephone calls, media announcements,
and conferences.
SIP invitations are used to create sessions and carry session descriptions that enable
participants to agree on a set of compatible media types. SIP uses elements called Proxy
servers to help route requests to the user's current location, authenticate and authorize
users for services, implement provider call-routing policies and provide features to users.
SIP also provides a registration function that enables users to upload their current locations
for use by Proxy servers. SIP implemented in the gateway, complies with the Internet
Engineering Task Force (IETF) RFC 3261 (refer to http://www.ietf.org).

The SIP call flow, shown in the figure below, describes SIP messages exchanged between
two devices during a basic call. In this call flow example, device 10.8.201.108 with phone
number 6000, dials device 10.8.201.161 with phone number 2000.
Figure 1-2: SIP Call Flow
 F1 INVITE - 10.8.201.108 to 10.8.201.161:

INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 10.8.201.108;branch=z9hG4bKacsiJkDGd
From: <sip:[email protected]>;tag=1c5354
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 18153 INVITE
Contact: <sip:[email protected];user=phone>
User-Agent: Audiocodes-Sip-Gateway/MediaPack/v.6.60.010.006
Supported: 100rel,em
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,
NOTIFY,PRACK,REFER,INFO
Content-Type: application/sdp
Content-Length: 208
v=0
o=AudiocodesGW 18132 74003 IN IP4 10.8.201.108
s=Phone-Call
c=IN IP4 10.8.201.108
t=0 0
m=audio 4000 RTP/AVP 8 96
a=rtpmap:8 pcma/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=ptime:20
 F2 TRYING - 10.8.201.161 to 10.8.201.108:
SIP/2.0 100 Trying
To: <sip:[email protected]>
Server: Audiocodes-Sip-Gateway/MediaPack/v.6.60.010.006

MP-11x & MP-124
CSeq: 18153 INVITE

Content-Length: 0
 F3 RINGING 180 - 10.8.201.161 to 10.8.201.108:
SIP/2.0 180 Ringing
To: <sip:[email protected]>;tag=1c7345
CSeq: 18153 INVITE
Content-Length: 0
Note: Phone 2000 answers the call and then sends a SIP 200 OK response to
device 10.8.201.108.
 F4 200 OK - 10.8.201.161 to 10.8.201.108:

SIP/2.0 200 OK
CSeq: 18153 INVITE
Contact: <sip:[email protected];user=phone>
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,
NOTIFY,PRACK,REFER,INFO
Content-Type: application/sdp
Content-Length: 206
v=0
o=AudiocodesGW 30221 87035 IN IP4 10.8.201.161
s=Phone-Call
c=IN IP4 10.8.201.10
t=0 0
m=audio 7210 RTP/AVP 8 96
a=rtpmap:8 pcma/8000
a=ptime:20
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
 F5 ACK - 10.8.201.108 to 10.8.201.10:
ACK sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 10.8.201.108;branch=z9hG4bKacZYpJWxZ
CSeq: 18153 ACK
Content-Length: 0
Note: Phone 6000 goes on-hook and device 10.8.201.108 sends a BYE to device
10.8.201.161 and a voice path is established.

 F6 BYE - 10.8.201.108 to 10.8.201.10:

BYE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 10.8.201.108;branch=z9hG4bKacRKCVBud
CSeq: 18154 BYE
Content-Length: 0
 F7 OK 200 - 10.8.201.10 to 10.8.201.108:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.8.201.108;branch=z9hG4bKacRKCVBud
CSeq: 18154 BYE
Content-Length: 0

MP-11x & MP-124

Part I
Getting Started with Initial
Connectivity
User's Manual 2. Assigning the OAMP IP Address
2 Assigning the OAMP IP Address

The device is shipped with a factory default IP address for its operations, administration,
maintenance, and provisioning (OAMP) interface, as shown in the table below:
Table 2-1: Default OAMP IP Address
IP Address Value
IP Address  FXS and FXS / FXO devices: 10.1.10.10

 FXO device: 10.1.10.11
Note: FXO interfaces are applicable only to MP-11x series devices.
Subnet Mask 255.255.0.0
Default Gateway IP 0.0.0.0
Address
The default IP address can be used for initially accessing the device, using any of its
management tools (i.e., embedded Web server, EMS, or Telnet). Once accessed, you can
change this default IP address to correspond with your networking scheme in which the
device is deployed. After changing the IP address, you can re-access the device with this
new OAMP IP address and start configuring and managing the device as desired.
This section describes the different methods for changing the device's default IP address to
suit your networking environment:
 Embedded command line interface (CLI) - see 'CLI' on page 30
 Embedded HTTP/S-based Web server - see 'Web Interface' on page 27
 Bootstrap Protocol (BootP) - see BootP/TFTP Server on page 29
 FXS telephone voice menu - see FXS Voice Menu Guidance on page 32
2.1 Web Interface

The procedure below describes how to assign an OAMP IP address using the Web
interface.
 To assign an OAMP IP address using the Web interface:

1. Disconnect the network cables (if connected) from the device.
2. Connect the Ethernet port located on the rear panel (labeled Ethernet) directly to the
network interface of your computer, using a straight-through Ethernet cable.
Figure 2-1: MP-11x Ethernet Connection to PC for Initial Connectivity

MP-11x & MP-124
Figure 2-2: MP-124 Ethernet Connection to PC for Initial Connectivity (e.g., MP-124 Rev. E)
3. Change the IP address and subnet mask of your computer to correspond with the
default IP address and subnet mask of the device.
4. Access the Web interface:
a. On your computer, start a Web browser and in the URL address field, enter the
default IP address of the device; the Web interface's Login screen appears:
Figure 2-3: Web Login Screen
b. In the 'Username' and 'Password' fields, enter the default login user name
("Admin" - case-sensitive) and password ("Admin" - case-sensitive), and then
click Login; the device's Web interface is accessed.
5. Change the default IP address to one that corresponds with your network:
a. Open the Multiple Interface Table page (Configuration tab > VoIP menu >
Network submenu > IP Settings).
Figure 2-4: IP Settings Page (Single Network Interface)

b. Select the 'Index' radio button corresponding to the "OAMP + Media + Control"
application type, and then click Edit.
c. Change the IP address, subnet mask, and Default Gateway IP address to
correspond with your network IP addressing scheme.
d. Click Apply, and then click Done to validate your settings.
6. Save your settings to the flash memory with a device reset (see Resetting the Device
on page 363).
7. Disconnect the computer from the device and then reconnect the device to your
network.
2.2 BootP/TFTP Server

You can assign an IP address to the device using BootP/TFTP protocols. This can be done
using the AudioCodes AcBootP utility (supplied) or any standard compatible BootP server.
Note: You can also use the AcBootP utility to load the software file (.cmp) and
configuration file (.ini). For a detailed description of the AcBootP utility, refer to
AcBootP Utility User's Guide.
 To assign an IP address using BootP/TFTP:

1. Start the AcBootP utility.
2. Select the Preferences tab, and then set the 'Timeout' field to "50".
3. Select the Client Configuration tab, and then click the Add New Client button.
Figure 2-5: BootP Client Configuration Screen
4. Configure the following fields:

• ‘Client MAC’: Enter the device's MAC address. The MAC address is printed on
the label located on the underside of the device. Ensure that the check box to the
right of the field is selected in order to enable the client.

MP-11x & MP-124
• 'Client IP’: Enter the new IP address (in dotted-decimal notation) that you want to
assign the device.
• ‘Subnet’: Enter the new subnet mask (in dotted-decimal notation) that you want to
assign the device.
• ‘Gateway’: Enter the IP address of the Default Gateway (if required).
5. Click Apply to save the new client.
6. Physically reset the device by powering it down and then up again. This enables the
device to receive its new networking parameters through the BootP process.
2.3 CLI
The procedure below describes how to assign an OAMP IP address, using CLI.
Note: Assigning an IP address using CLI is not applicable to MP-112 as this model
does not provide RS-232 serial interface.
 To assign an OAMP IP address using CLI:

1. Connect the RS-232 port of the device to the serial communication port on your
computer. For more information, refer to the Hardware Installation Manual.
Figure 2-6: MP-11x Serial Connection with PC for CLI Communication

Figure 2-7: MP-124 Serial Connection with PC for CLI Communication (e.g., MP-124 Rev. E)
2. Establish serial communication with the device using a terminal emulator program
(such as HyperTerminal) with the following communication port settings:
• Baud Rate: 115,200 bps for MP-124 and 9,600 bps for MP-11x
• Data Bits: 8
• Parity: None
• Stop Bits: 1
• Flow Control: None
3. At the prompt, type the login username (default is "Admin" - case sensitive):
login: Admin
4. At the prompt, type the password (default is "Admin" - case sensitive):
password: Admin
5. At the prompt, type the following command to access the Configuration folder:
/>CONF
6. View the current network settings, by typing the following command:
/CONFiguration>GCP IP
7. Change the network settings, by typing the following command:
/CONFiguration>SCP IP <IP address> <subnet mask> <Default Gateway>
You must enter all three network parameters, each separated by a space, for
example:
/CONFiguration>SCP IP 10.13.77.7 255.255.0.0 10.13.0.1
8. Save your changes and reset the device, by typing the following command:
/CONFiguration>SAR

MP-11x & MP-124
2.4 FXS Voice Menu Guidance

You can assign an IP address that suits your networking scheme using a standard touch-
tone telephone connected to one of the FXS ports. The FXS voice menu can also be used
to query and modify basic configuration parameters.
Notes: If you want to disable the FXS voice menu, do one of the following:
• Set the VoiceMenuPassword parameter to 'disable'.
• Change the Web login password for the Admin user from its default value (i.e.,
"Admin") to any other value, and then reset the device.
 To assign an IP address using the voice menu:

1. Connect a telephone to one of the FXS ports.
2. Lift the handset and dial ***12345 (three stars followed by the digits 1, 2, 3, 4, and 5).
Notes: When dialing ***, a fast-busy tone may be heard.
3. Wait for the 'configuration menu' voice prompt to be played.

4. To change the IP address:
a. Press 1 followed by the pound key (#); the current IP address of the device is
played.
b. Press the # key.
c. Dial the new IP address, using the star (*) key instead of periods (.), e.g.,
192*168*0*4, and then press # to finish.
d. Review the new IP address, and then press 1 to save.
5. To change the subnet mask:
a. Press 2 followed by the # key; the current subnet mask of the device is played.
b. Press the # key.
c. Dial the new subnet mask (e.g., 255*255*0*0), and then press # to finish.
d. Review the new subnet mask, and then press 1 to save.

6. To change the Default Gateway IP address:

a. Press 3 followed by the # key; the current Default Gateway address is played.
b. Press the # key.
c. Dial the new Default Gateway address (e.g., 192*168*0*1), and then press # to
finish.
d. Review the new Default Gateway address, and then press 1 to save.
7. Hang up (on-hook) the handset.
Alternatively, initial configuration may be performed using an HTTP server. The Voice
Menu may be used to specify the configuration URL.
 To set a configuration URL:

1. Obtain the IP address of the configuration HTTP server (e.g., 36.44.0.6).
2. Connect a telephone to one of the FXS ports.
3. Lift the handset and dial ***12345 (three stars followed by the digits 1, 2, 3, 4, and 5).
4. Wait for the "configuration menu" voice prompt to be played.
5. Dial 31 followed by the # key; the current IP address is played.
6. To change the IP address:
a. Press the # key.
b. Dial the configuration server's IP address. Use the star (*) key instead of dots
("."), e.g., 36*44*0*6, and then press # to finish.
c. Review the configuration server's IP address, and then press 1 to save.
7. Dial 32 followed by the # key, and then do the following to change the configuration
file name pattern:
a. Press the # key.
b. Select one of the patterns listed in the table below (aa.bb.cc.dd denotes the IP
address of the configuration server):
# Configuration File Name Pattern Description

1 http://aa.bb.cc.dd/config.ini Standard config.ini.
2 https://aa.bb.cc.dd/config.ini Secure HTTP.
The device's MAC address is appended to the file
3 http://aa.bb.cc.dd/audiocodes/<MAC>.ini name (e.g.,
http://36.44.0.6/audiocodes/00908f012300.ini).
4 http://aa.bb.cc.dd:8080/config.ini HTTP on port 8080.
5 http://aa.bb.cc.dd:1400/config.ini HTTP on port 1400.
Generating configuration per IP/MAC address
http://aa.bb.cc.dd/cgi-
6 dynamically, using a CGI script. See perl example
bin/acconfig.cgi?mac=<MAC>&ip=<IP>
below.
a. Press the selected pattern code, and then press # to finish.

8. Press 1 to save, and then hang up the handset. The device retrieves the configuration
from the HTTP server.
The following is an example perl CGI script, suitable for most Apache-based HTTP servers
for generating configuration dynamically per pattern #6 above. Copy this script to
/var/www/cgi-bin/acconfig.cgi on your Apache server and edit it as required:
#!/usr/bin/perl
use CGI;

MP-11x & MP-124
$query = new CGI;

$mac = $query->param('mac');
$ip = $query->param('ip');
print "Content-type: text/plain\n\n";
print "; INI file generator CGI\n";
print "; Request for MAC=$mac IP=$ip\n\n";
print <<"EOF";
SyslogServerIP = 36.44.0.15
EnableSyslog = 1
SSHServerEnable = 1
EOF
The table below lists the configuration parameters that can be viewed and modified using
the voice menu:
Table 2-2: Voice Menu Configuration Parameters
Item Number at
Description
Menu Prompt
1 IP address.
2 Subnet mask.
3 Default Gateway IP address.
4 Primary DNS server IP address.
7 DHCP enable / disable.
31 Configuration server IP address.
32 Configuration file name pattern.
Voice menu password (initially 12345).
99 Note: The voice menu password can also be changed using the Web interface
or ini file parameter VoiceMenuPassword.

Part II
Management Tools
User's Manual 3. Introduction
3 Introduction
This part provides an overview of the various management tools that can be used to
configure the device. It also provides step-by-step procedures on how to configure the
management settings.
The following management tools can be used to configure the device:
 Embedded HTTP/S-based Web server - see 'Web-based Management' on page 39
 Command Line Interface (CLI) - see 'CLI-Based Management' on page 77
 AudioCodes Element Management System - see EMS-Based Management on page
93
 Simple Network Management Protocol (SNMP) browser software - see 'SNMP-Based
Management' on page 87
 Configuration ini file - see 'INI File-Based Management' on page 95
 TR-069 - see TR-069 Based Management on page 101
Notes:
• Some configuration settings can only be done using a specific management tool.
For example, some configuration can only be done using the Configuration ini file
method.
• Throughout this manual, where a parameter is mentioned, its corresponding Web,
CLI, and ini parameter is mentioned. The ini file parameters are enclosed in
square brackets [...].
• For a list and description of all the configuration parameters, see 'Configuration
Parameters Reference' on page 475.

MP-11x & MP-124

User's Manual 4. Web-Based Management
4 Web-Based Management
The device provides an embedded Web server (hereafter referred to as Web interface),
supporting fault management, configuration, accounting, performance, and security
(FCAPS), including the following:
 Full configuration
 Software and configuration upgrades
 Loading auxiliary files, for example, the Call Progress Tones file
 Real-time, online monitoring of the device, including display of alarms and their
severity
 Performance monitoring of voice calls and various traffic parameters
The Web interface provides a user-friendly, graphical user interface (GUI), which can be
accessed using any standard Web browser (e.g., Microsoft™ Internet Explorer).
Access to the Web interface is controlled by various security mechanisms such as login
user name and password, read-write privileges, and limiting access to specific IP
addresses.
Notes:
• The Web interface allows you to configure most of the device's settings. However,
additional configuration parameters may exist that are not available in the Web
interface and which can only be configured using other management tools.
• Some Web interface pages and/or parameters are available only for certain
hardware configurations or software features. The software features are
determined by the installed Software License Key (see 'Software License Key' on
page 381).
4.1 Getting Acquainted with the Web Interface

This section provides a description of the Web interface.
4.1.1 Computer Requirements

The client computer requires the following to work with the Web interface of the device:
 A network connection to the device
 One of the following Web browsers:
• Microsoft™ Internet Explorer™ (Version 6.0 and later)
• Mozilla Firefox® (Versions 5 through 9.0)
 Recommended screen resolutions: 1024 x 768 pixels, or 1280 x 1024 pixels
Note: Your Web browser must be JavaScript-enabled to access the Web interface.

MP-11x & MP-124
4.1.2 Accessing the Web Interface

The procedure below describes how to access the Web interface.
 To access the Web interface:

1. Open a standard Web browser (see 'Computer Requirements' on page 39).
2. In the Web browser, specify the IP address of the device (e.g., http://10.1.10.10); the
Web interface's Login window appears, as shown below:
Figure 4-1: Web Login Screen
3. In the 'Username' and 'Password' fields, enter the case-sensitive, user name and
password respectively.
4. Click Login; the Web interface is accessed, displaying the Home page. For a detailed
description of the Home page, see 'Viewing the Home Page' on page 63.
Notes:
• The default username and password is "Admin". To change the login user name
and password, see 'Configuring the Web User Accounts' on page 66.
• If you want the Web browser to remember your password, select the 'Remember
Me' check box and then agree to the browser's prompt (depending on your
browser) to save the password for future logins. On your next login attempt, simply
press the Tab or Enter keys to auto-fill the 'Username' and 'Password' fields, and
then click Login.

4.1.3 Areas of the GUI

The areas of the Web interface's GUI are shown in the figure below and described in the
subsequent table.
Figure 4-2: Main Areas of the Web Interface GUI
Table 4-1: Description of the Web GUI Areas
Item # Description
1 Displays AudioCodes (corporate) logo image.

2 Displays the product name.
3 Toolbar, providing frequently required command buttons. For more information, see
'Toolbar Description' on page 42.
4 Displays the username of the Web user that is currently logged in.
5 Navigation bar, providing the following tabs for accessing various functionalities in
the Navigation tree:
 Configuration, Maintenance, and Status & Diagnostics tabs: Access the
configuration menus (see 'Working with Configuration Pages' on page 45)
 Scenarios tab: Creates configuration scenarios (see Working with Scenarios on
page 52)
 Search tab: Enables a search engine for searching configuration parameters (see
'Searching for Configuration Parameters' on page 51)
6 Navigation tree, displaying a tree-like structure of elements (configuration menus,
Scenario steps, or search engine) pertaining to the selected tab on the Navigation
bar. For more information, see 'Navigation Tree' on page 43.
7 Work pane, displaying the configuration page of the selected menu in the Navigation
tree. This is where configuration is done. For more information, see 'Working with
Configuration Pages' on page 45.

MP-11x & MP-124
4.1.4 Toolbar Description

The toolbar provides frequently required command buttons, described in the table below:
Table 4-2: Description of Toolbar Buttons
Icon Button Description

Name
Submit Applies parameter settings to the device (see 'Saving Configuration'

on page 366).
Note: This icon is grayed out when not applicable to the currently
opened page.
Burn Saves parameter settings to flash memory (see 'Saving
Configuration' on page 366).
Device Opens a drop-down list with frequently needed commands:
Actions  Load Configuration File: Opens the Configuration File page for
loading an ini file to the device (see 'Backing Up and Loading
Configuration File' on page 388).
 Save Configuration File: Opens the Configuration File page for
saving the ini file to a folder on a computer (see 'Backing Up and
Loading Configuration File' on page 388).
 Reset: Opens the Maintenance Actions page for performing
various maintenance procedures such as resetting the device
(see 'Resetting the Device' on page 363).
 Software Upgrade Wizard: starts the Software Upgrade wizard
for upgrading the device's software (see 'Software Upgrade
Wizard' on page 385).
Home Opens the Home page (see 'Viewing the Home Page' on page 63).
Help Opens the Online Help topic of the currently opened configuration
page (see 'Getting Help' on page 61).
Log off Logs off a session with the Web interface (see 'Logging Off the Web
Interface' on page 62).
Note: If you modify a parameter that takes effect only after a device reset, after you
click the Submit button in the configuration page, the toolbar displays "Reset", as
shown in the figure below. This is a reminder that you need to later save your settings
to flash memory and reset the device.
Figure 4-3: "Reset" Displayed on Toolbar

4.1.5 Navigation Tree

The Navigation tree is located in the Navigation pane and displays a tree-like structure of
menus pertaining to the selected tab on the Navigation bar. You can drill-down to the
required page item level to open its corresponding page in the Work pane.
The terminology used throughout this manual for referring to the hierarchical structure of
the tree is as follows:
 Menu: first level (highest level)
 Submenu: second level - contained within a menu
 Page item: last level (lowest level in a menu) - contained within a menu or submenu
Figure 4-4: Navigating in Hierarchical Menu Tree (Example)
Note: The figure above is used only as an example. The displayed menus depend
on supported features based on the Software License Key installed on your device.

MP-11x & MP-124
4.1.5.1 Displaying Navigation Tree in Basic and Full View

You can view an expanded or reduced display of the Navigation tree. This affects the
number of displayed menus and submenus in the tree. The expanded (Full) view displays
all the menus pertaining to the selected configuration tab; the reduced (Basic) view
displays only commonly used menus. This is relevant when using the configuration tabs
(i.e., Configuration, Maintenance, and Status & Diagnostics) on the Navigation bar. The
advantage of the Basic view is that it prevents "cluttering" of the Navigation tree with
menus that may not be required.
 To toggle between Full and Basic view:

 To display a reduced menu tree, select the Basic option (default).
 To display all the menus and submenus in the Navigation tree, select the Full option.
Figure 4-5: Basic and Full View Options
Notes:
• After you reset the device, the Web GUI is displayed in Basic view.
• When in Scenario mode (see Scenarios on page 52), the Navigation tree is
displayed in Full view.

4.1.5.2 Showing / Hiding the Navigation Pane

You can hide the Navigation pane to provide more space for elements displayed in the
Work pane. This is especially useful when the Work pane displays a wide table. The arrow
button located below the Navigation bar is used to hide and show the pane.
 To hide and show the Navigation pane:
 To hide the Navigation pane: Click the left-pointing arrow ; the pane is hidden
and the button is replaced by the right-pointing arrow button.
 To show the Navigation pane: Click the right-pointing arrow ; the pane is
displayed and the button is replaced by the left-pointing arrow button.
Figure 4-6: Show and Hide Button (Navigation Pane in Hide View)
4.1.6 Working with Configuration Pages

The configuration pages contain the parameters for configuring the device and are
displayed in the Work pane.
4.1.6.1 Accessing Pages

The configuration pages are accessed by clicking the required page item in the Navigation
tree.
 To open a configuration page:

1. On the Navigation bar, click the required tab (Configuration, Maintenance, or Status
& Diagnostics); the menus pertaining to the selected tab appear in the Navigation
tree.
2. Navigate to the required page item, by performing the following:
• Drill-down using the plus sign to expand the menu and submenus.
• Drill-up using the minus sign to collapse the menu and submenus.
3. Click the required page item; the page opens in the Work pane.
You can also access previously opened pages by clicking the Web browser's Back button
until you have reached the required page. This is useful if you want to view pages in which
you have performed configurations in the current Web session.

MP-11x & MP-124
Notes:
• You can also access certain pages from the Device Actions button located on the
toolbar (see 'Toolbar Description' on page 42).
• To view all the menus in the Navigation tree, ensure that the Navigation tree is in
Full view (see 'Displaying Navigation Tree in Basic and Full View' on page 43).
• To get Online Help for the currently displayed page, see 'Getting Help' on page 61.
• Certain pages may not be accessible or may be read-only, depending on the
access level of your Web user account (see 'Configuring Web User Accounts' on
page 66). If a page is read-only, "Read-Only Mode" is displayed at the bottom of
the page.
4.1.6.2 Viewing Parameters

Some pages allow you to view a reduced or expanded display of parameters. The Web
interface provides two methods for displaying page parameters:
 Displaying "basic" and "advanced" parameters - see 'Displaying Basic and Advanced
Parameters' on page 46
 Displaying parameter groups - see 'Showing / Hiding Parameter Groups' on page 47
4.1.6.2.1 Displaying Basic and Advanced Parameters

Some pages provide a toggle button that allows you to show and hide parameters that
typically are used only in certain deployments. This button is located on the top-right corner
of the page and has two display states:
 Advanced Parameter List button with down-pointing arrow: click this button to
display all parameters.
 Basic Parameter List button with up-pointing arrow: click this button to show only
common (basic) parameters.
The figure below shows an example of a page displaying basic parameters only. If you
click the Advanced Parameter List button (shown below), the page will also display the
advanced parameters.
Figure 4-7: Toggling between Basic and Advanced View

Notes:
• When the Navigation tree is in Full mode (see 'Navigation Tree' on page 43),
configuration pages display all their parameters.
• If a page contains only basic parameters, the Basic Parameter List button is not
displayed.
• If you reset the device, the Web pages display only the basic parameters.
• The basic parameters are displayed in a dark blue background.
4.1.6.2.2 Showing / Hiding Parameter Groups

Some pages provide groups of parameters, which can be hidden or shown. To toggle
between hiding and showing a group, simply click the group title button that appears above
each group. The button appears with a down-pointing or up-pointing arrow, indicating that it
can be collapsed or expanded when clicked, respectively.
Figure 4-8: Expanding and Collapsing Parameter Groups
4.1.6.3 Modifying and Saving Parameters

When you modify a parameter value on a page, the Edit symbol appears to the right of
the parameter. This indicates that the parameter has been modified, but has yet to be
applied (submitted). After you apply your modifications, the symbol disappears.
Figure 4-9: Edit Symbol after Modifying Parameter Value

MP-11x & MP-124
 To save configuration changes on a page to the device's volatile memory (RAM),

do one of the following:
 On the toolbar, click the Submit button.
 At the bottom of the page, click the Submit button.

When you click Submit, modifications to parameters with on-the-fly capabilities are
immediately applied to the device and take effect. Parameters displayed on the page with
the lightning symbol take effect only after a device reset. For resetting the device, see
'Resetting the Device' on page 363.
Note: Parameters saved to the volatile memory (by clicking Submit), revert to their
previous settings after a hardware or software reset, or if the device is powered down.
Therefore, to ensure parameter changes (whether on-the-fly or not) are retained,
save ('burn') them to the device's non-volatile memory, i.e., flash (see 'Saving
If you enter an invalid parameter value (e.g., not in the range of permitted values) and then
click Submit, a message box appears notifying you of the invalid value. In addition, the
parameter value reverts to its previous value and is highlighted in red, as shown in the
figure below:
Figure 4-10: Value Reverts to Previous Valid Value
4.1.6.4 Working with Tables

This section describes how to work with configuration tables, which are provided in basic or
enhanced design, depending on the configuration page.

4.1.6.4.1 Basic Design Tables

A few of the tables in the Web interface are in basic design format. The figure below
displays a typical table in the basic design format and the subsequent table describes its
command buttons.
Figure 4-11: Adding an Index Entry to a Table
Table 4-3: Basic Table Design Description
Item # Button / Field
1 Add Index (or Add) button Adds an index entry row to the table.
2 Edit Edits the selected row.
3 Delete Removes the selected row from the table.
4 'Add Index' field Defines the index number. When adding a new row, enter
the required index number in this field, and then click Add
Index.
5 Index radio button Selects the row for editing and deleting.
- Compact button Organizes the index entries in ascending, consecutive
order, starting from index 0. For example, assume you
have three index entries, 0, 4 and 6. After you click
Compact, index entry 4 is re-assigned to index 1 and index
entry 6 is re-assigned to index 2.
- Apply button Saves the row configuration. Click this button after you add
or edit each index entry.

MP-11x & MP-124
4.1.6.4.2 Enhanced Design Tables

Most of the tables in the Web interface are designed in the enhanced table format. The
figure below displays a typical table in the enhanced design format and the subsequent
table describes its command buttons and areas.
Figure 4-12: Displayed Details Pane
Table 4-4: Enhanced Table Design Description
Item # Button
1 Add Adds a new index entry row to the table. When you click this button, a
dialog box appears with parameters for configuring the new entry.
When you have completed configuration, click the Submit button in
the dialog box to add it to the table.
2 Edit Edits the selected row.
3 Delete Removes the selected row from the table. When you click this button,
a confirmation box appears requesting you to confirm deletion. Click
Delete to accept deletion.
4 Show/Hide Toggles between displaying and hiding the full configuration of a

selected row. This configuration is displayed below the table (see Item
#6) and is useful for large tables that cannot display all its columns in
the work pane.
5 - Selected index row entry for editing, deleting and showing
configuration.
6 - Displays the full configuration of the selected row when you click the
Show/Hide button.
7 - Links to access additional configuration tables related to the current
configuration.

If the configuration of an entry row is invalid, the index of the row is highlighted in red, as
shown below:
Figure 4-13: Invalid Configuration with Index Highlighted in Red
The table also enables you to define the number of rows to display on the page and to
navigate between pages displaying multiple rows. This is done using the page navigation
area located below the table, as shown in the figure below:
Figure 4-14: Viewing Table Rows per Page
Table 4-5: Row Display and Page Navigation
Item # Description
1 Defines the page that you want to view. Enter the required page number or use the
following page navigation buttons:
 - Displays the next page
 - Displays the last page
 - Displays the previous page
 - Displays the first page
2 Defines the number of rows to display per page. You can select 5 or 10, where the
default is 10.
3 Displays the currently displayed page number.
4.1.7 Searching for Configuration Parameters

You can locate the exact Web page on which a specific parameter appears, by using the
device's Search feature. The Web parameter's corresponding ini file parameter name is
used as the search key. The search key can include the full parameter name (e.g.,
"EnableIPSec") or a substring of it (e.g., "sec"). If you search for a substring, all parameters
containing the specified substring in their names are listed in the search result.
Note: If an ini file parameter is not configurable in the Web interface, the search
fails.
 To search for a parameter:

1. On the Navigation bar, click the Search tab; the Search engine appears in the

MP-11x & MP-124
Navigation pane.
2. In the field alongside the Search button, enter the parameter name or a substring of
the name for which you want to search. If you have done a previous search for such a
parameter, instead of entering the required string, you can use the 'Search History'
drop-down list to select the string saved from a previous search.
3. Click Search; a list of found parameters based on your search key appears in the
Navigation pane. Each searched result displays the following:
• ini file parameter name
• Link (in green) to the Web page on which the parameter appears
• Brief description of the parameter
• Menu navigation path to the Web page on which the parameter appears
4. In the searched list, click the required parameter (green link) to open the page on
which the parameter appears; the relevant page opens in the Work pane and the
searched parameter is highlighted in the page for easy identification, as shown in the
figure below:
Figure 4-15: Searched Result Screen
Table 4-6: Search Description
Item # Description
1 Search field for entering search key and Search button for activating the search
process.
2 Search results listed in Navigation pane.
3 Found parameter, highlighted on relevant Web page
4.1.8 Working with Scenarios

The Web interface allows you to create your own menu (Scenario) of up to 20 pages,
selected from the menus in the Navigation tree (i.e., pertaining to the Configuration,
Maintenance, and Status & Diagnostics tabs). Each page in the Scenario is referred to
as a Step. For each Step, you can select up to 25 parameters on the page to include in the
Scenario. Therefore, the Scenario feature is useful in that it allows you quick-and-easy
access to commonly used configuration parameters specific to your network environment.
When you log in to the Web interface, your Scenario is displayed in the Navigation tree.
Instead of creating a new Scenario, you can load a saved Scenario on a computer to the
device (see 'Loading a Scenario to the Device' on page 57).

4.1.8.1 Creating a Scenario

The procedure below describes how to create a Scenario.
 To create a Scenario:
1. On the Navigation bar, click the Scenarios tab; a message box appears, requesting
you to confirm creation of a Scenario:
Figure 4-16: Create Scenario Confirmation Message Box
Note: If a Scenario already exists, the Scenario Loading message box appears.
2. Click OK; the Scenario mode appears in the Navigation tree as well as the menus of
the Configuration tab.
3. In the 'Scenario Name' field, enter an arbitrary name for the Scenario.
4. On the Navigation bar, click the Configuration or Maintenance tab to display their
respective menus in the Navigation tree.
5. In the Navigation tree, select the required page item for the Step, and then in the page
itself, select the required parameters by selecting the check boxes corresponding to
the parameters.
6. In the 'Step Name' field, enter a name for the Step.
7. Click the Next button located at the bottom of the page; the Step is added to the
Scenario and appears in the Scenario Step list.
8. Repeat steps 5 through 7 to add additional Steps (i.e., pages).
9. When you have added all the required Steps for your Scenario, click the Save &
Finish button located at the bottom of the Navigation tree; a message box appears
informing you that the Scenario has been successfully created.

MP-11x & MP-124
10. Click OK; the Scenario mode is quit and the menu tree of the Configuration tab
appears in the Navigation tree.
Figure 4-17: Creating a Scenario
Table 4-7: Scenario Description
Description
Selected page item in the Navigation tree whose page contains the parameter that you
1
want to add to the Scenario Step.
2 Name of a Step that has been added to the Scenario.
3 'Scenario Name' field for defining a name for the Scenario.
4 'Step Name' field for defining a name for a Scenario Step.
5 Save & Finish button to save your Scenario.
6 Selected parameter(s) that you want added to a Scenario Step.
Next button to add the current Step to the Scenario and enables you to add additional
7
Steps.
Notes:
• You can add up to 20 Steps per Scenario, where each Step can contain up to 25
parameters.
• When in Scenario mode, the Navigation tree is in 'Full' display (i.e., all menus are
displayed in the Navigation tree) and the configuration pages are in 'Advanced
Parameter List' display (i.e., all parameters are shown in the pages). This ensures
accessibility to all parameters when creating a Scenario. For a description on the
Navigation tree views, see 'Navigation Tree' on page 43.
• If you previously created a Scenario and you click the Create Scenario button, the
previously created Scenario is deleted and replaced with the one you are creating.
• Only Security Administrator Web users can create Scenarios.

4.1.8.2 Accessing a Scenario

Once you have created the Scenario, you can access it by following the procedure below:
 To access the Scenario:

1. On the Navigation bar, select the Scenario tab; a message box appears, requesting
you to confirm the loading of the Scenario.
2. Click OK; the Scenario and its Steps appear in the Navigation tree, as shown in the
example below:
Figure 4-18: Scenario Example
Table 4-8: Loaded Scenario Description
Item Description
1 Scenario name.
2 Scenario Steps.
3 Scenario configuration command buttons.
4 Parameters available on a page for the selected Scenario Step. These are displayed in
a blue background; unavailable parameters are displayed in a gray or light-blue
background.
5 Navigation buttons for navigating between Scenario Steps:
 Next button to open the next Step listed in the Scenario
 Previous button to open the previous Step listed in the Scenario
Note: If you reset the device while in Scenario mode, after the device resets, you are
returned once again to the Scenario mode.

MP-11x & MP-124
4.1.8.3 Editing a Scenario

You can modify a Scenario as described in the procedure below.
Note: Only Security Administrator Web users can edit a Scenario.
 To edit a Scenario:
1. Open the Scenario.
2. Click the Edit Scenario button located at the bottom of the Navigation pane; the
'Scenario Name' and 'Step Name' fields appear.
3. You can perform the following edit operations:
• Add Steps:
a. On the Navigation bar, select the desired tab (i.e., Configuration or
Maintenance); the tab's menu appears in the Navigation tree.
b. In the Navigation tree, navigate to the desired page item; the corresponding
page opens in the Work pane.
c. On the page, select the required parameters by marking their corresponding
check boxes.
d. Click Next.
• Add or Remove Parameters:
a. In the Navigation tree, select the required Step; the corresponding page
opens in the Work pane.
b. To add parameters, select the check boxes corresponding to the desired
parameters.
c. To remove parameters, clear the check boxes corresponding to the desired
parameters.
d. Click Next.
• Edit Step Name:
a. In the Navigation tree, select the required Step.
b. In the 'Step Name' field, modify the Step name.
c. On the page, click Next.
• Edit Scenario Name:
a. In the 'Scenario Name' field, edit the Scenario name.
b. On the displayed page, click Next.
• Remove a Step:
a. In the Navigation tree, select the required Step; the corresponding page
opens in the Work pane.
b. On the page, clear all the check boxes corresponding to the parameters.
c. Click Next.
4. After clicking Next, a message box appears notifying you of the change. Click OK.
5. Click Save & Finish; a message box appears informing you that the Scenario has
been successfully modified. The Scenario mode is exited and the menus of the
Configuration tab appear in the Navigation tree.

4.1.8.4 Saving a Scenario to a PC

You can save a Scenario (as a dat file) to a folder on your computer. This is useful when
you need multiple Scenarios to represent different deployments. Once you create a
Scenario and save it to your computer, you can then keep on saving modifications to it
under different Scenario file names. When you require a specific network environment
setup, you can load the suitable Scenario file from your computer (see 'Loading a Scenario
to the Device' on page 57).
 To save a Scenario to a computer:

1. On the Navigation bar, click the Scenarios tab; the Scenario appears in the
Navigation tree.
2. Click the Get/Send Scenario File button, located at the bottom of the Navigation tree;
the Scenario File page appears, as shown below:
Figure 4-19: Scenario File Page
3. Click the Get Scenario File button; the File Download window appears.
4. Click Save, and then in the Save As window navigate to the folder to where you want
to save the Scenario file. When the file is successfully downloaded to your computer,
the Download Complete window appears.
5. Click Close to close the window.
4.1.8.5 Loading a Scenario to the Device

The procedure below describes how to load a previously saved Scenario file (data file)
from your computer to the device. For saving a Scenario, see 'Saving a Scenario to a PC'
on page 57.
 To load a Scenario to the device:

1. On the Navigation bar, click the Scenarios tab; the Scenario appears in the
Navigation tree.
2. Click the Get/Send Scenario File button, located at the bottom of the Navigation tree;
the Scenario File page appears.
3. Click the Browse button, and then navigate to the Scenario file saved on your
computer.

MP-11x & MP-124
4. Click the Send File button.
Notes:
• You can only load a Scenario file to a device that has the same hardware
configuration as the device on which it was created.
• The loaded Scenario replaces any existing Scenario.
• You can also load a Scenario file using BootP, by loading an ini file that contains
the ini file parameter ScenarioFileName (see Web and Telnet Parameters on page
486). The Scenario file must be located in the same folder as the ini file. For
information on using AudioCodes AcBootP utility, refer to AcBootP Utility User's
Guide.
4.1.8.6 Deleting a Scenario

You can delete the Scenario, as described in the procedure below.
 To delete the Scenario:

1. On the Navigation bar, click the Scenarios tab; a message box appears, requesting
you to confirm:
2. Click OK; the Scenario mode appears in the Navigation tree.
3. Click the Delete Scenario File button; a message box appears requesting
confirmation for deletion.
Figure 4-20: Message Box for Confirming Scenario Deletion

4. Click OK; the Scenario is deleted and the Scenario mode closes.
Note: You can also delete a Scenario using the following alternative methods:
• Loading an empty dat file (see 'Loading a Scenario to the Device' on page 57).
• Loading an ini file with the ScenarioFileName parameter set to no value (i.e.,
ScenarioFileName = "").
4.1.8.7 Quitting Scenario Mode

Follow the procedure below to quit the Scenario mode.
 To quit the Scenario mode:

1. On the Navigation bar, click any tab except the Scenarios tab, or click the Cancel
Scenarios button located at the bottom of the Navigation tree; a message box
appears, requesting you to confirm exiting Scenario mode, as shown below.
Figure 4-21: Confirmation Message Box for Exiting Scenario Mode
2. Click OK to exit.

MP-11x & MP-124
4.1.9 Creating a Login Welcome Message

You can create a Welcome message box that is displayed on the Web Login page for
logging in to the Web interface. The figure below displays an example of a Welcome
message:
Figure 4-22: User-Defined Web Welcome Message after Login
To enable and create a Welcome message, use the WelcomeMessage table ini file
parameter. If this parameter is not configured, no Welcome message is displayed.
Table 4-9: ini File Parameter for Welcome Login Message
Parameter Description
[WelcomeMessage] Enables and defines a Welcome message that appears on the Web Login
page for logging in to the Web interface.
The format of this parameter is as follows:
[WelcomeMessage]
FORMAT WelcomeMessage_Index = WelcomeMessage_Text;
[\WelcomeMessage]
For Example:
[WelcomeMessage ]
FORMAT WelcomeMessage_Index = WelcomeMessage_Text;
WelcomeMessage 1 = "*********************************";
WelcomeMessage 2 = "********* This is a Welcome message **";
WelcomeMessage 3 = "*********************************";
[\WelcomeMessage]
Each index row represents a line of text in the Welcome message box. Up
to 20 lines (or rows) of text can be defined.

4.1.10 Getting Help

The Web interface provides you with context-sensitive Online Help. The Online Help
provides brief descriptions of parameters pertaining to the currently opened page.
 To view the Help topic of a currently opened page:
1. On the toolbar, click the Help button; the Help topic pertaining to the opened
page appears, as shown below:
Figure 4-23: Help Topic for Current Page
2. To view a description of a parameter, click the plus sign to expand the parameter.
To collapse the description, click the minus sign.
3. To close the Help topic, click the close button located on the top-right corner of
the Help topic window or simply click the Help button.
Note: Instead of clicking the Help button for each page you open, you can open it
once for a page and then simply leave it open. Each time you open a different page,
the Help topic pertaining to that page is automatically displayed.

MP-11x & MP-124
4.1.11 Logging Off the Web Interface

The procedure below describes how to log off the Web interface.
 To log off the Web interface:
1. On the toolbar, click the Log Off icon; the following confirmation message box
appears:
Figure 4-24: Log Off Confirmation Box
2. Click OK; you are logged off the Web session and the Web Login dialog box appears
enabling you to re-login, if required.

4.2 Viewing the Home Page

The Home page is displayed when you access the device's Web interface. The Home page
provides you with a graphical display of the device's front panel, showing color-coded
status icons for various operations device.
 To access the Home page:
 On the toolbar, click the Home icon.

Figure 4-25: MP-11x Home Page
Figure 4-26: MP-124 Home Page
Note: The displayed number and type (FXO and/or FXS) of channels depends on
the ordered model (e.g., MP-118 or MP-114).
In addition to the color-coded status information depicted on the graphical display of the
device, the Home page displays various read-only information in the General Information
pane:
 IP Address: IP address of the device
 Subnet Mask: Subnet mask address of the device
 Default Gateway Address: Default gateway used by the device
 Firmware Version: Software version running on the device
 Protocol Type: Signaling protocol currently used by the device (i.e. SIP)
 Gateway Operational State:
• "LOCKED": device is locked (i.e. no new calls are accepted)
• "UNLOCKED": device is not locked
• "SHUTTING DOWN": device is currently shutting down
To perform these operations, see 'Basic Maintenance' on page 363.
The table below describes the areas of the Home page.

MP-11x & MP-124
Table 4-10: Home Page Description
Label Description
Alarms Displays the highest severity of an active alarm raised (if any) by the device:
 Green = no alarms
 Red = Critical alarm
 Orange = Major alarm
 Yellow = Minor alarm
To view active alarms, click this Alarms area to open the Active Alarms page (see
Viewing Active Alarms on page 413).
Channel/Ports Displays the status of the ports (channels):
 (red): Line not connected or port out of service due to Serial Peripheral
Interface (SPI) failure (applicable only to FXO interfaces)
 (grey): Channel inactive
 (blue): Handset is off-hook
 (green): Active RTP stream
If you click a port, a shortcut menu appears with commands allowing you to
perform the following:
 (Analog ports only) Reset the channel port (see Resetting an Analog Channel
on page 367)
 View the port settings (see 'Viewing Analog Port Information' on page 415)
 Assign a name to the port (see 'Assigning a Port Name' on page 65)
Uplink (MP-11x) If clicked, the Ethernet Port Information page opens, displaying Ethernet port
LAN (MP-124 configuration settings (see Viewing Ethernet Port Information on page 411).
Fail Currently not supported.

Ready Currently not supported.
Power Always lit green, indicating power received by the device.

4.2.1 Assigning a Port Name

The Home page allows you to assign an arbitrary name or a brief description to each port.
This description appears as a tooltip when you move your mouse over the port.
Note: Only alphanumerical characters can be used in the port description.
 To add a port description:

1. Click the required port icon; a shortcut menu appears, as shown below:
Figure 4-27: Shortcut Menu (Example using MP-11x)
2. From the shortcut menu, choose Update Port Info; a text box appears.
Figure 4-28: Typing in Port Name (Example using MP-11x)
3. Type a brief description for the port, and then click Apply Port Info.

MP-11x & MP-124
4.3 Configuring Web User Accounts

You can create up to 5 Web user accounts for the device. Up to five Web users can
simultaneously be logged in to the device's Web interface. Web user accounts prevent
unauthorized access to the Web interface, enabling login access only to users with correct
credentials (i.e., username and password). Each Web user account is composed of the
following attributes:
 Username and password: Credentials that enable authorized login access to the
Web interface.
 Access level (user type): Access privileges specifying what the user can view in the
Web interface and its read/write privileges. The table below describes the different
types of Web user account access levels:
Table 4-11: Access Levels of Web User Accounts
Numeric
User Access Level Privileges
Representation*
Master Read / write privileges for all pages. Can create all user
types, including additional Master users and Security
220
Administrators. It can delete all users except the last
Security Administrator.
Security Read / write privileges for all pages. It can create all user
Administrator types and is the only one that can create the first Master
200 user.
Note: There must be at least one Security Administrator.
Administrator Read / write privileges for all pages except security-
100
related pages, which are read-only.
Monitor No access to security-related and file-loading pages;
50
read-only access to other pages.
No Access No access to any page.
0 Note: This access level is not applicable when using
advanced Web user account configuration in the Web
Users table.
* The numeric representation of the access level is used only to define accounts in a RADIUS server
(the access level ranges from 1 to 255).
By default, the device is pre-configured with the following two Web user accounts:
Table 4-12: Pre-configured Web User Accounts
User Access Level Username Password

(Case-Sensitive) (Case-Sensitive)
Security Administrator Admin Admin

Monitor User User

After you log in to the Web interface, the username is displayed on the toolbar.
If the Web session is idle (i.e., no actions are performed) for more than five minutes, the
Web session expires and you are once again requested to login with your username and
password. Users can be banned for a period of time upon a user-defined number of
unsuccessful login attempts. Login information (such as how many login attempts were
made and the last successful login time) can be presented to the user.
 To prevent user access after a specific number of failed logins:

1. From the 'Deny Access On Fail Count' drop-down list, select the number of failed
logins after which the user is prevented access to the device for a user-defined time
(see next step).
2. In the 'Deny Authentication Timer' field, enter the interval (in seconds) that the user
needs to wait before a new login attempt from the same IP address can be done after
reaching the number of failed login attempts (defined in the previous step).
Notes:
• For security, it's recommended that you change the default username and
password.
• The Security Administrator user can change all attributes of all Web user
accounts. Web users with access levels other than Security Administrator can
change only their password and username.
• To restore the two Web user accounts to default settings (usernames and
passwords), set the ini file parameter ResetWebPassword to 1.
• To log in to the Web interface with a different Web user, click the Log off button
and then login with with a different username and password.
• You can set the entire Web interface to read-only (regardless of Web user access
levels), by using the ini file parameter DisableWebConfig (see 'Web and Telnet
Parameters' on page 486).
• You can define additional Web user accounts using a RADIUS server (see
'Configuring RADIUS Settings' on page 76).
4.3.1 Basic User Accounts Configuration

This section describes basic Web user account configuration. This is relevant only if the
two default, pre-configured Web user accounts - Security Administrator ("Admin") and
Monitor ("User") - are sufficient for your management scheme.
For the Security Administrator, you can change only the username and password; not its
access level. For the Monitor user, you can change username and password as well as
access level (Administrator, Monitor, or No Access).
Notes:
• The access level of the Security Administrator cannot be modified.
• The access level of the second user account can be modified only by the Security
Administrator.
• The username and password can be a string of up to 19 characters. When you log
in to the Web interface, the username and password string values are case-
sensitive, according to your configuration.
• Up to two users can be logged in to the Web interface at the same time, and they
can be of the same user.

MP-11x & MP-124
 To configure the two pre-configured Web user accounts:

1. Open the Web User Accounts page (Configuration tab > System menu > Web User
Accounts). If you are logged in as Security Administrator, both Web user accounts
are displayed (as shown below). If you are logged in with the second user account,
only the details of this user account are displayed.
Figure 4-29: WEB User Accounts Page (for Users with 'Security Administrator' Privileges)
2. To change the username of an account:

a. In the 'User Name' field, enter the new user name.
b. Click Change User Name; if you are currently logged in to the Web interface with
this account, the 'Web Login' dialog box appears.
c. Log in with your new user name.
3. To change the password of an account:
a. In the 'Current Password' field, enter the current password.
b. In the 'New Password' and 'Confirm New Password' fields, enter the new
password.
c. Click Change Password; if you are currently logged in to the Web interface with
this account, the 'Web Login' dialog box appears.
d. Log in with your new password.
4. To change the access level of the optional, second account:
a. Under the Account Data for User: User group, from the 'Access Level' drop-
down list, select a new access level user.
b. Click Change Access Level; the new access level is applied immediately.

4.3.2 Advanced User Accounts Configuration

This section describes advanced Web user account configuration. This is relevant if you
need the following management scheme:
 Enhanced security settings per Web user (e.g., limit session duration)
 More than two Web user accounts (up to 5 Web user accounts)
 Master users
This advanced Web user configuration is done in the Web Users table, which is initially
accessed from the Web User Accounts page (see procedure below). Once this table is
accessed, subsequent access immediately opens the Web Users table instead of the Web
User Accounts page.
Notes:
• Only the Security Administrator user can initially access the Web Users table.
• Only Security Administrator and Master users can add, edit, or delete users.
• Admin users have read-only privileges in the Web Users table. Monitor users have
no access to this page.
• If you delete a user who is currently in an active Web session, the user is
immediately logged off by the device.
• All users can change their own passwords. This is done in the WEB Security
Settings page (see 'Configuring Web Security Settings' on page 73).
• To remove the Web Users table and revert to the Web User Accounts page with
the pre-configured, default Web user accounts, set the ResetWebPassword ini file
parameter to 1. This also deletes all other Web users.
• Once the Web Users table is accessed, Monitor users and Admin users can only
change their passwords in the Web Security Settings page (see 'Configuring Web
Security Settings' on page 73). The new password must have at least four different
characters than the previous password. (The Security Administrator users and
Master users can change their passwords in the Web Users table and in the Web
Security Settings page.)
• This table can only be configured using the Web interface.
 To add Web user accounts with advanced settings:

1. Open the Web Users Table page:
• Upon initial access:
a. Open the Web User Accounts page (Configuration tab > System menu >
Web User Accounts).
b. Under the Web Users Table group, click the Create Table button.
• Subsequent access: Configuration tab > System menu > Web User Accounts.
The Web Users table appears, listing the two default, pre-configured Web use
accounts - Security Administrator ("Admin") and Monitor ("User"):
Figure 4-30: Web Users Table Page

MP-11x & MP-124
2. Click the Add button; the following dialog box is displayed:

Figure 4-31: Web Users Table - Add Record Dialog Box
3. Add a user as required. For a description of the parameters, see the table below.
4. Click Submit.
Table 4-13: Web User Parameters Description
Web: Username Defines the Web user's username.

[WebUsers_Username] The valid value is a string of up to 40 alphanumeric characters,
including the period ".", underscore "_", and hyphen "-" signs.
Web: Password Defines the Web user's password.
[WebUsers_Password] The valid value is a string of 8 to 40 ASCII characters, which must
include the following:
 At least eight characters
 At least two letters that are upper case (e.g., "AA")
 At least two letters that are lower case (e.g., "aa")
 At least two numbers
 At least two signs (e.g., the dollar "$" sign)
 No spaces in the string
 At least four characters different to the previous password
Note: For security, password characters are not shown in the Web
interface and ini file. In the Web interface, they are displayed as dots
when you enter the password and then once applied, the password is
displayed as an asterisk (*) in the table. In the ini file, they are
displayed as an encrypted string.

Web: Status Defines the status of the Web user.

[WebUsers_Status]  New = (Default) User is required to change its password on the
next login. When the user logs in to the Web interface, the user is
immediately prompted to change the current password.
 Valid = User can log in to the Web interface as normal.
 Failed Access = This state is automatically set for users that
exceed a user-defined number of failed login attempts, set by the
'Deny Access on Fail Count' parameter (see 'Configuring Web
Security Settings' on page 73). These users can log in only after a
user-defined timeout configured by the 'Block Duration' parameter
(see below) or if their status is changed (to New or Valid) by a
System Administrator or Master.
 Old Account = This state is automatically set for users that have
not accessed the Web interface for a user-defined number of days,
set by the 'User Inactivity Timer' (see 'Configuring Web Security
Settings' on page 73). These users can only log in to the Web
interface if their status is changed (to New or Valid) by a System
Administrator or Master.
Notes:
 The Old Account status is applicable only to Admin and Monitor
users; System Administrator and Master users can be inactive
indefinitely.
 For security, it is recommended to set the status of a newly added
user to New in order to enforce password change.
Web: Password Age Defines the duration (in days) of the validity of the password. When
[WebUsers_PwAgeInterval] this duration elapses, the user is prompted to change the password;
otherwise, access to the Web interface is blocked.
The valid value is 0 to 10000, where 0 means that the password is
always valid. The default is 90.
Web: Session Limit Defines the maximum number of Web interface sessions allowed for
[WebUsers_SessionLimit] the user. In other words, this allows the same user account to log in to
the device from different sources (i.e., IP addresses).
The valid value is 0 to 5. The default is 2.
Note: Up to 5 users can be concurrently logged in to the Web
interface.
Web: Session Timeout Defines the duration (in minutes) of Web inactivity of a logged-in user,
[WebUsers_SessionTimeout] after which the user is automatically logged off the Web interface.
The valid value is 0 to 100000. The default is according to the settings
of the 'Session Timeout' global parameter (see 'Configuring Web
Security Settings' on page 73).
Web: Block Duration Defines the duration (in seconds) for which the user is blocked when
[WebUsers_BlockTime] the user exceeds a user-defined number of failed login attempts. This
is configured by the 'Deny Access On Fail Count' parameter (see
'Configuring Web Security Settings' on page 73).
The valid value is 0 to 100000, where 0 means that the user can do as
many login failures without getting blocked. The default is according to
the settings of the 'Deny Authentication Timer' parameter (see
'Configuring Web Security Settings' on page 73).
Note: The 'Deny Authentication Timer' parameter relates to failed
Web logins from specific IP addresses.

MP-11x & MP-124
Web: User Level Defines the user's access level.

[WebUsers_UserLevel]  Monitor = (Default) Read-only user. This user can only view Web
pages and access to security-related pages is denied.
 Admin = Read/write privileges for all pages, except security-related
pages including the Web Users table where this user has only
read-only privileges.
 SecAdmin = Read/write privileges for all pages. This user is the
Security Administrator.
 Master-User = Read/write privileges for all pages. This user also
functions as a security administrator.
Notes:
 At least one Security Administrator must exist. The last remaining
Security Administrator cannot be deleted.
 The first Master user can be added only by a Security
Administrator user.
 Additional Master users can be added, edited and deleted only by
Master users.
 If only one Master user exists, it can be deleted only by itself.
 Master users can add, edit, and delete Security Administrators (but
cannot delete the last Security Administrator).
 Only Security Administrator and Master users can add, edit, and
delete Admin and Monitor users.

4.4 Displaying Login Information upon Login

The device can display login information immediately upon Web login.
 To enable display of user login information upon a successful login:

1. Open the WEB Security Settings page (Configuration tab > System menu >
Management submenu > WEB Security Settings).
2. From the 'Display Login Information' drop-down list, select Yes.
3. Click Submit to apply your changes.
Once enabled, the Login Information window is displayed upon a successful login, as
shown in the example below:
Figure 4-32: Login Information Window
4.5 Configuring Web Security Settings

The WEB Security Settings page is used to define a secure Web access communication
method. For a description of these parameters, see 'Web and Telnet Parameters' on page
486.
 To define Web access security:

1. Open the WEB Security Settings page (Configuration tab > System menu >
Management submenu > WEB Security Settings).

MP-11x & MP-124
2. Configure the parameters as required.

4. To save the changes to flash memory, see 'Saving Configuration' on page 366.
4.6 Web Login Authentication using Smart Cards

You can enable Web login authentication using certificates from a third-party, common
access card (CAC) with user identification. When a user attempts to access the device
through the Web browser (HTTPS), the device retrieves the Web user’s login username
(and other information, if required) from the CAC. The user attempting to access the device
is only required to provide the login password. Typically, a TLS connection is established
between the CAC and the device’s Web interface, and a RADIUS server is implemented to
authenticate the password with the username. Therefore, this feature implements a two-
factor authentication - what the user has (i.e., the physical card) and what the user knows
(i.e., the login password).
This feature is enabled using the EnableMgmtTwoFactorAuthentication parameter.
Note: For specific integration requirements for implementing a third-party smart card
for Web login authentication, contact your AudioCodes representative.
 To log in to the Web interface using CAC:

1. Insert the Common Access Card into the card reader.
2. Access the device using the following URL: https://<host name or IP address>; the
device prompts for a username and password.
3. Enter the password only. As some browsers require that the username be provided,
it’s recommended to enter the username with an arbitrary value.

4.7 Configuring Web and Telnet Access List

The Web & Telnet Access List page is used to define IP addresses (up to ten) that are
permitted to access the device's Web, Telnet, and SSH interfaces. Access from an
undefined IP address is denied. If no IP addresses are defined, this security feature is
inactive and the device can be accessed from any IP address. The Web and Telnet Access
List can also be defined using the ini file parameter WebAccessList_x (see 'Web and
Telnet Parameters' on page 486).
 To add authorized IP addresses for Web, Telnet, and SSH interfaces access:
1. Open the Web & Telnet Access List page (Configuration tab > System menu >
Management submenu > Web & Telnet Access List).
Figure 4-33: Web & Telnet Access List Page - Add New Entry
2. To add an authorized IP address, in the 'Add an authorized IP address' field, enter the
required IP address, and then click Add New Entry; the IP address you entered is
added as a new entry to the Web & Telnet Access List table.
Figure 4-34: Web & Telnet Access List Table

MP-11x & MP-124
3. To delete authorized IP addresses, select the Delete Row check boxes corresponding
to the IP addresses that you want to delete, and then click Delete Selected
Addresses; the IP addresses are removed from the table and these IP addresses can
no longer access the Web and Telnet interfaces.
Notes:
• The first authorized IP address in the list must be your PC's (terminal) IP address;
otherwise, access from your PC is denied.
• Delete your PC's IP address last from the 'Web & Telnet Access List page. If it is
deleted before the last, subsequent access to the device from your PC is denied.
4.8 Configuring RADIUS Settings

The RADIUS Settings page is used for configuring the Remote Authentication Dial In User
Service (RADIUS) accounting parameters. For a description of these parameters, see
'Configuration Parameters Reference' on page 475.
 To configure RADIUS:
1. Open the RADIUS Settings page (Configuration tab > System menu > Management
submenu > RADIUS Settings).
Figure 4-35: RADIUS Parameters Page


User's Manual 5. CLI-Based Management
5 CLI-Based Management
This section provides an overview of the CLI-based management and configuration relating
to CLI management. The device's CLI-based management interface can be accessed
using the RS-232 serial port or by using Secure SHell (SSH) or Telnet through the Ethernet
interface.
Warning: If you are using the PuTTY terminal emulator for CLI, you must enable the
use of the backspace key in the CLI; otherwise, an error will be generated and your
settings will not be applied. To enable backspace functionality, start PuTTY and then
in the PuTTY Configuration window, expand the Terminal folder, click Keyboard, and
then select the Control-H option under the 'The Backspace key' group.
Notes:
• For security, CLI is disabled by default.
• For information on accessing the CLI interface through the RS-232 port interface,
see 'CLI' on page 30.
• CLI is used only for debugging and mainly allows you to view various information
regarding device configuration and performance.
5.1 Enabling CLI using Telnet

The device's CLI can be accessed using Telnet. Secure Telnet using Secure Socket Layer
(SSL) can be configured whereby information is not transmitted in the clear. If SSL is used,
a special Telnet client is required on your PC to connect to the Telnet interface over a
secured connection; examples include C-Kermit for UNIX and Kermit-95 for Windows.
For security, some organizations require the display of a proprietary notice upon starting a
Telnet session. You can use the configuration ini file parameter, WelcomeMessage to
configure such a message (see Creating a Login Welcome Message on page 60).
 To enable Telnet:
1. Open the Telnet/SSH Settings page (Configuration tab > System menu >
Management > Telnet/SSH Settings).
Figure 5-1: Telnet Settings on Telnet/SSH Settings Page
2. Set the ‘Embedded Telnet Server’ parameter to Enable Unsecured or Enable

Secured (i.e, SSL).
3. Configure the other Tenet parameters as required. For a description of these
parameters, see Telnet Parameters on page 489.
4. Click Submit.
5. Save the changes to flash memory with a device reset.

MP-11x & MP-124
5.2 Enabling CLI using SSH and RSA Public Key

The device's CLI can be accessed using Telnet. However, unless configured for TLS,
Telnet is not secure as it requires passwords to be transmitted in clear text. To overcome
this, Secure SHell (SSH) is used, which is the de-facto standard for secure CLI. SSH 2.0 is
a protocol built above TCP, providing methods for key exchange, authentication,
encryption, and authorization.
SSH requires appropriate client software for the management PC. Most Linux distributions
have OpenSSH pre-installed; Windows-based PCs require an SSH client software such as
PuTTY, which can be downloaded from
http://www.chiark.greenend.org.uk/~sgtatham/putty/.
By default, SSH uses the same username and password as the Telnet and Web server.
SSH supports 1024/2048-bit RSA public keys, providing carrier-grade security. Follow the
instructions below to configure the device with an administrator RSA key as a means of
strong authentication.
 To enable SSH and configure RSA public keys for Windows (using PuTTY SSH):
1. Start the PuTTY Key Generator program, and then do the following:
a. Under the 'Parameters' group, do the following:
♦ Select the SSH-2 RSA option.
♦ In the 'Number of bits in a generated key' field, enter "1024" bits.
b. Under the 'Actions' group, click Generate and then follow the on-screen
instructions.
c. Under the 'Actions' group, click Save private key to save the new private key to a
file (*.ppk) on your PC.
d. Under the 'Key' group, select the displayed encoded text between "ssh-rsa" and
"rsa-key-….", as shown in the example below:
Figure 5-2: Selecting Public RSA Key in PuTTY
Management > Telnet/SSH Settings), and then do the following:
a. Set the 'Enable SSH Server' parameter to Enable.
b. Paste the public key that you copied in Step 1.d into the 'Admin Key' field, as
shown below:

Figure 5-3: SSH Settings - Pasting Public RSA Key in 'Admin Key' Field
c. For additional security, you can set the 'Require Public Key' to Enable. This
ensures that SSH access is only possible by using the RSA key and not by using
user name and password.
d. Configure the other SSH parameters as required. For a description of these
parameters, see SSH Parameters on page 511.
e. Click Submit.
3. Start the PuTTY Configuration program, and then do the following:
a. In the 'Category' tree, drill down to Connection, then SSH, and then Auth; the
'Options controlling SSH authentication' pane appears.
b. Under the 'Authentication parameters' group, click Browse and then locate the
private key file that you created and saved in Step 4.
4. Connect to the device with SSH using the username "Admin"; RSA key negotiation
occurs automatically and no password is required.
 To configure RSA public keys for Linux (using OpenSSH 4.3):

1. Run the following command to create a new key in the admin.key file and to save the
public portion to the admin.key.pub file:
ssh-keygen -f admin.key -N "" -b 1024
2. Open the admin.key.pub file, and then copy the encoded string from "ssh-rsa" to the
white space.
Management > Telnet/SSH Settings), and then paste the value copied in Step 2 into
the 'Admin Key' field.
4. Click Submit.
5. Connect to the device with SSH, using the following command:
ssh -i admin.key xx.xx.xx.xx
where xx.xx.xx.xx is the device's IP address. RSA-key negotiation occurs
automatically and no password is required.

MP-11x & MP-124
5.3 Establishing a CLI Session

The procedure below describes how to establish a CLI session with the device.
Notes:
• The default login username and password are both "Admin" (case-sensitive).
• Only the primary User Account, which has Security Administration access level
(200) can access the device using Telnet. For configuring the username and
password, see Configuring Web User Accounts on page 66.
 To establish a CLI session with the device:

1. Establish a Telnet or SSH session with the device using its OAMP IP address.
2. Log in to the session using the username and password assigned to the Admin user of
the Web interface.
3. At the login prompt, type the username, and then press Enter:
login: Admin
4. At the password prompt, type the password, and then press Enter:
password: Admin
After logging in, the current directory (root), available commands, available
subdirectories, and a welcome message are displayed at the CLI prompt:
login: Admin
password:
ready. Type "exit" to close the connection.
SIP/ SECurity/ DebugRecording/ MGmt/ ControlProtocol/ CONFiguration/
IPNetworking/ TPApp/ BSP/ PING SHow
/>

5.4 Command Shell

The Command Shell interface is used mainly to display current configuration and
performance.
5.4.1 Getting Familiar with the Command Shell

This section provides a description on how to work with the CLI. This includes basic
navigation commands and accessing subdirectories.
5.4.1.1 Basic Command Shell Commands

The table below describes the commands for navigating the Command Shell interface:
Table 5-1: Basic CLI Commands
Purpose Commands Description
Help h Displays the help for a specific command, action, or

parameter.
Navigation cd Enters another directory.
cd root Returns you to the root directory "/>".
.. Goes up one level.
exit Terminates the CLI session.
5.4.1.2 Accessing Subdirectories

The Command Shell commands are organized in subdirectories. When you establish a CLI
session, you are located in the root directory, indicated by the following prompt:
/>
To access a subdirectory, type its name, and then press Enter. For example, to enter the
MGmt subdirectory, type the following:
/>MGmt
Instead of typing the full name of the subdirectory, you can simply type its abbreviated
format. The abbreviated format is shown in upper case (i.e., capital letters). For example,
for the MGmt subdirectory, you can simply type the following:
/>MG
If you know the full path to a command inside one of the subdirectories, the short format
can be used to run it directly. For example, to run the PERFormance command in the
MGmt subdirectory, type the following:
/>MG/PERF
Note: The subdirectory names and commands are case-insensitive. For example, it
does not matter whether you type "MGmt" or "mgmt".

MP-11x & MP-124
5.4.2 Status Commands

The following table summarizes the Show commands and their corresponding options.
Show CLI Commands
Command Short Format Arguments Description
SHow sh info | dsp | ip | Displays operational data.

log  info: Displays general device information
 dsp: Displays DSP resource information
 ip: Displays information about IP interfaces
SHow INFO sh info - Displays device hardware information,
versions, uptime, temperature reading, and
the last reset reason.
SHow DSP sh dsp status | perf Displays status and version for each DSP
device, along with overall performance
statistics.
SHow IP sh ip conf | perf | Displays IP interface status and configuration,
route along with performance statistics.
Note: The display format may change
according to the configuration.
SHow LOG sh log [stop] Displays (or stops displaying) Syslog
messages in the CLI session.
Example:
/>sh info
Board type: gateway SDH, firmware version 6.60.000.020
Uptime: 0 days, 0 hours, 3 minutes, 54 seconds
Memory usage: 63%
Temperature reading: 39 C
Last reset reason:
Board was restarted due to issuing of a reset from Web interface
Reset Time : 7.1.2012 21.51.13
/>sh dsp status
DSP firmware: 491096AE8 Version:0660.03 Used=0 Free=480 Total=480
DSP device 0: Active Used=16 Free= 0 Total=16
DSP device 6: Inactive

PSEC - DSP firmware: AC491IPSEC Version: 0660.03

CONFERENCE - DSP firmware: AC491256C Version: 0660.03
/>sh dsp perf
DSP Statistics (statistics for 968 seconds):
Active DSP resources: 480
Total DSP resources: 480
DSP usage %: 100
/>sh ip perf
Networking Statistics (statistics for 979 seconds):
IP KBytes TX: 25
IP KBytes RX: 330
IP KBytes TX per second: 0
IP KBytes RX per second: 1
IP Packets TX: 1171
IP Packets RX: 5273
IP Packets TX per second: 3
IP Packets RX per second: 12
Peak KByte/s TX in this interval: 18
Peak KByte/s RX in this interval: 4
Discarded packets: 186
DHCP requests sent: 0
IPSec Security Associations: 0
/>/mg/perf reset
Done.
/>sh ip perf
Networking Statistics (statistics for 2 seconds):
IP KBytes TX: 2
IP KBytes RX: 4
IP KBytes TX per second: 0
IP KBytes RX per second: 1
IP Packets TX: 24
IP Packets RX: 71
IP Packets TX per second: 3
IP Packets RX per second: 12
Peak KByte/s TX in this interval: 18
Peak KByte/s RX in this interval: 4
Discarded packets: 0
DHCP requests sent: 0
IPSec Security Associations: 0
/>sh ip conf
Interface IP Address Subnet Mask Default Gateway
--------- ------------------ -------------- ----------
OAM 10.4.64.13 55.255.0.0 10.4.0.1
Media 10.4.64.13 255.255.0.0 10.4.0.1
Control 10.4.64.13 255.255.0.0 10.4.0.1
MAC address: 00-90-8f-04-5c-e9
/>sh ip route
Destination Mask Gateway Intf Flags
------------- ---------------- ------------------------
0.0.0.0 0.0.0.0 10.4.0.1 OAM A S
10.4.0.0 255.255.0.0 10.4.64.13 OAM A L
127.0.0.0 255.0.0.0 127.0.0.1 AR S
127.0.0.1 255.255.255.255 127.0.0.1 A L H
Flag legend: A=Active R=Reject L=Local S=Static E=rEdirect
M=Multicast
B=Broadcast H=Host I=Invalid
End of routing table, 4 entries displayed.

MP-11x & MP-124
5.5 Ping Command

The Ping command is described in the following table:
Ping Command
PING ping [-n count] Sends ICMP echo request packets to a

[-l size] specified IP address.
[-w timeout]  count: number of packets to send.
[-p cos]  size: payload size in each packet.
 timeout: time (in seconds) to wait for a reply
ip-address
to each packet.
 cos: Class-of-Service (as per 802.1p) to
use.
Example:
/>ping 10.31.2.10
Ping process started for address 10.31.2.10. Process ID - 27.
Reply from 10.31.2.10: bytes=0 time<0ms
Ping statistics for 10.31.2.10:
Packets:Sent = 4, Received = 4, Lost 0 (0% loss),
Approximate round trip times in milli-seconds:
Minimum = 0ms, Maximum = 0ms, Average = 0ms
5.6 Management Commands

The commands under the MGmt directory, described in the table below, display current
performance values.
CLI Management Command
/MGmt/PERFormance /mg/perf basic | control | dsp | Displays performance statistics.

net | reset The reset argument clears all
statistics to zero.

5.7 Configuration Commands

The commands under the CONFiguration directory query and modify the current device
configuration. The following commands are available:
Configuration CLI Commands
Command Short Arguments Description

Format
SetConfigParam IP /conf/scp ip ip-addr subnet def- Sets the IP address, subnet mask,
gw and default gateway address of the
device (on-the-fly).
Note: This command may cause
disruption of service. The CLI session
may disconnect since the device
changes its IP address.
RestoreFactorySettings /conf/rfs Restores all parameters to factory
settings.
SaveAndRestart /conf/sar Saves all current configurations to the
non-volatile memory and resets the
device.
ConfigFile /conf/cf view | get | set Retrieves the full ini file from the
device and allows loading a new ini
file directly in the CLI session.
Note: The argument view displays the
file, page by page. The argument get
displays the file without breaks.

MP-11x & MP-124

User's Manual 6. SNMP-Based Management
6 SNMP-Based Management
The device provides an embedded SNMP Agent to operate with a third-party SNMP
Manager (e.g., element management system or EMS) for operation, administration,
maintenance, and provisioning (OAMP) of the device. The SNMP Agent supports standard
Management Information Base (MIBs) and proprietary MIBs, enabling a deeper probe into
the interworking of the device. The SNMP Agent can also send unsolicited events (SNMP
traps) towards the SNMP Manager. All supported MIB files are supplied to customers as
part of the release.
This section provides configuration relating to SNMP management.
Note: For more information on SNMP support such as SNMP traps, refer to the
SNMP User's Guide.
6.1 Configuring SNMP Community Strings

The SNMP Community String page allows you to configure up to five read-only and up to
five read-write SNMP community strings and to configure the community string that is used
for sending traps.
For detailed descriptions of the SNMP parameters, see 'SNMP Parameters' on page 490.
 To configure the SNMP community strings:

1. Open the SNMP Community String page (Configuration tab > System menu >
Management submenu > SNMP submenu > SNMP Community String).
2. Configure the SNMP community strings parameters according to the table below.

MP-11x & MP-124
To delete a community string, select the Delete check box corresponding to the community
string that you want to delete, and then click Submit.
Table 6-1: SNMP Community String Parameters Description
Community String  Read Only [SNMPReadOnlyCommunityString_x]: Up to five

read-only community strings (up to 19 characters each). The
default string is 'public'.
 Read / Write [SNMPReadWriteCommunityString_x]: Up to
five read / write community strings (up to 19 characters each).
The default string is 'private'.
Trap Community String Community string used in traps (up to 19 characters).
[SNMPTrapCommunityString] The default string is 'trapuser'.

6.2 Configuring SNMP Trap Destinations

The SNMP Trap Destinations page allows you to configure up to five SNMP trap
managers. You can associate a trap destination with SNMPv2 users and specific SNMPv3
users. Associating a trap destination with SNMPv3 users sends encrypted and
authenticated traps to the SNMPv3 destination. By default, traps are sent unencrypted
using SNMPv2.
 To configure SNMP trap destinations:

1. Open the SNMP Trap Destinations page (Configuration tab > System menu >
Management submenu > SNMP > SNMP Trap Destinations).
Figure 6-1: SNMP Trap Destinations Page
2. Configure the SNMP trap manager parameters according to the table below.
3. Select the check box corresponding to the SNMP Manager that you wish to enable.
Note: Only row entries whose corresponding check boxes are selected are applied
when clicking Submit; otherwise, settings revert to their defaults.
Table 6-2: SNMP Trap Destinations Parameters Description
Web: SNMP Manager Enables the SNMP Manager to receive traps and checks
[SNMPManagerIsUsed_x] the validity of the configured destination (IP address and
port number).
 [0] (check box cleared) = (Default) Disables SNMP
Manager
 [1] (check box selected) = Enables SNMP Manager
Web: IP Address Defines the IP address (in dotted-decimal notation, e.g.,
[SNMPManagerTableIP_x] 108.10.1.255) of the remote host used as the SNMP
Manager. The device sends SNMP traps to this IP
address.
Trap Port Defines the port number of the remote SNMP Manager.
[SNMPManagerTrapPort_x] The device sends SNMP traps to this port.
The valid value range is 100 to 4000. The default is 162.

MP-11x & MP-124
Web: Trap User Associates a trap user with the trap destination. This
[SNMPManagerTrapUser] determines the trap format, authentication level, and
encryption level.
 v2cParams (default) = SNMPv2 user community string
 SNMPv3 user configured in 'Configuring SNMP V3
Users' on page 91
Trap Enable Activates the sending of traps to the SNMP Manager.
[SNMPManagerTrapSendingEnable_x]  [0] Disable
 [1] Enable (Default)
6.3 Configuring SNMP Trusted Managers

The SNMP Trusted Managers page allows you to configure up to five SNMP Trusted
Managers, based on IP addresses. By default, the SNMP agent accepts SNMP Get and
Set requests from any IP address, as long as the correct community string is used in the
request. Security can be enhanced by using Trusted Managers, which is an IP address
from which the SNMP agent accepts and processes SNMP requests.
Notes: The SNMP Trusted Managers table can also be configured using the table ini
file parameter, SNMPTrustedMgr_x (see 'SNMP Parameters' on page 490).
 To configure SNMP Trusted Managers:

1. Open the SNMP Trusted Managers page (Configuration tab > System menu >
Management submenu > SNMP submenu > SNMP Trusted Managers).
Figure 6-2: SNMP Trusted Managers
2. Select the check box corresponding to the SNMP Trusted Manager that you want to
enable and for whom you want to define an IP address.
3. Define an IP address in dotted-decimal notation.
5. To save the changes, see 'Saving Configuration' on page 366.

6.4 Configuring SNMP V3 Users

The SNMP v3 Users page allows you to configure authentication and privacy for up to 10
SNMP v3 users.
 To configure SNMP v3 users:

1. Open the SNMP v3 Users page (Configuration tab > System menu > Management
submenu > SNMP submenu > SNMP V3 Users).
2. Click Add; the following dialog box appears:
Figure 6-3: SNMP V3 Setting Page - Add Record Dialog Box
3. Configure the SNMP V3 Setting parameters according to the table below.

4. Click Submit to apply your settings.
Notes:
• If you delete a user that is associated with a trap destination (in 'Configuring
SNMP Trap Destinations' on page 89), the configured trap destination becomes
disabled and the trap user reverts to default (i.e., SNMPv2).
• The SNMP v3 Users table can also be configured using the table ini file
parameter, SNMPUsers (see 'SNMP Parameters' on page 490).
Table 6-3: SNMP V3 Users Parameters
Index The table index.

[SNMPUsers_Index] The valid range is 0 to 9.
User Name Name of the SNMP v3 user. This name must be unique.
[SNMPUsers_Username]
Authentication Protocol Authentication protocol of the SNMP v3 user.
[SNMPUsers_AuthProtocol]  [0] None (default)
 [1] MD5
 [2] SHA-1

MP-11x & MP-124
Privacy Protocol Privacy protocol of the SNMP v3 user.

[SNMPUsers_PrivProtocol]  [0] None (default)
 [1] DES
 [2] 3DES
 [3] AES-128
 [4] AES-192
 [5] AES-256
Authentication Key Authentication key. Keys can be entered in the form of a text
[SNMPUsers_AuthKey] password or long hex string. Keys are always persisted as long hex
strings and keys are localized.
Privacy Key Privacy key. Keys can be entered in the form of a text password or
[SNMPUsers_PrivKey] long hex string. Keys are always persisted as long hex strings and
keys are localized.
Group The group with which the SNMP v3 user is associated.
[SNMPUsers_Group]  [0] Read-Only (default)
 [1] Read-Write
 [2] Trap
Note: All groups can be used to send traps.

User's Manual 7. EMS-Based Management
7 EMS-Based Management
AudioCodes Element Management System (EMS) is an advanced solution for standards-
based management of gateways within VoP networks, covering all areas vital for the
efficient operation, administration, management and provisioning (OAM&P) of AudioCodes'
families of gateways. The EMS enables Network Equipment Providers (NEPs) and System
Integrators (SIs) the ability to offer customers rapid time-to-market and inclusive, cost-
effective management of next-generation networks. The standards-compliant EMS uses
distributed SNMP-based management software, optimized to support day-to-day Network
Operation Center (NOC) activities, offering a feature-rich management framework. It
supports fault management, configuration and security.
Note: For more information on using the EMS tool, refer to the EMS User's Manual
and EMS Server IOM Manual.

MP-11x & MP-124

User's Manual 8. INI File-Based Management
8 INI File-Based Management

The device can be configured using an ini file, which is a text-based file with an ini file
extension name that can be created using any standard text-based editor such as
Notepad. Each configuration element of the device has a corresponding ini file parameter
that you can use in the ini file for configuring the device. When you have created the ini file
with your ini file parameter settings, you apply these settings to the device by installing
(loading) the ini file to the device.
Notes:
• For a list and description of the ini file parameters, see 'Configuration Parameters
Reference' on page 475.
• To restore the device to default settings using the ini file, see 'Restoring Factory
Defaults' on page 389.
8.1 INI File Format

The ini file can be configured with any number of parameters. These ini file parameters can
be one of the following types:
 Individual parameters - see 'Configuring Individual ini File Parameters' on page 95
 Table parameters - see 'Configuring Table ini File Parameters' on page 96
8.1.1 Configuring Individual ini File Parameters

The syntax for configuring individual ini file parameters in the ini file is as follows:
 An optional, subsection name (or group name) enclosed in square brackets "[...]". This
is used to conveniently group similar parameters by their functionality.
 Parameter name, followed by an equal "=" sign and then its value.
 Comments must be preceded by a semicolon ";".
[subsection name]
parameter name = value
parameter name = value
; this is a comment line
; for example:
[System Parameters]
SyslogServerIP = 10.13.2.69
EnableSyslog = 1
For general ini file formatting rules, see 'General ini File Formatting Rules' on page 97.

MP-11x & MP-124
8.1.2 Configuring Table ini File Parameters

The table ini file parameters allow you to configure tables, which include multiple
parameters (columns) and row entries (indices). When loading an ini file to the device, it's
recommended to include only tables that belong to applications that are to be configured
(dynamic tables of other applications are empty, but static tables are not).
The table ini file parameter is composed of the following elements:
 Title of the table: The name of the table in square brackets, e.g.,
[MY_TABLE_NAME].
 Format line: Specifies the columns of the table (by their string names) that are to be
configured.
• The first word of the Format line must be "FORMAT", followed by the Index field
name and then an equal "=" sign. After the equal sign, the names of the columns
are listed.
• Columns must be separated by a comma ",".
• The Format line must only include columns that can be modified (i.e., parameters
that are not specified as read-only). An exception is Index fields, which are
mandatory.
• The Format line must end with a semicolon ";".
 Data line(s): Contain the actual values of the columns (parameters). The values are
interpreted according to the Format line.
• The first word of the Data line must be the table’s string name followed by the
Index field.
• Columns must be separated by a comma ",".
• A Data line must end with a semicolon ";".
 End-of-Table Mark: Indicates the end of the table. The same string used for the
table’s title, preceded by a backslash "\", e.g., [\MY_TABLE_NAME].
The following displays an example of the structure of a table ini file parameter.
[Table_Title]
; This is the title of the table.
FORMAT Index = Column_Name1, Column_Name2, Column_Name3;
; This is the Format line.
Index 0 = value1, value2, value3;
Index 1 = value1, $$, value3;
; These are the Data lines.
[\Table_Title]
; This is the end-of-the-table-mark.
The table ini file parameter formatting rules are listed below:
 Indices (in both the Format and the Data lines) must appear in the same order. The
Index field must never be omitted.
 The Format line can include a subset of the configurable fields in a table. In this case,
all other fields are assigned with the pre-defined default values for each configured
line.
 The order of the fields in the Format line isn’t significant (as opposed to the Index
fields). The fields in the Data lines are interpreted according to the order specified in
the Format line.
 The double dollar sign ($$) in a Data line indicates the default value for the parameter.
 The order of the Data lines is insignificant.
 Data lines must match the Format line, i.e., it must contain exactly the same number
of Indices and Data fields and must be in exactly the same order.

 A row in a table is identified by its table name and Index field. Each such row may
appear only once in the ini file.
 Table dependencies: Certain tables may depend on other tables. For example, one
table may include a field that specifies an entry in another table. This method is used
to specify additional attributes of an entity, or to specify that a given entity is part of a
larger entity. The tables must appear in the order of their dependency (i.e., if Table X
is referred to by Table Y, Table X must appear in the ini file before Table Y).
For general ini file formatting rules, see 'General ini File Formatting Rules' on page 97.
The table below displays an example of a table ini file parameter:
[ CodersGroup0 ]
FORMAT CodersGroup0_Index = CodersGroup0_Name, CodersGroup0_pTime,
CodersGroup0_rate, CodersGroup0_PayloadType, CodersGroup0_Sce;
CodersGroup0 0 = g711Alaw64k, 20, 0, 255, 0;
CodersGroup0 1 = eg711Ulaw, 10, 0, 71, 0;
[ \CodersGroup0 ]
Note: Do not include read-only parameters in the table ini file parameter as this can
cause an error when attempting to load the file to the device.
8.1.3 General ini File Formatting Rules

The ini file must adhere to the following formatting rules:
 The ini file name must not include hyphens "-" or spaces; if necessary, use an
underscore "_" instead.
 Lines beginning with a semi-colon ";" are ignored. These can be used for adding
remarks in the ini file.
 A carriage return (i.e., Enter) must be done at the end of each line.
 The number of spaces before and after the equals sign "=" is irrelevant.
 Subsection names for grouping parameters are optional.
 If there is a syntax error in the parameter name, the value is ignored.
 Syntax errors in the parameter's value can cause unexpected errors (parameters may
be set to the incorrect values).
 Parameter string values that denote file names (e.g., CallProgressTonesFileName)
must be enclosed with inverted commas, e.g., CallProgressTonesFileName =
'cpt_usa.dat'.
 The parameter name is not case-sensitive.
 The parameter value is not case-sensitive, except for coder names.
 The ini file must end with at least one carriage return.

MP-11x & MP-124
8.2 Loading an ini File

You can load an ini file to the device using the following methods:
 Web interface, using any of the following pages:
• Configuration File - see 'Backing Up and Loading Configuration File' on page 388
• Load Auxiliary Files - see 'Loading Auxiliary Files' on page 369
 AudioCodes AcBootP utility, which uses Bootstrap Protocol (BootP) and acts as a
TFTP server. For information on using the AcBootP utility, refer to AcBootP Utility
User's Guide.
 Any standard TFTP server. This is done by storing the ini file on a TFTP server and
then having the device download the file from it.
When loaded to the device, the configuration settings of the ini file are saved to the
device's non-volatile memory. If a parameter is not included in the loaded ini file, the
following occurs:
 Using the Load Auxiliary Files page: Current settings for parameters that were not
included in the loaded ini file are retained.
 All other methods: The default is assigned to the parameters that were not included in
the loaded ini file and thereby, overriding values previously configured for these
parameters.
Notes:
• For a list and description of the ini file parameters, see 'Configuration Parameters
• Some parameters are configurable only through the ini file (and not the Web
interface).
• To restore the device to default settings using the ini file, see 'Restoring Factory
Defaults' on page 389.

8.3 Modifying an ini File

You can modify an ini file currently used by the device. Modifying an ini file instead of
loading an entirely new ini file preserves the device's current configuration.
 To modify an ini file:

1. Save the device's configuration as an ini file on your computer, using the Web
interface (see 'Loading an ini File' on page 98).
2. Open the ini file using a text file editor such as Notepad, and then modify the ini file
parameters as required.
3. Save the modified ini file, and then close the file.
4. Load the modified ini file to the device (see 'Loading an ini File' on page 98).
Tip: Before loading the ini file to the device, verify that the file extension of the file
is .ini.
8.4 Secured Encoded ini File

The ini file contains sensitive information that is required for the functioning of the device.
The file may be loaded to the device using TFTP or HTTP. These protocols are not secure
and are vulnerable to potential hackers. To overcome this security threat, the AudioCodes
DConvert utility allows you to binary-encode (encrypt) the ini file before loading it to the
device. For more information, refer to DConvert Utility User's Guide.
Notes:
• The procedure for loading an encoded ini file is identical to the procedure for
loading an unencoded ini file (see 'Loading an ini File' on page 98).
• If you download from the device (to a folder on your computer) an ini file that was
loaded encoded to the device, the file is saved as a regular ini file (i.e.,
unencoded).
8.5 Configuring Password Display in ini File

Passwords can be displayed in the ini file in one of the following formats, configured by the
INIPasswordsDisplayType ini file parameter:
 Obscured: The password characters are concealed and displayed as an encoded
string. The password is displayed using the syntax, $1$<obscured password>, for
example, $1$S3p+fno=.
 Hidden: the password is replaced with an asterisk (*).
When you save an ini file from the device to a PC, the passwords are displayed according
to the chosen format. When you load an ini file to the device, obscured passwords are
parsed and applied to the device; hidden passwords are ignored.
By default, the enabled format is obscured passwords, thus enabling their full recovery in
case of configuration restore or copy to another device.
When obscured password mode is enabled, you can enter a password in the ini file using
any of the following formats:

MP-11x & MP-124
 $1$<obscured password>: Password in obscured format as generated by the device;

useful for restoring device configuration and copying configuration from one device to
another.
$0$<plain text>: Password can be entered in plain text; useful for configuring a new
password. When the ini file is loaded to the device and then later saved from the device to
a PC, the password is displayed obscured (i.e., $1$<obscured password>).

User's Manual 9. TR-069 Based Management
9 TR-069 Based Management

The device supports TR-069 CPE WAN Management Protocol (CWMP) based
management, which is used for remote management of CPE devices. This allows the
device to be configured and monitored from a management application running on a
remote Auto-Configuration Server (ACS).
9.1 TR-069
TR-069 (Technical Report 069) is a specification published by Broadband Forum
(www.broadband-forum.org) entitled CPE WAN Management Protocol (CWMP). It defines
an application layer protocol for remote management of end-user devices.
TR-069 uses a bi-directional SOAP/HTTP protocol for communication between the
customer premises equipment (CPE) and the Auto Configuration Servers (ACS). The TR-
069 connection to the ACS can be done on the LAN or WAN interface.
The protocol stack looks as follows:
Table 9-1: TR-069 Protocol Stack
CPE/ACS Management Application
RPC Methods
SOAP
HTTP
SSL/TLS
TCP/IP
Version 6.6 101 MP-11x & MP-124

MP-11x & MP-124
Communication is typically established by the CPE; hence, messages from CPE to ACS
are typically carried in HTTP requests, and messages from ACS to CPE in HTTP
responses.
Figure 9-1: TR-069 Session Example
Communication between ACS and CPE is defined via Remote Procedure Call (RPC)
methods. TR-069 defines a generic mechanism by which an ACS can read or write
parameters to configure a CPE and monitor CPE status and statistics. It also defines the
mechanism for file transfer and firmware/software management. However, it does not
define individual parameters; these are defined in separate documents, as described
below. Some of the RPC methods are Configuration File Download, Firmware upgrade,
Get Parameter Value, Set Parameter Value, Reboot, and the upload and download files.

TR-106 defines the “data model” template for TR-069 enabled devices. The Data Model
consists of objects and parameters hierarchically organized in a tree with a single Root
Object, typically named Device. Arrays of objects are supported by appending a numeric
index to the object name (e.g. ABCService.1 in the example below); such objects are
called “multi-instance objects”.
Figure 9-2: TR-069 Model Data Example
Below is a list of some of the TR-069 methods:

 CPE Methods:
• GetRPCMethods: Used by the CPE or ACS to discover the set of methods
supported by the Server or CPE it is in communication with.
• SetParameterValues: Used by the ACS to modify the value of CPE parameter(s).
• GetParameterValues: Used by the ACS to obtain the value of CPE parameter(s).
• GetParameterNames: Used by the ACS to discover the parameters accessible on
a particular CPE.
• SetParameterAttributes: Used by the ACS to modify attributes associated with
CPE parameter(s).
• GetParameterAttributes: Used by the ACS to read the attributes associated with
CPE parameter(s).
• AddObject: Used by the ACS to create a new instance of a multi-instance
object—a collection of parameters and/or other objects for which multiple
instances are defined.
• DeleteObject: Removes a particular instance of an object.
• Download: Used by the ACS to cause the CPE to download the following file(s)
from a designated location:
♦ Firmware Upgrade Image (File Type = 1) - cmp file.
♦ Vendor Configuration File (File Type = 3) - output of show running-
config CLI command, which includes Data and Voice configuration.
Version 6.6 103 MP-11x & MP-124

MP-11x & MP-124
The CPE responds to the Download method, indicating successful or

unsuccessful completion via one of the following:
♦ A DownloadResponse with the Status argument set to zero (indicating
success), or a fault response to the Download request (indicating failure).
♦ A TransferComplete message sent later in the same session as the
Download request (indicating either success or failure). In this case, the
Status argument in the corresponding DownloadResponse has a value of
one.
♦ A TransferComplete message sent in a subsequent session (indicating
success or failure). In this case, the Status argument in the corresponding
DownloadResponse has a value of one.
Figure 9-3: Download Method Executuion Example
• Upload: Used by the ACS to cause the CPE to upload (to the ACS) the following
files to a designated location:
♦ Vendor Configuration File (File Type = 1 or 3): Output of show running-
config CLI command, which includes Data and Voice configuration. For
File Type 3 (where index is included – see below) only one instance of the
file is supported.

♦ Vendor Log File (File Type = 2 or 4): “Aggregated” log file. For File Type 2,
the last file is supported. For File Type 4 (where index is included – see
below), multiple files is supported.
The CPE responds to the Upload method, indicating successful or unsuccessful
completion via the UploadResponse or TransferComplete method.
For a complete description of the Upload method, refer to TR-069 Amendment 3
section A.4.1.5.
• Reboot: Reboots the CPE. The CPE sends the method response and completes
the remainder of the session prior to rebooting.
• X_0090F8_CommandResponse: Runs CLI commands.
 ACS Methods:
• Inform: A CPE must call this method to initiate a transaction sequence whenever
a connection to an ACS is established.
• TransferComplete: Informs the ACS of the completion (either successful or
unsuccessful) of a file transfer initiated by an earlier Download or Upload method
call.
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9.2 TR-104
The device supports TR-104 for configuration. This support is for the SIP (VoIP) application
layer and applies to FXS interfaces (lines) only. TR-104 defines a "data model" template for
TR-069 enabled devices. The "data model" that is applicable to the AudioCodes device is
defined in the DSL Forum TR-104 – "DSLHome™ Provisioning Parameters for VoIP CPE"
at http://www.broadband-forum.org/technical/download/TR-104.pdf.
The hierarchical tree structure of the supported TR-104 objects is shown below:
Figure 9-4: Hierarchical Tree Structure of TR-104 Objects
 InternetGatewayDevice.Services.VoiceService: Top-level object.

 InternetGatewayDevice.Services.VoiceService.1.Capabilities: (Read-Only) Displays
the overall capabilities of the device.
• InternetGatewayDevice.Services.VoiceService.1.Capabilities.Codecs: (Read-
Only) Lists supported codecs (according to devices installed Software Feature
Key).
• InternetGatewayDevice.Services.VoiceService.1.Capabilities.SIP: (Read-Only)
Displays various SIP settings such as SIP transport type.
 InternetGatewayDevice.Services.VoiceService.1.VoiceProfile.1: Corresponds to one
or more FXS lines that share the same basic configuration:
• InternetGatewayDevice.Services.VoiceService.1.VoiceProfile.1.FaxT38:
Configures fax T.38 relay.
• InternetGatewayDevice.Services.VoiceService.1.VoiceProfile.1.Line:
Corresponds to an FXS line (as configured in the Trunk Group table). It enables
and configures each FXS line (number).

♦ InternetGatewayDevice.Services.VoiceService.1.VoiceProfile.1.Line.{i}.Code
c.List.{i}: Configures voice coder used by specific FXS line.
♦ InternetGatewayDevice.Services.VoiceService.1.VoiceProfile.1.Line.{i}.Callin
gFeatures: Configures voice parameters per FXS line such as caller ID.
♦ InternetGatewayDevice.Services.VoiceService.1.VoiceProfile.1.Line.{i}.SIP:
Configures username/password per FXS line. AudioCodes maps this object
to the corresponding entry in the Authentication table
• InternetGatewayDevice.Services.VoiceService.1.VoiceProfile.1.SIP: Configures
SIP parameters specific to the UA such as Proxy server.
• InternetGatewayDevice.Services.VoiceService.1.VoiceProfile.1.RTP: Configures
various RTP parameters for the FXS lines such as RTCP and SRTP.
9.3 Configuring TR-069

The CWMP/TR-069 Settings page is used to enable and configure TR-069.
 To configure TR-069:
1. Open the CWMP/TR-069 Settings page (Configuration tab > System menu >
Management > CWMP).
Figure 9-5: CWMP/TR-069 Settings Page
2. Configure the parameters as required. For a description of the TR-069 parameters,

see ''TR-069 Parameters'' on page 493.
3. Click Submit.
4. Reset the device with a burn-to-flash memory for the settings to take effect.
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Part III
General System Settings
User's Manual 10. Configuring Certificates
10 Configuring Certificates
The Certificates page allows you to configure X.509 certificates, which are used for secure
management of the device, secure SIP transactions, and other security applications.
Note: The device is shipped with an active TLS setup. Thus, configure certificates
only if required.
10.1 Replacing the Device's Certificate

The device is supplied with a working TLS configuration consisting of a unique self-signed
server certificate. If an organizational Public Key Infrastructure (PKI) is used, you may wish
to replace this certificate with one provided by your security administrator.
 To replace the device's certificate:

1. Your network administrator should allocate a unique DNS name for the device (e.g.,
dns_name.corp.customer.com). This DNS name is used to access the device and
therefore, must be listed in the server certificate.
2. If the device is operating in HTTPS mode, then set the 'Secured Web Connection
(HTTPS)' parameter (HTTPSOnly) to HTTP and HTTPS (see 'Configuring Web
Security Settings' on page 73). This ensures that you have a method for accessing the
device in case the new certificate does not work. Restore the previous setting after
testing the configuration.
3. Open the Certificates page (Configuration tab > System menu > Certificates).
4. Under the Certificate Signing Request group, do the following:
a. In the 'Subject Name [CN]' field, enter the DNS name.
b. Fill in the rest of the request fields according to your security provider's
instructions.
c. Click the Create CSR button; a textual certificate signing request is displayed in
the area below the button:
Figure 10-1: Certificate Signing Request Group
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5. Copy the text and send it to your security provider. The security provider, also known
as Certification Authority or CA, signs this request and then sends you a server
certificate for the device.
6. Save the certificate to a file (e.g., cert.txt). Ensure that the file is a plain-text file
containing the"‘BEGIN CERTIFICATE" header, as shown in the example of a Base64-
Encoded X.509 Certificate below:
-----BEGIN CERTIFICATE-----
MIIDkzCCAnugAwIBAgIEAgAAADANBgkqhkiG9w0BAQQFADA/MQswCQYDVQQGEw
JGUjETMBEGA1UEChMKQ2VydGlwb3N0ZTEbMBkGA1UEAxMSQ2VydGlwb3N0ZSBT
ZXJ2ZXVyMB4XDTk4MDYyNDA4MDAwMFoXDTE4MDYyNDA4MDAwMFowPzELMAkGA1
UEBhMCRlIxEzARBgNVBAoTCkNlcnRpcG9zdGUxGzAZBgNVBAMTEkNlcnRpcG9z
dGUgU2VydmV1cjCCASEwDQYJKoZIhvcNAQEBBQADggEOADCCAQkCggEAPqd4Mz
iR4spWldGRx8bQrhZkonWnNm`+Yhb7+4Q67ecf1janH7GcN/SXsfx7jJpreWUL
f7v7Cvpr4R7qIJcmdHIntmf7JPM5n6cDBv17uSW63er7NkVnMFHwK1QaGFLMyb
FkzaeGrvFm4k3lRefiXDmuOe+FhJgHYezYHf44LvPRPwhSrzi9+Aq3o8pWDguJ
uZDIUP1F1jMa+LPwvREXfFcUW+w==
-----END CERTIFICATE-----
7. Scroll down to the Upload certificates files from your computer group, click the
Browse button corresponding to the 'Send Device Certificate...' field, navigate to the
cert.txt file, and then click Send File.
8. After the certificate successfully loads to the device, save the configuration with a
device reset (see 'Saving Configuration' on page 366); the Web interface uses the
provided certificate.
9. Open the Certificates page again and verify that under the Certificate information
group (at the top of the page), the 'Private key' read-only field displays "OK";
otherwise, consult your security administrator:
Figure 10-2: Private key "OK" in Certificate Information Group
10. If the device was originally operating in HTTPS mode and you disabled it in Step 2,
then return it to HTTPS by setting the 'Secured Web Connection (HTTPS)' parameter
to HTTPS Only, and then reset the device with a flash burn.
Notes:
• The certificate replacement process can be repeated when necessary (e.g., the
new certificate expires).
• It is possible to use the IP address of the device (e.g., 10.3.3.1) instead of a
qualified DNS name in the Subject Name. This is not recommended since the IP
address is subject to change and may not uniquely identify the device.
• The device certificate can also be loaded via the Automatic Update Facility by
using the HTTPSCertFileName ini file parameter.

10.2 Loading a Private Key

The device is shipped with a self-generated random private key, which cannot be extracted
from the device. However, some security administrators require that the private key be
generated externally at a secure facility and then loaded to the device through
configuration. Since private keys are sensitive security parameters, take precautions to
load them over a physically-secure connection such as a back-to-back Ethernet cable
connected directly to the managing computer.
 To replace the device's private key:

1. Your security administrator should provide you with a private key in either textual PEM
(PKCS #7) or PFX (PKCS #12) format. The file may be encrypted with a short pass-
phrase, which should be provided by your security administrator.
2. If the device is operating in HTTPS mode, then set the 'Secured Web Connection
(HTTPS)' field (HTTPSOnly) to HTTP and HTTPS (see 'Configuring Web Security
Settings' on page 73). This ensures that you have a method for accessing the device
in case the new configuration does not work. Restore the previous setting after testing
the configuration.
3. Open the Certificates page (Configuration tab > System menu > Certificates) and
scroll down to the Upload certificate files from your computer group.
Figure 10-3: Upload Certificate Files from your Computer Group
4. Fill in the 'Private key pass-phrase' field, if required.

5. Click the Browse button corresponding to the 'Send Private Key' field, navigate to the
key file, and then click Send File.
6. If the security administrator has provided you with a device certificate file, load it using
the 'Send Device Certificate' field.
7. After the files successfully load to the device, save the configuration with a device
reset (see 'Saving Configuration' on page 366); the Web interface uses the new
configuration.
8. Open the Certificates page again, and verify that under the Certificate information
group (at the top of the page) the 'Private key' read-only field displays "OK"; otherwise,
consult your security administrator.
9. If the device was originally operating in HTTPS mode and you disabled it in Step 2,
then enable it by setting the 'Secured Web Connection (HTTPS)' field to HTTPS Only.
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10.3 Mutual TLS Authentication

By default, servers using TLS provide one-way authentication. The client is certain that the
identity of the server is authentic. When an organizational PKI is used, two-way
authentication may be desired - both client and server should be authenticated using X.509
certificates. This is achieved by installing a client certificate on the managing PC and
loading the root CA's certificate to the device's Trusted Root Certificate Store. The Trusted
Root Certificate file may contain more than one CA certificate combined, using a text
editor.
Since X.509 certificates have an expiration date and time, the device must be configured to
use NTP (see 'Simple Network Time Protocol Support' on page 119) to obtain the current
date and time. Without the correct date and time, client certificates cannot work.
 To enable mutual TLS authentication for HTTPS:

1. Set the 'Secured Web Connection (HTTPS)' field to HTTPS Only (see 'Configuring
Web Security Settings' on page 73) to ensure you have a method for accessing the
device in case the client certificate does not work. Restore the previous setting after
testing the configuration.
2. Open the Certificates page (see 'Replacing the Device's Certificate' on page 111).
3. In the Upload certificate files from your computer group, click the Browse button
corresponding to the 'Send Trusted Root Certificate Store ...' field, navigate to the file,
and then click Send File.
4. When the operation is complete, set the 'Requires Client Certificates for HTTPS
connection' field to Enable (see 'Configuring Web Security Settings' on page 73).
5. Save the configuration with a device reset (see 'Saving Configuration' on page 366).
When a user connects to the secured Web interface of the device:
 If the user has a client certificate from a CA that is listed in the Trusted Root Certificate
file, the connection is accepted and the user is prompted for the system password.
 If both the CA certificate and the client certificate appear in the Trusted Root
Certificate file, the user is not prompted for a password (thus, providing a single-sign-
on experience - the authentication is performed using the X.509 digital signature).
 If the user does not have a client certificate from a listed CA or does not have a client
certificate, the connection is rejected.
Notes:
• The process of installing a client certificate on your PC is beyond the scope of this
document. For more information, refer to your operating system documentation,
and/or consult your security administrator.
• The root certificate can also be loaded via the Automatic Update facility, using the
HTTPSRootFileName ini file parameter.
• You can enable the device to check whether a peer's certificate has been revoked
by an Online Certificate Status Protocol (OCSP) server (see Configuring
Certificate Revocation Checking (OCSP) on page 116.

10.4 Self-Signed Certificates

The device is shipped with an operational, self-signed server certificate. The subject name
for this default certificate is 'ACL_nnnnnnn', where nnnnnnn denotes the serial number of
the device. However, this subject name may not be appropriate for production and can be
changed while still using self-signed certificates.
 To change the subject name and regenerate the self-signed certificate:

1. Before you begin, ensure the following:
• You have a unique DNS name for the device (e.g.,
dns_name.corp.customer.com). This name is used to access the device and
should therefore, be listed in the server certificate.
• No traffic is running on the device. The certificate generation process is disruptive
to traffic and should be executed during maintenance time.
3. In the 'Subject Name [CN]' field, enter the fully-qualified DNS name (FQDN) as the
certificate subject, select the desired private key size (in bits), and then click Generate
self-signed; after a few seconds, a message appears displaying the new subject
name.
4. Save the configuration with a device reset (see 'Saving Configuration' on page 366)
for the new certificate to take effect.
10.5 TLS Server Certificate Expiry Check

The device can periodically check the validation date of the installed TLS server certificate.
This periodic check interval is user-defined. In addition, within a user-defined number of
days before the installed TLS server certificate expires, the device can be configured to
send the SNMP trap, acCertificateExpiryNotifiaction to notify of the impending certificate
expiration.
 To configure TLS certificate expiry checks and notification:

2. In the 'TLS Expiry Check Start' field, enter the number of days before the installed TLS
server certificate is to expire at which the device must send a trap to notify of this.
Figure 10-4: TLS Expiry Settings Group
3. In the 'TLS Expiry Check Period' field, enter the periodical interval (in days) for
checking the TLS server certificate expiry date. By default, the device checks the
certificate every 7 days.
4. Click the Submit TLS Expiry Settings button.
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10.6 Configuring Certificate Revocation Checking (OCSP)

Some Public-Key Infrastructures (PKI) can revoke a certificate after it has been issued.
You can configure the device to check whether a peer's certificate has been revoked, using
the Online Certificate Status Protocol (OCSP). When OCSP is enabled, the device queries
the OCSP server for revocation information whenever a peer certificate is received (IPSec,
TLS client mode, or TLS server mode with mutual authentication).
 To configure OCSP:
1. Open the General Security Settings page (Configuration tab > VoIP menu >
Security > General Security Settings).
Figure 10-5: OCSP Parameters
2. Configure the OCSP parameters as required. For a description of these parameters,

see OCSP Parameters on page 515.
3. Click Submit.
Notes:
• The device does not query OCSP for its own certificate.
• Some PKIs do not support OCSP but generate Certificate Revocation Lists
(CRLs). For such cases, set up an OCSP server such as OCSPD.

10.7 Loading Certificate Chain for Trusted Root

A certificate chain is a sequence of certificates where each certificate in the chain is signed
by the subsequent certificate. The last certificate in the list of certificates is the Root CA
certificate, which is self-signed. The purpose of a certificate chain is to establish a chain of
trust from a child certificate to the trusted root CA certificate. The CA vouches for the
identity of the child certificate by signing it. A client certificate is considered trusted if one of
the CA certificates up the certificate chain is found in the server certificate directory.
Figure 10-6: Certificate Chain Hierarchy
For the device to trust a whole chain of certificates, you need to combine the certificates
into one text file (using a text editor). Once done, upload the file using the 'Trusted Root
Certificate Store' field in the Certificates page.
Notes: The maximum supported size of the combined file of trusted chain of
certificates is 100,000 bytes (including the certificate's headers).
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User's Manual 11. Date and Time
11 Date and Time

The date and time of the device can be configured manually or it can be obtained
automatically from a Simple Network Time Protocol (SNTP) server.
11.1 Configuring Date and Time Manually

The date and time of the device can be configured manually.
 To manually configure the device's date and time, using the Web interface:
1. Open the Regional Settings page (Configuration tab > System menu > Regional
Settings).
Figure 11-1: Regional Settings Page
2. Enter the current date and time of the geographical location in which the device is
installed.
3. Click the Submit button.
Notes:
• If the device is configured to obtain the date and time from an SNTP server, the
fields on this page are read-only, displaying the received date and time.
• After performing a hardware reset, the date and time are returned to their defaults
and thus, should be updated.
11.2 Automatic Date and Time through SNTP Server

The Simple Network Time Protocol (SNTP) client functionality generates requests and
reacts to the resulting responses using the NTP version 3 protocol definitions (according to
RFC 1305). Through these requests and responses, the NTP client synchronizes the
system time to a time source within the network, thereby eliminating any potential issues
should the local system clock 'drift' during operation. By synchronizing time to a network
time source, traffic handling, maintenance, and debugging become simplified for the
network administrator.
The NTP client follows a simple process in managing system time: the NTP client requests
an NTP update, receives an NTP response, and then updates the local system clock based
on a configured NTP server within the network.
The client requests a time update from a specified NTP server at a specified update
interval. In most situations, this update interval is every 24 hours based on when the
system was restarted. The NTP server identity (as an IP address or FQDN) and the update
interval are user-defined, or an SNMP MIB object.
When the client receives a response to its request from the identified NTP server, it must
be interpreted based on time zone or location offset that the system is to a standard point
of reference called the Universal Time Coordinate (UTC). The time offset that the NTP
client uses is configurable.
If required, the clock update is performed by the client as the final step of the update
process. The update is performed in such a way as to be transparent to the end users. For
instance, the response of the server may indicate that the clock is running too fast on the
client. The client slowly robs bits from the clock counter to update the clock to the correct
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time. If the clock is running too slow, then in an effort to catch the clock up, bits are added
to the counter, causing the clock to update quicker and catch up to the correct time. The
advantage of this method is that it does not introduce any disparity in the system time that
is noticeable to an end user or that could corrupt call timeouts and timestamps.
The procedure below describes how to configure SNTP.
 To configure SNTP using the Web interface:

1. Open the Application Settings page (Configuration tab > System menu >
Application Settings).
Figure 11-2: SNTP Configuration in Application Settings Page
2. Configure the NTP parameters:

• 'NTP Server DN/IP' (NTPServerIP) - defines the IP address or FQDN of the NTP
server.
• 'NTP UTC Offset' (NTPServerUTCOffset) - defines the time offset in relation to
the UTC. For example, if your region is 2 hours ahead of the UTC, enter "2".
• 'NTP Updated Interval' (NTPUpdateInterval) - defines the period after which the
date and time of the device is updated.
• 'NTP Secondary Server IP' (NTPSecondaryServerIP) - defines the secondary
NTP server.
3. Configure daylight saving, if required:
• 'Day Light Saving Time' (DayLightSavingTimeEnable) - enables daylight saving
time.
• 'DST Mode' - Determines the range type for configuring the start and end date for
daylight saving:
♦ Day of Year: The range is configured by date of month, for example, from
January 4 to August 31.
♦ Day of month: The range is configured by day of month, for example, from
the second Sunday of May January to the last Sunday of August.
• 'Start Time' (DayLightSavingTimeStart) and 'End Time' (DayLightSavingTimeEnd)
- defines the period for which daylight saving time is relevant.
• 'Offset' (DayLightSavingTimeOffset) - defines the offset in minutes to add to the
time for daylight saving. For example, if your region has daylight saving of one
hour, the time received from the NTP server is 11:00, and the UTC offset for your
region is +2 (i.e., 13:00), you need to enter "60" to change the local time to 14:00.
4. Verify that the device is set to the correct date and time. You can do this by viewing
the date and time in the Regional Settings page, as described in 'Configuring Date
and Time Manually' on page 119.

Part IV
General VoIP Configuration
User's Manual 12. Network
12 Network
This section describes the network-related configuration.
12.1 Ethernet Interface Configuration

The device's Ethernet connection can be configured, using the ini file parameter
EthernetPhyConfiguration, to one of the following modes:
 Manual:
• 10Base-T Full-Duplex
• 100Base-TX Half-Duplex or 100Base-TX Full-Duplex
 Auto-Negotiation: chooses common transmission parameters such as speed and
duplex mode
The Ethernet connection should be configured according to the following recommended
guidelines:
 When the device's Ethernet port is configured for Auto-Negotiation, the opposite port
must also operate in Auto-Negotiation. Auto-Negotiation falls back to Half-Duplex
mode when the opposite port is not in Auto-Negotiation mode, but the speed in this
mode is always configured correctly. Configuring the device to Auto-Negotiation mode
while the opposite port is set manually to Full-Duplex is invalid as it causes the device
to fall back to Half-Duplex mode while the opposite port is Full-Duplex. Any mismatch
configuration can yield unexpected functioning of the Ethernet connection.
 When configuring the device's Ethernet port manually, the same mode (i.e., Half
Duplex or Full Duplex) and speed must be configured on the remote Ethernet port. In
addition, when the device's Ethernet port is configured manually, it is invalid to set the
remote port to Auto-Negotiation. Any mismatch configuration can yield unexpected
functioning of the Ethernet connection.
 It's recommended to configure the port for best performance and highest bandwidth
(i.e., Full Duplex with 100Base-TX), but at the same time adhering to the guidelines
listed above.
Note: For remote configuration, the device should be in the correct Ethernet setting
prior to the time this parameter takes effect. When, for example, the device is
configured using BootP/TFTP, the device performs many Ethernet-based transactions
prior to reading the ini file containing this device configuration parameter. To resolve
this problem, the device always uses the last Ethernet setup mode configured. In this
way, if you want to configure the device to operate in a new network environment in
which the current Ethernet setting of the device is invalid, you should first modify this
parameter in the current network so that the new setting holds next time the device is
restarted. After reconfiguration has completed, connect the device to the new network
and restart it. As a result, the remote configuration process that occurs in the new
network uses a valid Ethernet configuration
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12.2 Configuring IP Network Interfaces

You can configure a single VoIP network interface for all applications, which includes
OAMP (management traffic), call control (SIP messages), and media (RTP traffic), or you
can configure multiple logical, IP network interfaces for these applications. A need often
arises to have logically separated network segments for these applications for
administration and security. This can be achieved by employing Layer-2 VLANs and Layer-
3 subnets.
The figure below illustrates a typical network architecture where the device is configured
with three network interfaces for the OAMP, call control, and media applications. The
device is connected to a VLAN-aware switch for directing traffic from and to the device to
the three separated Layer-3 broadcast domains according to VLAN tags (middle pane).
Figure 12-1: Multiple Network Interfaces
The Multiple Interface Table page allows you to configure these network interfaces. Each
row of the table defines a logical IP interface with the following attributes:
 Application type allowed on the interface:
• Control - call control signaling traffic (i.e., SIP)
• Media - RTP traffic
• Operations, Administration, Maintenance and Provisioning (OAMP) -
management (such as Web- and SNMP-based management)
 IP address and subnet mask represented by prefix length
 VLAN ID (if VLANs are enabled)
 Default Gateway - traffic from this interface destined to a subnet that does not meet
any of the routing rules, local or static routes, are forwarded to this gateway (as long
this application type is allowed on this interface.
 Primary and secondary DNS IP address (optional)
You can configure up to 16 interfaces, consisting of up to 15 Control and Media interfaces
and 1 OAMP interface.

This page also provides VLAN-related parameters for enabling VLANs and defining the
Native VLAN ID. This is the VLAN ID to which incoming, untagged packets are assigned.
You can also configure Quality of Service (QoS) by assigning VLAN priorities and
Differentiated Services (DiffServ) for the supported Class of Service (CoS). For configuring
Quality of Service (QoS), see 'Configuring the QoS Settings' on page 139.
Complementing the Multiple Interface table is the IP Routing table, which allows you to
define static routing rules for non-local hosts/subnets. For more information, see
'Configuring the IP Routing Table' on page 135.
Notes:
• Before adding IP network interfaces to the Multiple Interface table, see Multiple
Interface Table Configuration Rules on page 129 for the rules on configuring valid
IP network interfaces.
• When booting using BootP/DHCP protocols, an IP address is obtained from the
server. This address is used as the OAMP address for the initial session,
overriding the address configured in the Multiple Interface table. The address
configured for OAMP applications in this table becomes available only after you
save the configuration to the device's flash with a reset. This enables the device to
operate with a temporary address for initial management and configuration while
retaining the address configured in this table for deployment.
• The Multiple Interface table can also be configured using the table ini file
parameter, InterfaceTable (see 'Networking Parameters' on page 475).
 To configure IP network interfaces:

1. Open the IP Settings page (Configuration tab > VoIP menu > Network submenu > IP
Settings).
Figure 12-2: IP Settings Page (Single Network Interface)
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Note: The IP Settings page appears only in the following circumstances:

• Upon initial configuration (i.e., IP interfaces have never been configured).
• The Multiple Interface Table button has not been clicked in any previous access to
this page and only a single IP address has been configured.
• The device has been restored to default settings.
If you have clicked the Multiple Interface Table button or have configured multiple
interfaces using any other non-Web management tool, the Multiple Interface Table
page appears instead of the IP Settings page.
2. To access the Multiple Interface table so that you can configure multiple network
interfaces, click the Multiple Interface Table button, located under the Multiple
Interface Settings group; a confirmation message box appears:
Figure 12-3: Confirmation Message for Accessing the Multiple Interface Table
3. Click OK; the Multiple Interface Table page appears:
4. In the 'Add Index' field, enter the desired index number for the new interface, and then
click Add Index; the index row is added to the table.
5. Configure the interface according to the table below.
6. Click the Apply button; the interface is added to the table and the Done button
appears.
7. Click Done to validate the interface. If the interface is not valid (e.g., if it overlaps with
another interface in the table or if it does not adhere to the other rules as summarized
in 'Multiple Interface Table Configuration Summary and Guidelines' on page 129), a
warning message is displayed.
8. Save the changes to flash memory and reset the device (see 'Saving Configuration' on
page 366).
To view configured network interfaces that are currently active, click the IP Interface
Status Table button. For more information, see Viewing Active IP Interfaces on page
415.
Table 12-1: Multiple Interface Table Parameters Description
Table parameters
Index Table index row of the interface.
[InterfaceTable_Index]
The range is 0 to 15.

Web: Application Type Defines the applications allowed on the interface.

EMS: Application Types  [0] OAMP = Operations, Administration, Maintenance and
[InterfaceTable_ApplicationTypes] Provisioning (OAMP) applications (e.g., Web, Telnet,
SSH, and SNMP).
 [1] Media = Media (i.e., RTP streams of voice).
 [2] Control = Call Control applications (e.g., SIP).
 [3] OAMP + Media = OAMP and Media applications.
 [4] OAMP + Control = OAMP and Call Control
applications.
 [5] Media + Control = Media and Call Control
applications.
 [6] OAMP + Media + Control = All application types are
allowed on the interface.
Note: For valid configuration, see Multiple Interface Table
Configuration Rules on page 129.
Web/EMS: IP Address Defines the IPv4 IP address in dotted-decimal notation.
[InterfaceTable_IPAddress] Note: For valid configuration, see Multiple Interface Table
Web/EMS: Prefix Length Defines the prefix length of the related IP address. This is a
[InterfaceTable_PrefixLength] Classless Inter-Domain Routing (CIDR)-style representation
of a dotted-decimal subnet notation. The CIDR-style
representation uses a suffix indicating the number of bits
which are set in the dotted-decimal format. For example,
192.168.0.0/16 is synonymous with 192.168.0.0 and subnet
255.255.0.0. This CIDR lists the number of ‘1’ bits in the
subnet mask (i.e., replaces the standard dotted-decimal
representation of the subnet mask for IPv4 interfaces). For
example, a subnet mask of 255.0.0.0 is represented by a
prefix length of 8 (i.e., 11111111 00000000 00000000
00000000) and a subnet mask of 255.255.255.252 is
represented by a prefix length of 30 (i.e., 11111111
11111111 11111111 11111100).
The prefix length is a Classless Inter-Domain Routing (CIDR)
style presentation of a dotted-decimal subnet notation. The
CIDR-style presentation is the latest method for
interpretation of IP addresses. Specifically, instead of using
eight-bit address blocks, it uses the variable-length subnet
masking technique to allow allocation on arbitrary-length
prefixes.
The prefix length for IPv4 can range from 0 to 30.
Web/EMS: Gateway Defines the IP address of the default gateway for the
[InterfaceTable_Gateway] interface. When traffic is sent from this interface to an
unknown destination (i.e., not in the same subnet and not
defined for any static routing rule), it is forwarded to this
default gateway.
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Web/EMS: VLAN ID Defines a VLAN ID for the interface. Incoming traffic tagged
[InterfaceTable_VlanID] with this VLAN ID is routed to the corresponding interface.
Outgoing traffic from this interface is tagged with this VLAN
ID.
Notes:
 To enable VLANs, use the 'VLAN Mode' parameter.
 The device can use the discovery protocol, Link Layer
Discovery Protocol (LLDP) to obtain (over the Layer-2
data link layer) the VLAN ID for its OAMP interface. For
further information, see the EnableLLDP parameter.
 For valid configuration, see Multiple Interface Table
Web/EMS: Interface Name Defines a name for this interface. It is also displayed in
[InterfaceTable_InterfaceName] management interfaces (Web, CLI, and SNMP) for clarity
where it has no functional use.
The valid value is a string of up to 16 characters.
Web/EMS: Primary DNS Server IP (Optional) Defines the primary DNS server's IP address (in
address dotted-decimal notation), which is used for translating
[InterfaceTable_PrimaryDNSServerI domain names into IP addresses for the interface.
PAddress] By default, no IP address is defined.
Web/EMS: Secondary DNS Server IP (Optional) Defines the secondary DNS server's IP address
address (in dotted-decimal notation), which is used for translating
[InterfaceTable_SecondaryDNSServ domain names into IP addresses for the interface.
erIPAddress] By default, no IP address is defined.
General Parameters
Web/EMS: VLAN Mode Enables VLANs tagging (IEEE 802.1Q).
[VLANMode]  [0] Disable (default)
 [1] Enable
Notes:
 For this parameter to take effect, a device reset is
required.
 To operate with multiple network interfaces, VLANs must
be enabled.
 VLANs are available only when booting the device from
flash. When booting using BootP/DHCP protocols,
VLANs are disabled to allow easier maintenance access.
In this scenario, multiple network interface capabilities are
unavailable.

Web/EMS: Native VLAN ID Defines the Native VLAN ID. This is the VLAN ID to which
[VLANNativeVLANID] untagged incoming traffic is assigned. Outgoing packets sent
to this VLAN are sent only with a priority tag (VLAN ID = 0).
When the Native VLAN ID is equal to one of the VLAN IDs
listed in the Multiple Interface table (and VLANs are
enabled), untagged incoming traffic is considered as
incoming traffic for that interface. Outgoing traffic sent from
this interface is sent with the priority tag (tagged with VLAN
ID = 0).
When the Native VLAN ID is different to any value in the
'VLAN ID' column in the table, untagged incoming traffic is
discarded and all outgoing traffic is tagged.
The default Native VLAN ID is 1.
Note: If this parameter is not configured (i.e., default is 1)
and one of the interfaces has a VLAN ID set to 1, this
interface is still considered the ‘Native’ VLAN. If you do not
wish to have a ‘Native’ VLAN ID and want to use VLAN ID 1,
set this parameter to a value other than any VLAN ID in the
table.
12.2.1 Assigning NTP Services to Application Types

You can associate the Network Time Protocol (NTP) application with the OAMP or Control
application type. This is done using the EnableNTPasOAM ini file parameter.
12.2.2 Multiple Interface Table Configuration Rules

The Multiple Interface table configuration must adhere to the following rules:
 Each interface must have its own subnet. Configuring two interfaces with addresses in
the same subnet (e.g., 192.168.0.1/16 and 192.168.100.1/16) is invalid.
 Subnets of different interfaces must not overlap (i.e. 10.0.0.1/8 and 10.50.10.1/24 is
invalid); each interface must have its own address space.
 Each interface must be assigned a unique IP address (i.e., two interfaces may not
share the same address space, or even part of it).
 The prefix length replaces the dotted-decimal subnet mask presentation and must
have a value of 0-30 for IPv4 addresses.
 Only one OAMP interface must be configured and this must be an IPv4 address. This
OAMP interface can be combined with Media and Control.
 At least one Control interface must be configured with an IPv4 address.
 At least one Media interface must be configured with an IPv4 address.
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 The network interface types can be combined:

• Example 1:
♦ One combined OAMP-Media-Control interface with an IPv4 address
• Example 2:
♦ One OAMP interface with an IPv4 address
♦ One or more Control interfaces with IPv4 addresses
♦ One or more Media interfaces with IPv4 interfaces (with VLANs)
• Example 3:
♦ One combined OAMP-Media interface with an IPv4 address
♦ One or more combined Media-Control interfaces with IPv4 addresses.
 Each network interface can be configured with a Default Gateway. The address of the
Default Gateway must be in the same subnet as the associated interface. Additional
static routing rules can be configured in the IP Routing table.
 The interface name must be configured (mandatory) and unique for each interface,
and can include up to 16 characters.
 For IPv4 addresses, the 'Interface Mode' column must be set to IPv4 Manual (numeric
value 10).
 A different VLAN ID can be assigned to different network interface types. For example,
VLAN ID 100 for OAMP, VLAN ID 200 for Media, and VLAN ID 300 for Control.
However, you cannot assign the same Each network interface must be assigned a
unique VLAN ID.
 When configuring more than one IP interface of the same address family, VLANs must
be enabled.
 For network configuration to take effect, you must save the configuration to the
device's flash memory (burn) with a device reset.
Notes:
• When configuring the network interfaces and VLANs in the Multiple Interface table
using the Web interface, it is recommended to check that your configuration is
valid, by clicking the Done button in the Multiple Interface Table page.
• Upon device start up, the Multiple Interface table is parsed and passes
comprehensive validation tests. If any errors occur during this validation phase,
the device sends an error message to the Syslog server and falls back to a "safe
mode", using a single interface and no VLANs. Ensure that you view the Syslog
messages that the device sends in system startup to see if any errors occurred.
12.2.3 Troubleshooting the Multiple Interface Table

If any of the Multiple Interface table guidelines are violated, the device falls back to a "safe
mode" configuration, consisting of a single IPv4 interface without VLANs. For more
information on validation failures, consult the Syslog messages.
Validation failures may be caused by one of the following:
 One of the Application Types (OAMP, Control, or Media) are missing in the IPv4
interfaces.
 There are too many interfaces for Application Type, OAMP. There is only one
interface defined, but the 'Application Types' column is not set to OAMP + Media +
Control (numeric value 6).
 An IPv4 interface was defined with 'Interface Type' other than IPv4 Manual (10).
 Two interfaces have the same VLAN ID value while VLANs are enabled.

 Two interfaces have the same name.

 At least two interfaces share the same address space or subnet.
Apart from these validation errors, connectivity problems may be caused by one of the
following:
 Trying to access the device with VLAN tags while booting from BootP/DHCP.
 Trying to access the device with untagged traffic when VLANs are on and Native
VLAN is not configured properly.
 The IP Routing table is not configured properly.
12.2.4 Networking Configuration Examples

This section provides configuration examples of networking interfaces.
12.2.4.1 One VoIP Interface for All Applications

This example describes the configuration of a single VoIP interface for all applications:
1. Multiple Interface table: Configured with a single interface for OAMP, Media and
Control:
Table 12-2: Example of Single VoIP Interface in Multiple Interface Table
Application Prefix VLAN Interface

Index IP Address Default
Type Length ID Name
0 OAMP, Media & 192.168.85.14 16 192.168.0.1 1 myInterface

Control
2. VLANS are not required and the Native VLAN ID is irrelevant. Class of Service
parameters may have default values.
3. IP Routing table: Two routes are configured for directing traffic for subnet
201.201.0.0/16 to 192.168.0.2, and all traffic for subnet 202.202.0.0/16 to 192.168.0.3:
Table 12-3: Example of IP Routing Table
Destination IP
Prefix Length Gateway IP Address Metric
Address
201.201.0.0 16 192.168.0.2 1
202.202.0.0 16 192.168.0.3 1
4. The NTP applications remain with their default application types.
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12.2.4.2 VoIP Interface per Application Type

This example describes the configuration of three VoIP interfaces; one for each application
type:
1. Multiple Interface table: Configured with three interfaces, each for a different
application type, i.e., one for OAMP, one for Call Control, and one for RTP Media, and
each with a different VLAN ID and default gateway:
Table 12-4: Example of VoIP Interfaces per Application Type in Multiple Interface Table
Application Prefix VLAN

Index IP Address Gateway Interface Name
Type Length ID
0 OAMP 192.168.85.14 16 0.0.0.0 1 ManagementIF

1 Control 200.200.85.14 24 200.200.85.1 200 myControlIF
2 Media 211.211.85.14 24 211.211.85.1 211 myMediaIF
2. VLANs are required and the Native VLAN ID is the same VLAN ID as the
Management interface (configured for Index 0):
• 'VLAN Mode' is set to Enable.
• 'Native VLAN ID' field is set to "1".
3. IP Routing table: A routing rule is required to allow remote management from a host
in 176.85.49.0 / 24:
Table 12-5: Example IP Routing Table
Destination IP Gateway IP Interface

Prefix Length Metric
Address Address Name
176.85.49.0 24 192.168.0.1 1 -
4. All other parameters are set to their respective default values. The NTP application
remains with its default application types.

12.2.4.3 VoIP Interfaces for Combined Application Types

This example describes the configuration of multiple interfaces for the following
applications:
 One interface for the OAMP application.
 Interfaces for Call Control and Media applications.
1. Multiple Interface table:
Table 12-6: Example of VoIP Interfaces of Combined Application Types in Multiple Interface
Table
Application Prefix VLAN Interface

Index IP Address Gateway
Type Length ID Name
0 OAMP 192.168.85.14 16 192.168.0.1 1 Mgmt

1 Media & 200.200.85.14 24 200.200.85.1 201 MediaCntrl1
Control
2 Media & 200.200.86.14 24 200.200.86.1 201 MediaCntrl2
Control
2. VLANs are required and the Native VLAN ID is the same VLAN ID as the
Management interface (index 0):
• 'VLAN Mode' is set to Enable.
• 'Native VLAN ID' field is set to "1".
3. IP Routing table: A routing rule is required to allow remote management from a host
in 176.85.49.0/24:
Table 12-7: Example of IP Routing Table

Prefix Length Metric
176.85.49.0 24 192.168.0.10 1 -
4. The NTP application is configured (using the ini file) to serve as OAMP applications:
EnableNTPasOAM = 1
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12.2.4.4 VoIP Interfaces with Multiple Default Gateways

Below is a configuration example using default gateways per IP network interface. In this
example, the default gateway 200.200.85.1 is available for applications allowed on
Interface #1, whereas outgoing management traffic (originating on Interface #0) is never
directed to this default gateway.
Table 12-8: Configured Default Gateway Example

Type Length ID
0 OAMP 192.168.085.214 16 0.0.0.0 100 Mgmt

Media &
1 200.200.85.14 24 200.200.85.1 200 CntrlMedia
Control
A separate IP routing table enables you to configure static routing rules. Configuring the
following static routing rules enables OAMP applications to access peers on subnet
17.17.0.0 through the gateway 192.168.0.1.
Table 12-9: Separate Routing Table Example

Prefix Length Metric Status
17.17.0.0 16 192.168.0.1 1 0 Active

12.3 Configuring the IP Routing Table

The IP Routing Table page allows you to define up to 30 static IP routing rules for the
device. These rules can be associated with a network interface (defined in the Multiple
Interface table) and therefore, the routing decision is based on the source subnet/VLAN. If
not associated with an IP interface, the static IP rule is based on destination IP address.
Traffic destined to the subnet specified in the routing rule is re-directed to the defined
gateway, reachable through the specified interface. Before sending an IP packet, the
device searches this table for an entry that matches the requested destination
host/network. If such an entry is found, the device sends the packet to the indicated router.
If no explicit entry is found, the packet is sent to the default gateway.
 To configure static IP routing:

1. Open the IP Routing Table page (Configuration tab > VoIP menu > Network
submenu > IP Routing Table).
Figure 12-4: IP Routing Table Page
2. In the Add a new table entry table, add a new static routing rule according to the
parameters described in the table below.
3. Click Add New Entry; the new routing rule is added to the IP routing table.
To delete a routing rule from the table, select the 'Delete Row' check box corresponding to
the required routing rule, and then click Delete Selected Entries.
Notes:
• You can delete only inactive routing rules.
• The IP Routing table can also be configured using the table ini file parameter,
StaticRouteTable.
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Table 12-10: IP Routing Table Description
Destination IP Address Defines the IP address of the destination host/network. The

[StaticRouteTable_Destination] destination can be a single host or a whole subnet, depending
on the Prefix Length configured for this routing rule.
Prefix Length Defines the Classless Inter-Domain Routing (CIDR)-style
[StaticRouteTable_PrefixLength] representation of a dotted-decimal subnet notation, of the
destination host/network. The CIDR-style representation uses
a suffix indicating the number of bits that are set in the dotted-
decimal format. For example, 16 is synonymous with subnet
255.255.0.0.
The address of the host/network you want to reach is determined by an AND operation that is applied
to the fields 'Destination IP Address' and 'Prefix Length'. For example, to reach the network 10.8.x.x,
enter 10.8.0.0 in the 'Destination IP Address' field and 16 in the 'Prefix Length'. As a result of the AND
operation, the value of the last two octets in the 'Destination IP Address' field is ignored. To reach a
specific host, enter its IP address in the 'Destination IP Address' field and 32 in the 'Prefix Length'
field.
Gateway IP Address Defines the IP address of the router (next hop) used for traffic
[StaticRouteTable_Gateway] destined to the subnet/host as defined in the 'Destination IP
Address' / 'Prefix Length' field.
Note: The Gateway address must be in the same subnet as
the IP address of the interface over which you configure this
static routing rule.
Metric Defines the number of hops needed to reach the specified
destination.
Note: The recommended value for this parameter is 1. This
parameter must be set to a number greater than 0 for the
routing rule to be valid. Routing entries with Hop Count equals
0 are local routes set automatically by the device.
Interface Name Assigns a network interface through which the 'Gateway IP
[StaticRouteTable_InterfaceName] Address' is reached. This is the string value as configured for
the network interface in the 'Interface Name' field of the
Multiple Interface table (see 'Configuring IP Network
Interfaces' on page 124).
Note: The IP address of the 'Gateway IP Address' field must
be in the same subnet as this interface's IP address.
Status Read-only field displaying the status of the static IP route:
 "Active" - routing rule is used ny the device.
 "Inactive" - routing rule is not applied. When the destination
IP address is not on the same segment with the next hop or
the interface does not exist, the route state changes to
"Inactive".

12.3.1 Interface Column

This example describes the configuration of static IP routing rules.
1. Configure network interfaces in the Multiple Interface table, as shown below:
Table 12-11: Configured Network Interfaces in Multiple Interface Table

Type Length ID
0 OAMP 192.168.0.2 16 192.168.0.1 501 Mng

Media &
1 10.32.174.50 24 10.32.174.1 2012 MediaCntrl
Control
2 Media 10.33.174.50 24 10.33.174.1 2013 Media1
3 Control 10.34.174.50 24 10.34.174.1 2014 Cntrl1
2. Configure static IP Routing rules in the IP Routing table, as shown below:

Table 12-12: Configured Static IP Routing Rules in IP Routing Table
Destination IP Gateway IP
Prefix Length Metric Interface Name
Address Address
10.31.174.0 24 192.168.11.1 1 Mng

174.96.151.15 24 10.32.174.12 1 MediaCntrl
10.35.174.0 24 10.34.174.240 1 Cntrl1
Note that the IP address configured in the 'Gateway IP Address' field (i.e., next hop)
must reside on the same subnet as the IP address of the associated network interface
that is specified in the 'Interface Name' field.
12.3.2 Routing Table Configuration Summary and Guidelines

The Routing table configurations must adhere to the following rules:
 Up to 30 different static routing rules can be configured.
 The 'Prefix Length' replaces the dotted-decimal subnet mask presentation. This
column must have a value of 0-31 for IPv4 interfaces.
 The 'Gateway IP Address' field must be on the same subnet as the IP address of the
associated interface specified in the 'Interface Name' field.
 The 'Metric; field must be set to 1.
 For the configuration settings to take effect, you must reset the device with a "burn" to
flash memory.
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12.3.3 Troubleshooting the Routing Table

When adding a new static routing rule, the added rule passes a validation test. If errors are
found, the routing rule is rejected and is not added to the IP Routing table. Failed routing
validations may result in limited connectivity (or no connectivity) to the destinations
specified in the incorrect routing rule. For any error found in the Routing table or failure to
configure a routing rule, the device sends a notification message to the Syslog server
reporting the problem.
Common routing rule configuration errors may include the following:
 The IP address specified in the 'Gateway IP Address' field is unreachable from the
interface specified in the 'Interface Name' field.
 The same destination is configured in two different routing rules.
 More than 30 routing rules have been configured.
Note: If an IP routing rule is required to access OAMP applications (for remote

management, for example) and the route is not configured correctly, the route is not
added and the device is not accessible remotely. To restore connectivity, the device
must be accessed locally from the OAMP subnet and the required routes be
configured.

12.4 Configuring Quality of Service

The QoS Settings page is used for configuring the Layer-2 and Layer-3 Quality of Service
(QoS) parameters. Differentiated Services (DiffServ) is an architecture providing different
types or levels of service for IP traffic. DiffServ (according to RFC 2474), prioritizes certain
traffic types based on their priority, thereby, accomplishing a higher-level QoS at the
expense of other traffic types. By prioritizing packets, DiffServ routers can minimize
transmission delays for time-sensitive packets such as VoIP packets.
You can assign different VLAN priorities (IEEE 802.1p) and DiffServ to the supported Class
of Service (CoS):
 Network Service class – network control traffic (ICMP, ARP)
 Premium Media service class – used for RTP media traffic
 Premium Control service class – used for call control (i.e., SIP) traffic
 Gold service class – used for streaming applications
 Bronze service class – used for OAMP applications
The Layer-2 QoS parameters define the values for the 3 priority bits in the VLAN tag of
frames related to a specific service class (according to the IEEE 802.1p standard). The
Layer-3 QoS parameters define the values of the DiffServ field in the IP Header of the
frames related to a specific service class.
The mapping of an application to its CoS and traffic type is shown in the table below:
Table 12-13: Traffic/Network Types and Priority
Application Traffic / Network Types Class-of-Service (Priority)
Debugging interface Management Bronze

Telnet Management Bronze
DHCP Management Network
Web server (HTTP) Management Bronze
SNMP GET/SET Management Bronze
Web server (HTTPS) Management Bronze
IPSec IKE Determined by the service Determined by the service
RTP traffic Media Premium media
RTCP traffic Media Premium media
T.38 traffic Media Premium media
SIP Control Premium control
SIP over TLS (SIPS) Control Premium control
Syslog Management Bronze
ICMP Management Determined by the initiator of the
request
ARP listener Determined by the initiator of the Network
request
SNMP Traps Management Bronze
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Application Traffic / Network Types Class-of-Service (Priority)
DNS client Varies according to DNS settings: Depends on traffic type:

 OAMP  Control: Premium Control
 Control  Management: Bronze
NTP Varies according to the interface Depends on traffic type:

type associated with NTP (see  Control: Premium control
'Assigning NTP Services to
 Management: Bronze
Application Types' on page 129):
 OAMP
 Control
NFS NFSServers_VlanType in the Gold
NFSServers table
 To configure QoS:
1. Open the QoS Settings page (Configuration tab > VoIP menu > Network submenu >
QoS Settings).
2. Configure the QoS parameters as required.

4. Save the changes to flash memory (see 'Saving Configuration' on page 366).

12.5 Disabling ICMP Redirect Messages

You can configure the device's handling of ICMP Redirect messages. These messages
can either be rejected (ignored) or permitted.
Note: You can also configure this feature using the ini file parameter
DisableICMPRedirects (see 'Routing Parameters' on page 477).
 To configure the handling of ICMP Redirect messages:

1. Open the Network Settings page (Configuration tab > VoIP menu > Network
submenu > Network Settings).
Figure 12-5: Disabling ICMP Redirect in Network Settings Page
2. From the 'Disable ICMP Redirects' drop-down list, select the required option.
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12.6 DNS
You can use the device's embedded domain name server (DNS) or an external, third-party
DNS to translate domain names into IP addresses. This is useful if domain names are used
as the destination in call routing. The device supports the configuration of the following
DNS types:
 Internal DNS table - see 'Configuring the Internal DNS Table' on page 142
 Internal SRV table - see 'Configuring the Internal SRV Table' on page 143
12.6.1 Configuring the Internal DNS Table

The Internal DNS Table page, similar to a DNS resolution, translates up to 20 host
(domain) names into IP addresses. This functionality can be used when a domain name
(FQDN) is configured as an IP destination for Tel-to-IP routing in the Tel to IP Routing. Up
to four different IP addresses can be assigned to the same host name. This is typically
needed for alternative Tel-to-IP call routing.
Notes:
• The device initially attempts to resolve a domain name using the Internal DNS
table. If the domain name isn't listed in the table, the device performs a DNS
resolution using an external DNS server for the related IP network interface,
configured in the Multiple Interface table (see 'Configuring IP Network Interfaces'
on page 124).
• You can also configure the DNS table using the table ini file parameter, DNS2IP
(see 'DNS Parameters' on page 482).
 To configure the internal DNS table:

1. Open the Internal DNS Table page (Configuration tab > VoIP menu > Network
submenu > DNS submenu > Internal DNS Table).
Figure 12-6: Internal DNS Table - Add Record Dialog Box
3. Configure the DNS rule, as required. For a description of the parameters, see the
table below.
4. Click Submit; the DNS rule is added to the table.

Table 12-14: Internal DNS Table Parameter Description
Domain Name Defines the host name to be translated.

[Dns2Ip_DomainName] The valid value is a string of up to 31 characters.
First IP Address Defines the first IP address (in dotted-decimal format notation) to
[Dns2Ip_FirstIpAddress] which the host name is translated.
Second IP Address Defines the second IP address (in dotted-decimal format notation)
[Dns2Ip_SecondIpAddress] to which the host name is translated.
Third IP Address Defines the third IP address (in dotted-decimal format notation) to
[Dns2Ip_ThirdIpAddress] which the host name is translated.
Fourth IP Address Defines the fourth IP address (in dotted-decimal format notation) to
[Dns2Ip_FourthIpAddress] which the host name is translated.
12.6.2 Configuring the Internal SRV Table

The Internal SRV Table page resolves host names to DNS A-Records. Three different A-
Records can be assigned to each host name, where each A-Record contains the host
name, priority, weight, and port.
Notes:
• If the Internal SRV table is configured, the device initially attempts to resolve a
domain name using this table. If the domain name isn't found, the device performs
a Service Record (SRV) resolution using an external DNS server configured in the
Multiple Interface table (see 'Configuring IP Network Interfaces' on page 124).
• The Internal SRV table can also be configured using the table ini file parameter,
SRV2IP (see 'DNS Parameters' on page 482).
 To configure the Internal SRV table:

1. Open the Internal SRV Table page (Configuration tab > VoIP menu > Network
submenu > DNS submenu > Internal SRV Table).
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Figure 12-7: Internal SRV Table Page
3. Configure the SRV rule, as required. For a description of the parameters, see the table
below.
4. Click Submit; the SRV rule is added to the table.
Table 12-15: Internal SRV Table Parameter Description
Domain Name Defines the host name to be translated.

[Srv2Ip_InternalDomain] The valid value is a string of up to 31 characters.
Transport Type Defines the transport type.
[Srv2Ip_TransportType]  [0] UDP (default)
 [1] TCP
 [2] TLS
DNS Name (1-3) Defines the first, second or third DNS A-Record to which the host
[Srv2Ip_Dns1/2/3] name is translated.
Priority (1-3) Defines the priority of the target host. A lower value means that it is
[Srv2Ip_Priority1/2/3] more preferred.
Weight (1-3) Defines a relative weight for records with the same priority.
[Srv2Ip_Weight1/2/3]
Port (1-3) Defines the TCP or UDP port on which the service is to be found.
[Srv2Ip_Port1/2/3]

12.7 Configuring NFS Settings

Network File System (NFS) enables the device to access a remote server's shared files
and directories and to handle them as if they're located locally. The device can use NFS to
load cmp, ini, and auxiliary files through the Automatic Update mechanism (see 'Automatic
Update' on page 389).
You can configure up to 16 different NFS file systems. As a file system, the NFS is
independent of machine types, operating systems and network architectures. Note that an
NFS file server can share multiple file systems. There must be a separate row for each
remote file system shared by the NFS file server that needs to be accessed by the device.
 To add remote NFS file systems:

2. Under the 'NFS Settings' group, click the NFS Table button; the NFS Table page
appears.
3. Click the Add button; the Add Record dialog box appears:
Figure 12-8: Add Record Dialog Box for NFS
4. Configure the NFS parameters according to the table below.

5. Click the Submit button; the remote NFS file system is immediately applied, which
can be verified by the appearance of the 'NFS mount was successful' message in the
Syslog server.
Notes:
• To avoid terminating current calls, a row must not be deleted or modified while the
device is currently accessing files on that remote NFS file system.
• The combination of 'Host Or IP' and 'Root Path' must be unique for each row in the
table. For example, the table must include only one row with a Host/IP of
192.168.1.1 and Root Path of /audio.
• The NFS table can also be configured using the table ini file parameter
NFSServers (see 'NFS Parameters' on page 481)
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Table 12-16: NFS Settings Parameters
Index The row index of the remote file system.

The valid range is 1 to 16.
Host Or IP The domain name or IP address of the NFS server. If a domain name is
[NFSServers_HostOrIP] provided, a DNS server must be configured.
Root Path Path to the root of the remote file system in the format: /[path]. For
[NFSServers_RootPath] example, '/audio'.
NFS Version NFS version used to access the remote file system.
[NFSServers_NfsVersion]  [2] NFS Version 2
 [3] NFS Version 3 (default)
Authentication Type Authentication method used for accessing the remote file system.
[NFSServers_AuthType]  [0] Null
 [1] Unix (default)
User ID User ID used in authentication when using Unix.
[NFSServers_UID] The valid range is 0 to 65537. The default is 0.
Group ID Group ID used in authentication when using Unix.
[NFSServers_GID] The valid range is 0 to 65537. The default is 1.
VLAN Type The VLAN type for accessing the remote file system.
[NFSServers_VlanType]  [0] OAM
 [1] MEDIA (default)
Note: This parameter applies only if VLANs are enabled or if Multiple
IPs is configured (see 'Configuring IP Network Interfaces' on page 124).

12.8 Network Address Translation Support

Network Address Translation (NAT) is a mechanism that maps internal IP addresses (and
ports) used within a private network to global IP addresses and vice versa, providing
transparent routing to end hosts. The primary advantages of NAT include (1) reduction in
the number of global IP addresses required in a private network (global IP addresses are
only used to connect to the Internet) and (2) better network security by hiding the internal
architecture.
The design of SIP creates a problem for VoIP traffic to pass through NAT. SIP uses IP
addresses and port numbers in its message body. However, the NAT server is unable to
modify the SIP messages and thus, can’t change local addresses to global addresses.
This section discusses the device's solutions for overcoming NAT traversal issues.
12.8.1 Device Located behind NAT

Two different streams traverse through NAT - signaling and media. A device located
behind a NAT, that initiates a signaling path has problems receiving incoming signaling
responses as they are blocked by the NAT server. Therefore, the initiating device must
inform the receiving device where to send the media. To resolve this NAT problem, the
following solutions are provided by the device, listed in priority of the selected method used
by the device:
a. If configured, uses an external STUN server to assign a NAT address to all interfaces
- see .Configuring STUN on page 148.
b. If configured, uses the single Static NAT IP address for all interfaces - see 'Configuring
a Static NAT IP Address for All Interfaces' on page 149.
If NAT is not configured by any of the above-mentioned methods, the device sends the
packet according to its IP address configured in the Multiple Interface table.
The figure below illustrates the NAT problem faced by the SIP networks where the device
is located behind a NAT:
Figure 12-9: Device behind NAT and NAT Issues
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12.8.1.1 Configuring STUN

Simple Traversal of UDP through NATs (STUN), based on RFC 3489 is a client / server
protocol that solves most of the NAT traversal problems. The STUN server operates in the
public Internet and the STUN clients are embedded in end-devices located behind NAT.
STUN is used for signaling and the media streams. STUN works with many existing NAT
types and does not require any special behavior.
STUN enables the device to discover the presence (and types) of NATs and firewalls
located between it and the public Internet. It provides the device with the capability to
determine the public IP address and port allocated to it by the NAT. This information is later
embedded in outgoing SIP / SDP messages and enables remote SIP user agents to reach
the device. It also discovers the binding lifetime of the NAT - the refresh rate necessary to
keep NAT ‘pinholes’ open.
On startup, the device sends a STUN Binding Request. The information received in the
STUN Binding Response (IP address:port) is used for SIP signaling. This information is
updated every user-defined period (NATBindingDefaultTimeout).
At the beginning of each call and if STUN is required (i.e., not an internal NAT call), the
media ports of the call are mapped. The call is delayed until the STUN Binding Response
(that includes a global IP:port) for each media (RTP, RTCP and T.38) is received.
Notes:
• STUN is applicable only to UDP connections (not TCP and TLS).
• STUN can’t be used when the device is located behind a symmetric NAT.
• Use either the STUN server IP address (STUNServerPrimaryIP) or domain name
(STUNServerDomainName) method, with priority to the first one.
 To enable STUN:
Figure 12-10: STUN Parameters in Application Settings Page
2. From the 'Enable STUN' (EnableSTUN) drop-down list, select Enable to enable the
STUN feature.
3. Configure the STUN server address using one of the following methods:
• Define the IP address of the primary and secondary (optional) STUN servers,
using the 'STUN Server Primary IP' field (STUNServerPrimaryIP) and 'STUN
Server Secondary IP' field. If the primary STUN server is unavailable, the device
attempts to communicate with the second server.
• Define the domain name of the STUN server using the ini file parameter,
STUNServerDomainName. The STUN client retrieves all STUN servers with an
SRV query to resolve this domain name to an IP address and port, sorts the
server list, and uses the servers according to the sorted list.
4. Configure the default NAT binding lifetime (in secondsUse) using the ini file
parameter, NATBindingDefaultTimeout. STUN refreshes the binding information after
this time expires.

12.8.1.2 Configuring a Static NAT IP Address for All Interfaces

You can configure a global (public) IP address of the router to enable static NAT between
the device and the Internet for all network interfaces. Thus, the device replaces the source
IP address for media of all outgoing SIP messages sent on any of its network interfaces to
this public IP address.
Note: The NAT IP address can also be configured using the ini file parameter,
StaticNATIP.
 To configure a single static NAT IP address for all interfaces:

1. Open the SIP General Parameters page (Configuration tab > VoIP menu > SIP
Definitions submenu > General Parameters).
Figure 12-11: Configuring Static NAT IP Address in SIP General Parameters Page
2. In the 'NAT IP Address' field, enter the NAT IP address in dotted-decimal notation.
3. Click Submit.
4. Save the setting to the device's flash memory with a device reset (see 'Saving
12.8.2 Remote UA behind NAT

If the remote User Agent with which the device needs to communicate with is located
behind NAT, the device can resolve the problem of activating the RTP/RTCP/T.38 streams
to an invalid IP address / UDP port.
To resolve this NAT traversal issue, the device offers the following features:
 First Incoming Packet Mechanism - see 'First Incoming Packet Mechanism' on page
150
 RTP No-Op packets according to the avt-rtp-noop draft - see 'No-Op Packets' on page
150
The figure below illustrates a typical network architecture where the remote UA is located
behind NAT:
Figure 12-12: Remote UA behind NAT
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12.8.2.1 First Incoming Packet Mechanism

If the remote device resides behind a NAT device, it’s possible that the device can activate
the RTP/RTCP/T.38 streams to an invalid IP address / UDP port. To avoid such cases, the
device automatically compares the source address of the first received incoming
RTP/RTCP/T.38 stream with the IP address and UDP port of the remote device when the
session was initially opened. If the two are not identical, then the destination IP address of
the outgoing RTP packets is set to the source IP address of the first incoming packet. The
RTP, RTCP and T.38 can thus have independent destination IP addresses and UDP ports.
 To enable NAT resolution using the First Incoming Packet mechanism:

1. Open the General Media Settings page (Configuration tab > VoIP menu > Media >
General Media Settings).
2. Set the 'NAT Traversal' parameter to Enable.
3. Click Submit.
The EnableIpAddrTranslation and EnableUdpPortTranslation parameters allow you to
specify the type of compare operation that occurs on the first incoming packet. To compare
only the IP address, set EnableIpAddrTranslation to 1, and EnableUdpPortTranslation to 0.
In this case, if the first incoming packet arrives with only a difference in the UDP port, the
sending addresses won’t change. If both the IP address and UDP port need to be
compared, then both parameters need to be set to 1.
12.8.2.2 No-Op Packets

The device's No-Op packet support can be used to verify Real-Time Transport Protocol
(RTP) and T.38 connectivity, and to keep NAT bindings and Firewall pinholes open. The
No-Op packets are available for sending in RTP and T.38 formats.
You can control the activation of No-Op packets by using the ini file parameter
NoOpEnable. If No-Op packet transmission is activated, you can control the time interval in
which No-Op packets are sent in the case of silence (i.e., no RTP or T.38 traffic). This is
done using the ini file parameter NoOpInterval. For a description of the RTP No-Op ini file
parameters, see 'Networking Parameters' on page 475.
 RTP No-Op: The RTP No-Op support complies with IETF Internet-Draft draft-wing-
avt-rtp-noop-03 ("A No-Op Payload Format for RTP"). This IETF document defines a
No-Op payload format for RTP. The draft defines the RTP payload type as dynamic.
You can control the payload type with which the No-Op packets are sent. This is
performed using the RTPNoOpPayloadType ini parameter (see 'Networking
Parameters' on page 475). The default payload type is 120.
 T.38 No-Op: T.38 No-Op packets are sent only while a T.38 session is activated. Sent
packets are a duplication of the previously sent frame (including duplication of the
sequence number).
Note: Receipt of No-Op packets is always supported.

12.9 Robust Receipt of Media Streams

The “robust-media” mechanism is an AudioCodes proprietary mechanism to filter out
unwanted media (i.e., RTP, RTCP, and T.38) streams that are sent to the same port
number on the device. In practice, the media RTP/RTCP ports may receive additional
multiple unwanted media streams as result of traces of previous calls, call control errors, or
deliberate attacks. When more than one media stream reaches the device on the same
port number, the “robust-media” mechanism detects the valid media stream and ignores
the rest.
12.10 Multiple Routers Support

Multiple routers support is designed to assist the device when it operates in a multiple
routers network. The device learns the network topology by responding to Internet Control
Message Protocol (ICMP) redirections and caches them as routing rules (with expiration
time).
When a set of routers operating within the same subnet serve as devices to that network
and intercommunicate using a dynamic routing protocol, the routers can determine the
shortest path to a certain destination and signal the remote host the existence of the better
route. Using multiple router support, the device can utilize these router messages to
change its next hop and establish the best path.
Note: Multiple Routers support is an integral feature that doesn’t require

configuration.
12.11 IP Multicasting
The device supports IP Multicasting level 1, according to RFC 2236 (i.e., IGMP version 2)
for RTP channels. The device is capable of transmitting and receiving multicast packets.
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User's Manual 13. Security
13 Security
This section describes the VoIP security-related configuration.
13.1 Configuring Firewall Settings

The device provides an internal firewall that enables you to configure network traffic
filtering rules (access list). You can add up to 50 firewall rules. The access list offers the
following firewall possibilities:
 Block traffic from known malicious sources
 Allow traffic only from known "friendly" sources, and block all other traffic
 Mix allowed and blocked network sources
 Limit traffic to a user-defined rate (blocking the excess)
 Limit traffic to specific protocols, and specific port ranges on the device
For each packet received on the network interface, the table is scanned from top to bottom
until the first matching rule is found. This rule can either permit (allow) or deny (block) the
packet. Once a rule in the table is located, subsequent rules further down the table are
ignored. If the end of the table is reached without a match, the packet is accepted.
Notes:
• This firewall applies to a very low-level network layer and overrides your other
security-related configuration. Thus, if you have configured higher-level security
features (e.g., on the Application level), you must also configure firewall rules to
permit this necessary traffic. For example, if you have configured IP addresses to
access the Web and Telnet interfaces in the Web Access List (see 'Configuring
Web and Telnet Access List' on page 75), you must configure a firewall rule that
permits traffic from these IP addresses.
• Only Security Administrator users or Master users can configure firewall rules.
• Setting the 'Prefix Length' field to 0 means that the rule applies to all packets,
regardless of the defined IP address in the 'Source IP' field. Therefore, it is highly
recommended to set this parameter to a value other than 0.
• It is recommended to add a rule at the end of your table that blocks all traffic and
to add firewall rules above it that allow required traffic (with bandwidth limitations).
To block all traffic, use the following firewall rule:
- Source IP: 0.0.0.0
- Prefix Length: 0 (i.e., rule matches all IP addresses)
- Start Port - End Port: 0-65535
- Protocol: Any
- Action Upon Match: Block
• You can also configure the firewall settings using the table ini file parameter,
AccessList (see 'Security Parameters' on page 504).
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 To add firewall rules:

1. Open the Firewall Settings page (Configuration tab > VoIP menu > Security
submenu > Firewall Settings).
2. Click the Add button; the following dialog box appears:
Figure 13-1: Firewall Settings Page - Add Record
3. Configure the firewall parameters, as required. For a description of the parameters,

see the table below.
4. Click Submit to add the new firewall rule to the table.
5. Reset the device to activate the rules.
The table below provides an example of configured firewall rules:
Table 13-1: Firewall Rule Examples
Value per Rule

Parameter
1 2 3 4 5
Source IP 12.194.231.76 12.194.230.7 0.0.0.0 192.0.0.0 0.0.0.0
Prefix Length 16 16 0 8 0
Start Port and End
0-65535 0-65535 0-65535 0-65535 0-65535
Port
Protocol Any Any icmp Any Any
Use Specific
Enable Enable Disable Enable Disable
Interface
Interface Name WAN WAN None Voice-Lan None
Byte Rate 0 0 40000 40000 0

Value per Rule

Parameter
1 2 3 4 5
Burst Bytes 0 0 50000 50000 0
Action Upon Match Allow Allow Allow Allow Block
The firewall rules in the above configuration example do the following:

 Rules 1 and 2: Typical firewall rules that allow packets ONLY from specified IP
addresses (e.g., proxy servers). Note that the prefix length is configured.
 Rule 3: A more "advanced” firewall rule - bandwidth rule for ICMP, which allows a
maximum bandwidth of 40,000 bytes/sec with an additional allowance of 50,000 bytes.
If, for example, the actual traffic rate is 45,000 bytes/sec, then this allowance would be
consumed within 10 seconds, after which all traffic exceeding the allocated 40,000
bytes/sec is dropped. If the actual traffic rate then slowed to 30,000 bytes/sec, the
allowance would be replenished within 5 seconds.
 Rule 4: Allows traffic from the LAN voice interface and limits bandwidth.
 Rule 5: Blocks all other traffic.
Table 13-2: Internal Firewall Parameters
Source IP Defines the IP address (or DNS name) or a specific host name of the
[AccessList_Source_IP] source network (i.e., from where the incoming packet is received).
Source Port Defines the source UDP/TCP ports (of the remote host) from where
[AccessList_Source_Port] packets are sent to the device.
Note: When set to 0, this field is ignored and any source port
matches the rule.
Prefix Length (Mandatory) Defines the IP network mask - 32 for a single host or
[AccessList_PrefixLen] the appropriate value for the source IP addresses.
 A value of 8 corresponds to IPv4 subnet class A (network mask of
255.0.0.0).
 A value of 16 corresponds to IPv4 subnet class B (network mask
of 255.255.0.0).
 A value of 24 corresponds to IPv4 subnet class C (network mask
of 255.255.255.0).
The IP address of the sender of the incoming packet is trimmed in
accordance with the prefix length (in bits) and then compared to the
parameter ‘Source IP’.
The default is 0 (i.e., applies to all packets). You must change this
value to any of the above options.
Note: A value of 0 applies to all packets, regardless of the defined IP
address. Therefore, you must set this parameter to a value other
than 0.
Start Port Defines the destination UDP/TCP start port (on this device) to where
[AccessList_Start_Port] packets are sent.
Note: When the protocol type isn't TCP or UDP, the entire range
must be provided.
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End Port Defines the destination UDP/TCP end port (on this device) to where
[AccessList_End_Port] packets are sent.
Note: When the protocol type isn't TCP or UDP, the entire range
must be provided.
Protocol Defines the protocol type (e.g., UDP, TCP, ICMP, ESP or 'Any') or
[AccessList_Protocol] the IANA protocol number in the range of 0 (Any) to 255.
Note: This field also accepts the abbreviated strings 'SIP' and
'HTTP'. Specifying these strings implies selection of the TCP or UDP
protocols, and the appropriate port numbers as defined on the
device.
Use Specific Interface Determines whether you want to apply the rule to a specific network
[AccessList_Use_Specific_I interface defined in the Multiple Interface table (i.e., packets received
nterface] from that defined in the Source IP field and received on this network
interface):
 [0] Disable (default)
 [1] Enable
Notes:
 If enabled, then in the 'Interface Name' field (described below),
select the interface to which the rule is applied.
 If disabled, then the rule applies to all interfaces.
Interface Name Defines the network interface to which you want to apply the rule.
[AccessList_Interface_ID] This is applicable if you enabled the 'Use Specific Interface' field.
The list displays interface names as defined in the Multiple Interface
table in 'Configuring IP Network Interfaces' on page 124.
Packet Size Defines the maximum allowed packet size.
[AccessList_Packet_Size] The valid range is 0 to 65535.
Note: When filtering fragmented IP packets, this field relates to the
overall (re-assembled) packet size, and not to the size of each
fragment.
Byte Rate Defines the expected traffic rate (bytes per second), i.e., the allowed
[AccessList_Byte_Rate] bandwidth for the specified protocol. In addition to this field, the
'Burst Bytes' field provides additional allowance such that momentary
bursts of data may utilize more than the defined byte rate, without
being interrupted.
For example, if 'Byte Rate' is set to 40000 and 'Burst Bytes' to
50000, then this implies the following: the allowed bandwidth is
40000 bytes/sec with extra allowance of 50000 bytes; if, for example,
the actual traffic rate is 45000 bytes/sec, then this allowance would
be consumed within 10 seconds, after which all traffic exceeding the
allocated 40000 bytes/sec is dropped. If the actual traffic rate then
slowed to 30000 bytes/sec, then the allowance would be replenished
within 5 seconds.
Burst Bytes Defines the tolerance of traffic rate limit (number of bytes).
[AccessList_Byte_Burst] The default is 0.
Action Upon Match Defines the firewall action to be performed upon rule match.
[AccessList_Allow_Type]  "Allow" = (Default) Permits these packets
 "Block" = Rejects these packets

Match Count (Read-only) Displays the number of packets accepted or rejected by

[AccessList_MatchCount] the rule.
13.2 Configuring 802.1x Settings

The 802.1x Settings page is used to configure IEEE 802.1X Ethernet security. The device
can function as an IEEE 802.1X supplicant. IEEE 802.1X is a standard for port-level
security on secure Ethernet switches; when a device is connected to a secure port, no
traffic is allowed until the identity of the device is authenticated.
A typical 802.1X deployment consists of an Authenticator (secure LAN switch), an Access
Server (e.g. RADIUS), and one or more supplicants. The Authenticator blocks all traffic on
the secure port by default and communicates with the supplicant via EAP-over-LAN
frames. The supplicant provides credentials which are transmitted to the Access Server. If
the Access Server determines that the credentials are valid, it instructs the Authenticator to
authorize traffic on the secure port.
The device supports the following Extensible Authentication Protocol (EAP) variants:
 MD5-Challenge (EAP-MD5): Authentication is done with a user-defined 802.1X
username and password.
 Protected EAP (PEAPv0 with EAP-MSCHAPv2): Authentication is done with a user-
defined 802.1X username and password, however, the protocol is MSCHAPv2 over
an encrypted TLS tunnel.
 EAP-TLS: The device's certificate is used to establish a mutually-authenticated TLS
session with the Access Server. This requires prior configuration of the server
certificate and root CA. The user-defined 802.1X username is used to identify the
device, however, the 802.1X password is ignored.
For a description of the 802.1X parameters, see 802.1X Parameters on page 514.
 To configure the 802.1x parameters:

1. Open the 802.1x Settings page (Configuration tab > VoIP menu > Security submenu
> 802.1x Settings).
Figure 13-2: 8021x Settings Page
2. Configure the parameters as required, and then click Submit.
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13.3 Configuring General Security Settings

The General Security Settings page is used to configure various security features. For a
description of the parameters appearing on this page, refer 'Configuration Parameters
 To configure the general security parameters:

Security submenu > General Security Settings).

4. To save the changes to flash memory, refer to 'Saving Configuration' on page 366.

13.4 IPSec and Internet Key Exchange

IP security (IPSec) and Internet Key Exchange (IKE) protocols are part of the IETF
standards for establishing a secured IP connection between two applications (also referred
to as peers). Providing security services at the IP layer, IPSec and IKE are transparent to
IP applications. IPSec and IKE are used together to provide security for control and
management (e.g., SNMP and Web) protocols, but not for media (i.e., RTP, RTCP and
T.38).
IKE is used to obtain the Security Associations (SA) between peers (the device and the
application it’s trying to contact). The SA contains the encryption keys and profile used by
IPSec to encrypt the IP stream. IKE negotiation comprises the following two phases:
 Main Mode (creates a secured channel for the Quick mode by obtaining a "master"
encryption key, without any prior keys, and authenticates the peers to each other):
• SA negotiation: The peers negotiate their capabilities using up to four proposals.
Each proposal includes the Encryption method, Authentication algorithm, and the
Diffie-Hellman (DH) group. The master key’s lifetime is also negotiated.
• Key exchange (DH): The DH protocol creates the master key. DH requires both
peers to agree on certain mathematical parameters, known as the "group".
• Authentication: The two peers authenticate one another using a pre-shared key
configured in the IP Security Associations Table or by using certificate-based
authentication.
 Quick Mode (creates the encrypted IPSec tunnel once initial security is set up):
• SA negotiation: An IPSec SA is created by negotiating encryption and
authentication capabilities using the same proposal mechanism as in Main mode.
• Key exchange: A symmetrical key is created for encrypting IPSec traffic; the
peers communicate with each other in encrypted form, secured by the previously
negotiated "master" key.
IKE specifications summary:
 Authentication methods: pre-shared key or certificate-based authentication
 Main mode supported for IKE Phase 1
 DH group 1 or group 2
 Encryption algorithms: Data Encryption Standard (DES), Advanced Encryption
Standard (AES), and 3DES
 Hash algorithms: SHA1 and MD5
IPSec is responsible for securing the IP traffic. This is accomplished by using the
Encapsulation Security Payload (ESP) protocol to encrypt (and decrypt) the IP payload.
This is configured in the IPSec Security Association table, which defines the IP peers to
which IPSec security is applied.
IPSec specifications summary:
 Transport and Tunneling Mode
 Encapsulation Security Payload (ESP) only
 Encryption algorithms: AES, DES, and 3DES
 Hash types: SHA1 and MD5
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13.4.1 Enabling IPSec

To enable IKE and IPSec processing, you must enable the IPSec feature, as described
below.
 To enable IPSec:
Security > General Security Settings).
Figure 13-3: Enabling IPSec
2. Set the 'Enable IP Security' parameter to Enable.

3. Click Submit, and then reset the device with a flash burn.
13.4.2 Configuring IP Security Proposal Table

The IP Security Proposal Table page is used to configure Internet Key Exchange (IKE) with
up to four proposal settings. Each proposal defines an encryption algorithm, an
authentication algorithm, and a Diffie-Hellman group identifier. The same set of proposals
applies to both Main mode and Quick mode.
Note: You can also configure the IP Security Proposals table using the table ini file
parameter IPsecProposalTable (see 'Security Parameters' on page 504).
 To configure IP Security Proposals:

1. Open the IP Security Proposal Table page (Configuration tab > VoIP menu >
Security submenu > IPSec Proposal Table).
Figure 13-4: IP Security Proposals Table - Add Record Dialog Box
3. Configure the parameters, as required. For a description of the parameters, see the
table below.
4. Click Submit.

Table 13-3: IP Security Proposals Table Configuration Parameters
Parameter Name Description
Encryption Algorithm Defines the encryption (privacy) algorithm.

[IPsecProposalTable_EncryptionAlgorithm]  [0] NONE
 [1] DES CBC
 [2] 3DES CBC
 [3] AES (default)
Authentication Algorithm Defines the message authentication (integrity)
[IPsecProposalTable_AuthenticationAlgorithm] algorithm.
 [0] NONE
 [2] HMAC SHA1 96
 [4] HMAC MD5 96 (default)
Diffie Hellman Group Defines the length of the key created by the DH
[IPsecProposalTable_DHGroup] protocol for up to four proposals. For the ini file
parameter, X denotes the proposal number (0 to
3).
 [0] Group 1 (768 Bits) = DH-786-Bit
 [1] Group 2 (1024 Bits) (default) = DH-1024-
Bit
If no proposals are defined, the default settings (shown in the following table) are applied.
Table 13-4: Default IPSec/IKE Proposals
Proposal Encryption Authentication DH Group
Proposal 0 3DES SHA1 Group 2 (1024 bit)

Proposal 1 3DES MD5 Group 2 (1024 bit)
Proposal 2 3DES SHA1 Group 1 (786 bit)
Proposal 3 3DES MD5 Group 1 (786 bit)
13.4.3 Configuring IP Security Associations Table

The IP Security Associations Table page allows you to configure up to 20 peers (hosts or
networks) for IP security (IPSec)/IKE. Each of the entries in this table controls both Main
and Quick mode configuration for a single peer. Each row in the table refers to a different
IP destination. IPSec can be applied to all traffic to and from a specific IP address.
Alternatively, IPSec can be applied to a specific flow, specified by port (source or
destination) and protocol type.
The destination IP address (and optionally, destination port, source port and protocol type)
of each outgoing packet is compared to each entry in the table. If a match is found, the
device checks if an SA already exists for this entry. If no SA exists, the IKE protocol is
invoked and an IPSec SA is established and the packet is encrypted and transmitted. If a
match is not found, the packet is transmitted without encryption.
This table can also be used to enable Dead Peer Detection (RFC 3706), whereby the
device queries the liveliness of its IKE peer at regular intervals or on-demand. When two
peers communicate with IKE and IPSec, the situation may arise in which connectivity
between the two goes down unexpectedly. In such cases, there is often no way for IKE and
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IPSec to identify the loss of peer connectivity. As such, the Security Associations (SA)
remain active until their lifetimes naturally expire, resulting in a "black hole" situation where
both peers discard all incoming network traffic. This situation may be resolved by
performing periodic message exchanges between the peers. When no reply is received,
the sender assumes SA’s are no longer valid on the remote peer and attempts to
renegotiate.
Notes:
• Incoming packets whose parameters match one of the entries in the IP Security
Associations table but is received without encryption, is rejected.
• If you change the device's IP address on-the-fly, you must then reset the device
for IPSec to function properly.
• The proposal list must be contiguous.
• For security, once the IKE pre-shared key is configured, it is not displayed in any
of the device's management tools.
• You can also configure the IP Security Associations table using the table ini file
parameter IPsecSATable (see 'Security Parameters' on page 504).
 To configure the IPSec Association table:

1. Open the IP Security Associations Table page (Configuration tab > VoIP menu >
Security submenu > IPSec Association Table).
Figure 13-5: IP Security Associations Table Page - Add Record Dialog Box
3. Configure the parameters, as required. In the above figure, a single IPSec/IKE peer
(10.3.2.73) is configured. Pre-shared key authentication is selected with the pre-
shared key set to 123456789. In addition, a lifetime of 28800 seconds is set for IKE
and a lifetime of 3600 seconds is set for IPSec. For a description of the parameters,
see the table below.
4. Click Submit.

Table 13-5: IP Security Associations Table Configuration Parameters
Operational Mode Defines the IPSec mode of operation.

[IPsecSATable_IPsecMode]  [0] Transport (default)
 [1] Tunnel
Remote Endpoint Addr Defines the IP address or DNS host name of the peer.
[IPsecSATable_RemoteEndpointAd Note: This parameter is applicable only if the Operational
dressOrName] Mode is set to Transport.
Authentication Method Defines the method for peer authentication during IKE main
[IPsecSATable_AuthenticationMeth mode.
od]  [0] Pre-shared Key (default)
 [1] RSA Signature = in X.509 certificate
Note: For RSA-based authentication, both peers must be
provisioned with certificates signed by a common CA. For
more information on certificates, see 'Replacing the Device's
Certificate' on page 111.
Shared Key Defines the pre-shared key (in textual format). Both peers
[IPsecSATable_SharedKey] must use the same pre-shared key for the authentication
process to succeed.
Notes:
 This parameter is applicable only if the Authentication
Method parameter is set to pre-shared key.
 The pre-shared key forms the basis of IPSec security and
therefore, it should be handled with care (the same as
sensitive passwords). It is not recommended to use the
same pre-shared key for several connections.
 Since the ini file is plain text, loading it to the device over
a secure network connection is recommended. Use a
secure transport such as HTTPS, or a direct crossed-
cable connection from a management PC.
 After it is configured, the value of the pre-shared key
cannot be retrieved.
Source Port Defines the source port to which this configuration applies.
[IPsecSATable_SourcePort] The default is 0 (i.e., any port).
Destination Port Defines the destination port to which this configuration
[IPsecSATable_DestPort] applies.
The default is 0 (i.e., any port).
Protocol Defines the protocol type to which this configuration applies.
[IPsecSATable_Protocol] Standard IP protocol numbers, as defined by the Internet
Assigned Numbers Authority (IANA) should be used, for
example:
 0 = Any protocol (default)
 17 = UDP
 6 = TCP
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IKE SA Lifetime Defines the duration (in seconds) for which the negotiated
[IPsecSATable_Phase1SaLifetimeIn IKE SA (Main mode) is valid. After this time expires, the SA
Sec] is re-negotiated.
The default is 0 (i.e., unlimited).
Note: Main mode negotiation is a processor-intensive
operation; for best performance, do not set this parameter to
less than 28,800 (i.e., eight hours).
IPSec SA Lifetime (sec) Defines the duration (in seconds) for which the negotiated
[IPsecSATable_Phase2SaLifetimeIn IPSec SA (Quick mode) is valid. After this time expires, the
Sec] SA is re-negotiated.
The default is 0 (i.e., unlimited).
Note: For best performance, a value of 3,600 (i.e., one hour)
or more is recommended.
IPSec SA Lifetime (Kbs) Defines the maximum volume of traffic (in kilobytes) for
[IPsecSATable_Phase2SaLifetimeIn which the negotiated IPSec SA (Quick mode) is valid. After
KB] this specified volume is reached, the SA is re-negotiated.
The default is 0 (i.e., the value is ignored).
Dead Peer Detection Mode Defines dead peer detection (DPD), according to RFC 3706.
[IPsecSATable_DPDmode]  [0] DPD Disabled (default)
 [1] DPD Periodic = DPD is enabled with message
exchanges at regular intervals
 [2] DPD on demand = DPD is enabled with on-demand
checks - message exchanges as needed (i.e., before
sending data to the peer). If the liveliness of the peer is
questionable, the device sends a DPD message to query
the status of the peer. If the device has no traffic to send,
it never sends a DPD message.
Remote Tunnel Addr Defines the IP address of the peer router.
[IPsecSATable_RemoteTunnelAddr Note: This parameter is applicable only if the Operational
ess] Mode is set to Tunnel.
Remote Subnet Addr Defines the IP address of the remote subnet. Together with
[IPsecSATable_RemoteSubnetIPAd the Prefix Length parameter (below), this parameter defines
dress] the network with which the IPSec tunnel allows
communication.
Note: This parameter is applicable only if the Operational
Mode is set to Tunnel.
Remote Prefix Length Defines the prefix length of the Remote Subnet IP Address
[IPsecSATable_RemoteSubnetPrefi parameter (in bits). The prefix length defines the subnet
xLength] class of the remote network. A prefix length of 16
corresponds to a Class B subnet (255.255.0.0); a prefix
length of 24 corresponds to a Class C subnet
(255.255.255.0).
Note: This parameter is applicable only if the Operational
Mode is set to Tunnel.
Interface Name Assigns a network interface to this IPSec rule. The network
[IPsecSATable_InterfaceName] interfaces are defined in the Multiple Interface table
('Interface Name' column) in 'Configuring IP Network
Interfaces' on page 124

User's Manual 14. Media
14 Media
This section describes the media-related configuration.
14.1 Configuring Voice Settings

The Voice Settings page configures various voice parameters such as voice volume and
DTMF transport type. For a detailed description of these parameters, see 'Configuration
 To configure the voice parameters:

1. Open the Voice Settings page (Configuration tab > VoIP menu > Media submenu >
Voice Settings).
2. Configure the Voice parameters as required.

14.1.1 Configuring Voice Gain (Volume) Control

The device allows you to configure the level of the received (input gain) Tel-to-IP signal
and the level of the transmitted (output gain) IP-to-Tel signal. The gain can be set between
-32 and 31 decibels (dB).
The procedure below describes how to configure gain control using the Web interface:
 To configure gain control using the Web interface:

Voice Settings).
Figure 14-1: Voice Volume Parameters in Voice Settings Page
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2. Configure the following parameters:

• 'Voice Volume' (VoiceVolume) - Defines the voice gain control (in decibels) for IP-
to-Tel
• 'Input Gain' (InputGain) - Defines the PCM input gain control (in decibels) for Tel-
to-IP
14.1.2 Echo Cancellation

The device supports adaptive linear (line) echo cancellation according to G.168-2002.
Echo cancellation is a mechanism that removes echo from the voice channel. Echoes are
reflections of the transmitted signal.
In this line echo, echoes are generated when two-wire telephone circuits (carrying both
transmitted and received signals on the same wire pair) are converted to a four-wire circuit.
Echoes are reflections of the transmitted signal, which result from impedance mismatch in
the hybrid (bi-directional 2-wire to 4-wire converting device).
An estimated echo signal is built by feeding the decoder output signal to an RLS-like
adaptive filter, which adapts itself to the characteristics of the echo path. The ‘estimated
echo signal’ (the output of this filter) is then subtracted from the input signal (which is the
sum of the desired input signal and the undesired echo) to provide a clean signal. To
suppress the remaining residual echo, a Non Linear Processor (NLP) is used, as well as a
double-talk (two people speak at the same time) detector that prevents false adaptation
during near-end speech.

The procedure below describes how to configure echo cancellation using the Web
interface:
 To configure echo cancellation using the Web interface:

Voice Settings).
Figure 14-2: Enabling Echo Cancellation in Voice Settings Page
2. Set the 'Echo Canceller' field (EnableEchoCanceller) to Enable.
Note: The following additional echo cancellation parameters are configurable only
through the ini file:
• ECHybridLoss - defines the four-wire to two-wire worst-case Hybrid loss
• ECNLPMode - defines the echo cancellation Non-Linear Processing (NLP) mode
• EchoCancellerAggressiveNLP - enables Aggressive NLP at the first 0.5 second of
the call
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14.2 Fax and Modem Capabilities

This section describes the device's fax and modem capabilities and corresponding
configuration. The fax and modem configuration is done in the Fax/Modem/CID Settings
page.
Notes:
• Unless otherwise specified, the configuration parameters mentioned in this section
are available on this page.
• Some SIP parameters override these fax and modem parameters. For example,
the IsFaxUsed parameter and V.152 parameters in Section 'V.152 Support' on
page 177).
• For a detailed description of the parameters appearing on this page, see
'Configuration Parameters Reference' on page 475.
 To access the fax and modem parameters:

1. Open the Fax/Modem/CID Settings page (Configuration tab > VoIP menu > Media
submenu > Fax/Modem/CID Settings).
Figure 14-3: Fax/Modem/CID Settings Page
2. Configure the parameters, as required.


14.2.1 Fax/Modem Transport Modes

The device supports the following transport modes for fax per modem type
(V.22/V.23/Bell/V.32/V.34):
 T.38 fax relay (see 'T.38 Fax Relay Mode' on page 169)
 G.711 Transport: switching to G.711 when fax/modem is detected (see 'G.711 Fax /
Modem Transport Mode' on page 170)
 Fax fallback to G.711 if T.38 is not supported (see 'Fax Fallback' on page 171)
 Fax and modem bypass: a proprietary method that uses a high bit rate coder (see
'Fax/Modem Bypass Mode' on page 171)
 NSE Cisco’s Pass-through bypass mode for fax and modem (see 'Fax / Modem NSE
Mode' on page 173)
 Transparent with events: passing the fax / modem signal in the current voice coder
with adaptations (see 'Fax / Modem Transparent with Events Mode' on page 174)
 Transparent: passing the fax / modem signal in the current voice coder (see 'Fax /
Modem Transparent Mode' on page 174)
 RFC 2833 ANS Report upon Fax/Modem Detection (see 'RFC 2833 ANS Report upon
Fax/Modem Detection' on page 175)
‘Adaptations’ refer to automatic reconfiguration of certain DSP features for handling
fax/modem streams differently than voice.
14.2.1.1 T.38 Fax Relay Mode

In Fax Relay mode, fax signals are transferred using the T.38 protocol. T.38 is an ITU
standard for sending fax across IP networks in real-time mode. The device currently
supports only the T.38 UDP syntax.
T.38 can be configured in the following ways:
 Switching to T.38 mode using SIP Re-INVITE messages (see 'Switching to T.38 Mode
using SIP Re-INVITE' on page 169)
 Automatically switching to T.38 mode without using SIP Re-INVITE messages (see
'Automatically Switching to T.38 Mode without SIP Re-INVITE' on page 170)
When fax transmission ends, the reverse switching from fax relay to voice is automatically
performed at both the local and remote endpoints.
You can change the fax rate declared in the SDP, using the 'Fax Relay Max Rate'
parameter (FaxRelayMaxRate). This parameter does not affect the actual transmission
rate. You can also enable or disable Error Correction Mode (ECM) fax mode using the 'Fax
Relay ECM Enable' parameter (FaxRelayECMEnable).
When using T.38 mode, you can define a redundancy feature to improve fax transmission
over congested IP networks. This feature is activated using the 'Fax Relay Redundancy
Depth' parameter (FaxRelayRedundancyDepth) and the 'Fax Relay Enhanced
Redundancy Depth' parameter (FaxRelayEnhancedRedundancyDepth). Although this is a
proprietary redundancy scheme, it should not create problems when working with other
T.38 decoders.
14.2.1.1.1 Switching to T.38 Mode using SIP Re-INVITE

In the Switching to T.38 Mode using SIP Re-INVITE mode, upon detection of a fax signal
the terminating device negotiates T.38 capabilities using a Re-INVITE message. If the far-
end device doesn't support T.38, the fax fails. In this mode, the 'Fax Transport Mode'
parameter (FaxTransportMode) is ignored.
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 To configure T.38 mode using SIP Re-INVITE messages:

1. In the SIP General Parameters page (Configuration tab > VoIP menu > SIP
Definitions > General Parameters), set the 'Fax Signaling Method' parameter to T.38
Relay (IsFaxUsed = 1).
2. In the Fax/Modem/CID Settings page, configure the following optional parameters:
• 'Fax Relay Redundancy Depth' (FaxRelayRedundancyDepth)
• 'Fax Relay Enhanced Redundancy Depth'
(FaxRelayEnhancedRedundancyDepth)
• 'Fax Relay ECM Enable' (FaxRelayECMEnable)
• 'Fax Relay Max Rate' (FaxRelayMaxRate)
Note: The terminating gateway sends T.38 packets immediately after the T.38
capabilities are negotiated in SIP. However, the originating device by default, sends
T.38 (assuming the T.38 capabilities are negotiated in SIP) only after it receives T.38
packets from the remote device. This default behavior cannot be used when the
originating device is located behind a firewall that blocks incoming T.38 packets on
ports that have not yet received T.38 packets from the internal network. To resolve
this problem, the device should be configured to send CNG packets in T.38 upon
CNG signal detection (CNGDetectorMode = 1).
14.2.1.1.2 Automatically Switching to T.38 Mode without SIP Re-INVITE

In the Automatically Switching to T.38 Mode without SIP Re-INVITE mode, when a fax
signal is detected, the channel automatically switches from the current voice coder to
answer tone mode and then to T.38-compliant fax relay mode.
 To configure automatic T.38 mode:

Definitions > General Parameters), set the 'Fax Signaling Method' parameter to No
Fax (IsFaxUsed = 0).
2. In the Fax/Modem/CID Settings page, set the 'Fax Transport Mode' parameter to
RelayEnable (FaxTransportMode = 1).
3. Configure the following optional parameters:
• 'Fax Relay Redundancy Depth' (FaxRelayRedundancyDepth)
• 'Fax Relay Enhanced Redundancy Depth'
(FaxRelayEnhancedRedundancyDepth)
• 'Fax Relay ECM Enable' (FaxRelayECMEnable)
• 'Fax Relay Max Rate' (FaxRelayMaxRate)
14.2.1.2 G.711 Fax / Modem Transport Mode

In this mode, when the terminating device detects fax or modem signals (CED or AnsAM),
it sends a Re-INVITE message to the originating device, requesting it to re-open the
channel in G.711 VBD with the following adaptations:
 Echo Canceller = off
 Silence Compression = off
 Echo Canceller Non-Linear Processor Mode = off
 Dynamic Jitter Buffer Minimum Delay = 40
 Dynamic Jitter Buffer Optimization Factor = 13

After a few seconds upon detection of fax V.21 preamble or super G3 fax signals, the
device sends a second Re-INVITE enabling the echo canceller (the echo canceller is
disabled only on modem transmission).
A ‘gpmd’ attribute is added to the SDP according to the following format:
 For G.711 A-law:
a=gpmd:0 vbd=yes;ecan=on (or off for modems)
 For G.711 µ-law:
a=gpmd:8 vbd=yes;ecan=on (or off for modems)
The following parameters are ignored and automatically set to Events Only:
 'Fax Transport Mode' (FaxTransportMode)
 'Vxx ModemTransportType' (VxxModemTransportType)
 To configure fax / modem transparent mode:

 In the SIP General Parameters page (Configuration tab > VoIP menu > SIP
Definitions > General Parameters), set the 'Fax Signaling Method' parameter to
G.711 Transport (IsFaxUsed = 2).
14.2.1.3 Fax Fallback

In this mode, when the terminating device detects a fax signal, it sends a Re-INVITE
message to the originating device with T.38. If the remote device doesn’t support T.38
(replies with SIP response 415 "Media Not Supported"), the device sends a new Re-
INVITE with G.711 VBD with the following adaptations:
 Echo Canceller = on
 Silence Compression = off
 Dynamic Jitter Buffer Minimum Delay = 40
 Dynamic Jitter Buffer Optimization Factor = 13
When the device initiates a fax session using G.711, a ‘gpmd’ attribute is added to the SDP
according to the following format:
 For G.711A-law:
a=gpmd:0 vbd=yes;ecan=on
 For G.711 µ-law:
a=gpmd:8 vbd=yes;ecan=on
In this mode, the 'Fax Transport Mode' (FaxTransportMode) parameter is ignored and
automatically set to Disable (transparent mode).
 To configure fax fallback mode:

 In the SIP General Parameters page (Configuration tab > VoIP menu > SIP
Definitions > General Parameters), set the 'Fax Signaling Method' parameter to Fax
Fallback (IsFaxUsed = 3).
14.2.1.4 Fax/Modem Bypass Mode

In this proprietary mode, when fax or modem signals are detected, the channel
automatically switches from the current voice coder to a high bit-rate coder, according to
the 'Fax/Modem Bypass Coder Type' parameter (FaxModemBypassCoderType). The
channel is also automatically reconfigured with the following fax / modem adaptations:
 Disables silence suppression
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 Enables echo cancellation for fax

 Disables echo cancellation for modem
 Performs certain jitter buffering optimizations
The network packets generated and received during the bypass period are regular voice
RTP packets (per the selected bypass coder), but with a different RTP payload type
according to the following parameters:
 'Fax Bypass Payload Type' (FaxBypassPayloadType)
 ModemBypassPayloadType (ini file)
During the bypass period, the coder uses the packing factor, configured by the 'Fax/Modem
Bypass Packing Factor' parameter (FaxModemBypassM). The packing factor determines
the number of coder payloads (each the size of
FaxModemBypassBasicRTPPacketInterval) that are used to generate a single fax/modem
bypass packet. When fax/modem transmission ends, the reverse switching, from bypass
coder to regular voice coder is performed.
 To configure fax / modem bypass mode:

2. In the Fax/Modem/CID Settings page, do the following:
a. Set the 'Fax Transport Mode' parameter to Bypass (FaxTransportMode = 2).
b. Set the 'V.21 Modem Transport Type' parameter to Enable Bypass
(V21ModemTransportType = 2).
c. Set the 'V.22 Modem Transport Type' parameter to Enable Bypass
d. Set the 'V.23 Modem Transport Type' parameter to Enable Bypass
e. Set the 'V.32 Modem Transport Type' parameter to Enable Bypass
f. Set the 'V.34 Modem Transport Type' parameter to Enable Bypass
3. Set the ini file parameter, BellModemTransportType to 2 (Bypass).
• 'Fax/Modem Bypass Coder Type' (FaxModemBypassCoderType).
• 'Fax Bypass Payload Type' (FaxBypassPayloadType) - in the RTP/RTCP
Settings page (Configuration tab > VoIP menu > Media > RTP/RTCP Settings).
• ModemBypassPayloadType (ini file).
• FaxModemBypassBasicRTPPacketInterval (ini file).
• FaxModemBypasDJBufMinDelay (ini file).
Note: When the device is configured for modem bypass and T.38 fax, V.21 low-
speed modems are not supported and fail as a result.

Tip: When the remote (non-AudioCodes) gateway uses the G.711 coder for voice
and doesn’t change the coder payload type for fax or modem transmission, it is
recommended to use the Bypass mode with the following configuration:
• EnableFaxModemInbandNetworkDetection = 1.
• 'Fax/Modem Bypass Coder Type' = same coder used for voice.
• 'Fax/Modem Bypass Packing Factor'(FaxModemBypassM) = same interval as
voice.
• ModemBypassPayloadType = 8 if voice coder is A-Law or 0 if voice coder is Mu-
Law.
14.2.1.5 Fax / Modem NSE Mode

In this mode, fax and modem signals are transferred using Cisco-compatible Pass-through
bypass mode. Upon detection of fax or modem answering tone signal, the terminating
device sends three to six special NSE RTP packets (configured by the NSEpayloadType
parameter; usually to 100). These packets signal the remote device to switch to G.711
coder, according to the 'Fax/Modem Bypass Packing Factor' parameter. After a few NSE
packets are exchanged between the devices, both devices start using G.711 packets with
standard payload type (8 for G.711 A-Law and 0 for G.711 Mu-Law). In this mode, no Re-
INVITE messages are sent. The voice channel is optimized for fax/modem transmission
(same as for usual bypass mode).
The parameters defining payload type for AudioCodes proprietary Bypass mode -- 'Fax
Bypass Payload Type' (RTP/RTCP Settings page) and ModemBypassPayloadType (ini file)
-- are not used with NSE Bypass.
When configured for NSE mode, the device includes in its SDP the following line:
a=rtpmap:100 X-NSE/8000
Where 100 is the NSE payload type.
The Cisco gateway must include the following definition:
modem passthrough nse payload-type 100 codec g711alaw
 To configure NSE mode:

f. Set the 'V.34 Modem Transport Type' parameter to Enable Bypass
3. Set the ini file parameter, BellModemTransportType to 2 (Bypass).
4. Set the ini file parameter, NSEMode parameter to 1 (enables NSE).
5. Set the ini file parameter, NSEPayloadType parameter to 100.
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14.2.1.6 Fax / Modem Transparent with Events Mode

In this mode, fax and modem signals are transferred using the current voice coder with the
following automatic adaptations:
 Echo Canceller = on (or off for modems)
 Jitter buffering optimizations
 To configure fax / modem transparent with events mode:

a. Set the 'Fax Transport Mode' parameter to Events Only (FaxTransportMode =
3).
b. Set the 'V.21 Modem Transport Type' parameter to Events Only
c. Set the 'V.22 Modem Transport Type' parameter to Events Only
d. Set the 'V.23 Modem Transport Type' parameter to Events Only
e. Set the 'V.32 Modem Transport Type' parameter to Events Only
f. Set the 'V.34 Modem Transport Type' parameter to Events Only
3. Set the ini file parameter, BellModemTransportType to 3 (transparent with events).
14.2.1.7 Fax / Modem Transparent Mode

In this mode, fax and modem signals are transferred using the current voice coder without
notifications to the user and without automatic adaptations. It's possible to use Profiles (see
'Coders and Profiles' on page 219) to apply certain adaptations to the channel used for fax
/ modem. For example, to use the coder G.711, to set the jitter buffer optimization factor to
13, and to enable echo cancellation for fax and disable it for modem.
 To configure fax / modem transparent mode:

a. Set the 'Fax Transport Mode' parameter to Disable (FaxTransportMode = 0).
b. Set the 'V.21 Modem Transport Type' parameter to Disable
c. Set the 'V.22 Modem Transport Type' parameter to Disable
d. Set the 'V.23 Modem Transport Type' parameter to Disable
e. Set the 'V.32 Modem Transport Type' parameter to Disable
f. Set the 'V.34 Modem Transport Type' parameter to Disable
3. Set the ini file parameter, BellModemTransportType to 0 (transparent mode).


a. Coders table - (Configuration tab > VoIP menu > Coders and Profiles >
Coders).
b. 'Dynamic Jitter Buffer Optimization Factor' (DJBufOptFactor) - RTP/RTCP
Settings page (Configuration tab > VoIP menu > Media > RTP/RTCP Settings).
c. 'Echo Canceller' (EnableEchoCanceller) - Voice Settings page.
Note: This mode can be used for fax, but is not recommended for modem
transmission. Instead, use the Bypass (see 'Fax/Modem Bypass Mode' on page 171)
or Transparent with Events modes (see 'Fax / Modem Transparent with Events Mode'
on page 174) for modem.
14.2.1.8 RFC 2833 ANS Report upon Fax/Modem Detection

The device (terminator gateway) sends RFC 2833 ANS/ANSam events upon detection of
fax and/or modem answer tones (i.e., CED tone). This causes the originator to switch to
fax/modem. This parameter is applicable only when the fax or modem transport type is set
to bypass, Transparent-with-Events, V.152 VBD, or G.711 transport. When the device is
located on the originator side, it ignores these RFC 2833 events
 To configure RFC 2833 ANS Report upon fax/modem detection:

Fax or Fax Fallback (IsFaxUsed = 0 or 3).
b. Set the 'V.xx Modem Transport Type' parameters to Enable Bypass
(VxxModemTransportType = 2).
3. Set the ini file parameter, FaxModemNTEMode to 1 (enables this feature).
14.2.2 V.34 Fax Support

V.34 fax machines can transmit data over IP to the remote side using various methods.
The device supports the following modes for transporting V.34 fax data over IP:
 Bypass mechanism for V.34 fax transmission (see 'Bypass Mechanism for V.34 Fax
Transmission' on page 176)
 T38 Version 0 relay mode, i.e., fallback to T.38 (see 'Relay Mode for T.30 and V.34
Faxes' on page 176)
Note: The CNG detector is disabled in all the subsequent examples. To disable the
CNG detector, set the 'CNG Detector Mode' parameter (CNGDetectorMode) to
Disable.
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14.2.2.1 Bypass Mechanism for V.34 Fax Transmission

In this proprietary scenario, the device uses bypass (or NSE) mode to transmit V.34 faxes,
enabling the full utilization of its speed.
 To use bypass mode for T.30 and V.34 faxes:

 To use bypass mode for V.34 faxes, and T.38 for T.30 faxes:
a. Set the 'Fax Transport Mode' parameter to Relay (FaxTransportMode = 1).
14.2.2.2 Relay Mode for T.30 and V.34 Faxes

In this scenario, V.34 fax machines are forced to use their backward compatibility with T.30
faxes and operate in the slower T.30 mode.
 To use T.38 mode for V.34 and T.30 faxes:

a. Set the 'Fax Transport Mode' parameter to Relay (FaxTransportMode = 1).
b. Set the 'V.22 Modem Transport Type' parameter to Disable
c. Set the 'V.23 Modem Transport Type' parameter to Disable
d. Set the 'V.32 Modem Transport Type' parameter to Disable
e. Set the 'V.34 Modem Transport Type' parameter to Disable

14.2.3 V.152 Support

The device supports the ITU-T recommendation V.152 (Procedures for Supporting Voice-
Band Data over IP Networks). Voice-band data (VBD) is the transport of modem, facsimile,
and text telephony signals over a voice channel of a packet network with a codec
appropriate for such signals.
For V.152 capability, the device supports T.38 as well as VBD codecs (i.e., G.711 A-law
and G.711 μ-law). The selection of capabilities is performed using the coders table (see
'Configuring Coders' on page 219).
When in VBD mode for V.152 implementation, support is negotiated between the device
and the remote endpoint at the establishment of the call. During this time, initial exchange
of call capabilities is exchanged in the outgoing SDP. These capabilities include whether
VBD is supported and associated RTP payload types ('gpmd' SDP attribute), supported
codecs, and packetization periods for all codec payload types ('ptime' SDP attribute). After
this initial negotiation, no Re-INVITE messages are necessary as both endpoints are
synchronized in terms of the other side's capabilities. If negotiation fails (i.e., no match was
achieved for any of the transport capabilities), fallback to existing logic occurs (according to
the parameter IsFaxUsed).
Below is an example of media descriptions of an SDP indicating support for V.152. In the
example, V.152 implementation is supported (using the dynamic payload type 96 and
G.711 u-law as the VBD codec) as well as the voice codecs G.711 μ-law and G.729.
v=0
o=- 0 0 IN IPV4 <IPAdressA>
s=-
t=0 0
p=+1
c=IN IP4 <IPAddressA
m=audio <udpPort A> RTP/AVP 18 0
a=ptime:10
a=rtpmap:96 PCMU/8000
a=gpmd: 96 vbd=yes
Instead of using VBD transport mode, the V.152 implementation can use alternative relay
fax transport methods (e.g., fax relay over IP using T.38). The preferred V.152 transport
method is indicated by the SDP ‘pmft’ attribute. Omission of this attribute in the SDP
content means that VBD mode is the preferred transport mechanism for voice-band data.
To configure T.38 mode, use the CodersGroup parameter.
Note: You can also configure the device to handle G.711 coders received in INVITE
SDP offers as VBD coders, using the HandleG711asVBD parameter. For example, if
the device is configured with G.729 and G.711 VBD coders and it receives an INVITE
with an SDP offer containing G.729 and “regular” G.711 coders, it sends an SDP
answer containing G.729 and G.711 VBD coders, allowing subsequent bypass
(passthrough) sessions if fax / modem signals are detected during the call.
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14.2.4 Fax Transmission behind NAT

The device supports transmission from fax machines (connected to the device) located
inside (behind) a Network Address Translation (NAT). Generally, the firewall blocks T.38
(and other) packets received from the WAN, unless the device behind the NAT sends at
least one IP packet from the LAN to the WAN through the firewall. If the firewall blocks T.38
packets sent from the termination IP fax, the fax fails.
To overcome this, the device sends No-Op (“no-signal”) packets to open a pinhole in the
NAT for the answering fax machine. The originating fax does not wait for an answer, but
immediately starts sending T.38 packets to the terminating fax machine upon receipt of a
re-INVITE with T.38 only in the SDP, or T.38 and audio media in the SDP. This feature is
configured using the T38FaxSessionImmediateStart parameter. The No-Op packets are
enabled using the NoOpEnable and NoOpInterval parameters.

14.3 Configuring RTP/RTCP Settings

This section describes configuration relating to Real-Time Transport Protocol (RTP) and
RTP Control Protocol (RTCP).
14.3.1 Configuring the Dynamic Jitter Buffer

Voice frames are transmitted at a fixed rate. If the frames arrive at the other end at the
same rate, voice quality is perceived as good. However, some frames may arrive slightly
faster or slower than the other frames. This is called jitter (delay variation) and degrades
the perceived voice quality. To minimize this problem, the device uses a jitter buffer. The
jitter buffer collects voice packets, stores them and sends them to the voice processor in
evenly spaced intervals.
The device uses a dynamic jitter buffer that can be configured with the following:
 Minimum delay: Defines the starting jitter capacity of the buffer. For example, at 0
msec, there is no buffering at the start. At the default level of 10 msec, the device
always buffers incoming packets by at least 10 msec worth of voice frames.
 Optimization Factor: Defines how the jitter buffer tracks to changing network
conditions. When set at its maximum value of 12, the dynamic buffer aggressively
tracks changes in delay (based on packet loss statistics) to increase the size of the
buffer and doesn’t decay back down. This results in the best packet error
performance, but at the cost of extra delay. At the minimum value of 0, the buffer
tracks delays only to compensate for clock drift and quickly decays back to the
minimum level. This optimizes the delay performance but at the expense of a higher
error rate.
The default settings of 10 msec Minimum delay and 10 Optimization Factor should provide
a good compromise between delay and error rate. The jitter buffer ‘holds’ incoming packets
for 10 msec before making them available for decoding into voice. The coder polls frames
from the buffer at regular intervals in order to produce continuous speech. As long as
delays in the network do not change (jitter) by more than 10 msec from one packet to the
next, there is always a sample in the buffer for the coder to use. If there is more than 10
msec of delay at any time during the call, the packet arrives too late. The coder tries to
access a frame and is not able to find one. The coder must produce a voice sample even if
a frame is not available. It therefore compensates for the missing packet by adding a Bad-
Frame-Interpolation (BFI) packet. This loss is then flagged as the buffer being too small.
The dynamic algorithm then causes the size of the buffer to increase for the next voice
session. The size of the buffer may decrease again if the device notices that the buffer is
not filling up as much as expected. At no time does the buffer decrease to less than the
minimum size configured by the Minimum delay parameter.
In certain scenarios, the Optimization Factor is set to 13: One of the purposes of the
Jitter Buffer mechanism is to compensate for clock drift. If the two sides of the VoIP call are
not synchronized to the same clock source, one RTP source generates packets at a lower
rate, causing under-runs at the remote Jitter Buffer. In normal operation (optimization factor
0 to 12), the Jitter Buffer mechanism detects and compensates for the clock drift by
occasionally dropping a voice packet or by adding a BFI packet.
Fax and modem devices are sensitive to small packet losses or to added BFI packets.
Therefore, to achieve better performance during modem and fax calls, the Optimization
Factor should be set to 13. In this special mode the clock drift correction is performed less
frequently - only when the Jitter Buffer is completely empty or completely full. When such
condition occurs, the correction is performed by dropping several voice packets
simultaneously or by adding several BFI packets simultaneously, so that the Jitter Buffer
returns to its normal condition.
The procedure below describes how to configure the jitter buffer using the Web interface.
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 To configure jitter buffer using the Web interface:

1. Open the RTP/RTCP Settings page (Configuration tab > VoIP menu > Media
submenu > RTP/RTCP Settings). The relevant parameters are listed under the
'General Settings' group, as shown below:
Figure 14-4: Jitter Buffer Parameters in the RTP/RTCP Settings Page
2. Set the 'Dynamic Jitter Buffer Minimum Delay' parameter (DJBufMinDelay) to the
minimum delay (in msec) for the Dynamic Jitter Buffer.
3. Set the 'Dynamic Jitter Buffer Optimization Factor' parameter (DJBufOptFactor) to the
Dynamic Jitter Buffer frame error/delay optimization factor.
14.3.2 Comfort Noise Generation

The device can generate artificial background noise, called comfort noise, in the voice
channel during periods of silence (i.e. when no call party is speaking). This is useful in that
it reassures the call parties that the call is still connected. The device detects silence using
its Voice Activity Detection (VAD) mechanism. When the Calling Tone (CNG) is enabled
and silence is detected, the device transmits Silence Identifier Descriptors (SIDs)
parameters to reproduce the local background noise at the remote (receiving) side.
The Comfort Noise Generation (CNG) support also depends on the silence suppression
(SCE) setting for the coder used in the voice channel. For more information, see the
description of the CNG-related parameters.
The procedure below describes how to configure CNG using the Web interface.
 To configure CNG using the Web interface:

Figure 14-5: Comfort Noise Parameter in RTP/RTCP Settings Page
2. Set the 'Comfort Noise Generation Negotiation' parameter (ComfortNoiseNegotiation)

to Enable.

14.3.3 Dual-Tone Multi-Frequency Signaling

This section describes the configuration of Dual-Tone Multi-Frequency (DTMF) signaling.
14.3.3.1 Configuring DTMF Transport Types

The device supports various methods for transporting DTMF digits over the IP network to
the remote endpoint. These methods and their configuration are configured in the DTMF &
Dialing page (Configuration tab > VoIP menu > GW and IP to IP > DTMF and
Supplementary > DTMF & Dialing):
 Using INFO message according to Nortel IETF draft: DTMF digits are sent to the
remote side in INFO messages. To enable this mode, define the following:
a. Set the 'Declare RFC 2833 in SDP' parameter to No (RxDTMFOption = 0).
b. Set the '1st Tx DTMF Option' parameter to INFO (Nortel) (TxDTMFOption = 1).
Note: In this mode, DTMF digits are removed from the audio stream (and the 'DTMF
Transport Type' parameter is automatically set to Mute DTMF).
 Using INFO message according to Cisco’s mode: DTMF digits are sent to the
b. Set the '1st Tx DTMF Option' parameter to INFO (Cisco) (TxDTMFOption = 3).
 Using NOTIFY messages according to IETF Internet-Draft draft-mahy-sipping-
signaled-digits-01: DTMF digits are sent to the remote side using NOTIFY
messages. To enable this mode, define the following:
b. Set the '1st Tx DTMF Option' parameter to NOTIFY (TxDTMFOption = 2).
 Using RFC 2833 relay with Payload type negotiation: DTMF digits are sent to the
remote side as part of the RTP stream according to RFC 2833. To enable this mode,
define the following:
a. Set the 'Declare RFC 2833 in SDP' parameter to Yes (RxDTMFOption = 3).
b. Set the '1st Tx DTMF Option' parameter to RFC 2833 (TxDTMFOption = 4).
Note: To set the RFC 2833 payload type with a value other than its default, use the
RFC2833PayloadType parameter. The device negotiates the RFC 2833 payload type
using local and remote SDP and sends packets using the payload type from the
received SDP. The device expects to receive RFC 2833 packets with the same
payload type as configured by this parameter. If the remote side doesn’t include
‘telephony-event’ in its SDP, the device sends DTMF digits in transparent mode (as
part of the voice stream).
 Sending DTMF digits (in RTP packets) as part of the audio stream (DTMF Relay
is disabled): This method is typically used with G.711 coders. With other low-bit rate
(LBR) coders, the quality of the DTMF digits is reduced. To enable this mode, define
the following:
b. Set the '1st Tx DTMF Option' parameter to Not Supported (TxDTMFOption = 0).
c. Set the ini file parameter, DTMFTransportType to 2 (i.e., transparent).
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 Using INFO message according to Korea mode: DTMF digits are sent to the
b. Set the '1st Tx DTMF Option' parameter to INFO (Cisco) (TxDTMFOption = 3).
Notes:
• The device is always ready to receive DTMF packets over IP in all possible
transport modes: INFO messages, NOTIFY, and RFC 2833 (in proper payload
type) or as part of the audio stream.
• To exclude RFC 2833 Telephony event parameter from the device's SDP, set the
'Declare RFC 2833 in SDP' parameter to No.
The following parameters affect the way the device handles the DTMF digits:
 TxDTMFOption, RxDTMFOption, RFC2833TxPayloadType, and
RFC2833RxPayloadType
 MGCPDTMFDetectionPoint, DTMFVolume, DTMFTransportType, DTMFDigitLength,
and DTMFInterDigitInterval
14.3.3.2 Configuring RFC 2833 Payload

The procedure below describes how to configure the RFC 2833 payload using the Web
interface:
 To configure RFC 2833 payload using the Web interface:

Figure 14-6: RFC 2833 Payload Parameters in RTP/RTCP Settings Page
2. Configure the following parameters:

• 'RTP Redundancy Depth' (RTPRedundancyDepth) - enables the device to
generate RFC 2198 redundant packets.
• 'Enable RTP Redundancy Negotiation' (EnableRTPRedundancyNegotiation) -
enables the device to include the RTP redundancy dynamic payload type in the
SDP, according to RFC 2198.
• 'RFC 2833 TX Payload Type' (RFC2833TxPayloadType) - defines the Tx RFC
2833 DTMF relay dynamic payload type.
• 'RFC 2833 RX Payload Type' (RFC2833RxPayloadType) - defines the Rx RFC
2833 DTMF relay dynamic payload type.

• 'RFC 2198 Payload Type' (RFC2198PayloadType) - defines the RTP redundancy

packet payload type according to RFC 2198.
14.3.4 Configuring RTP Base UDP Port

You can configure the range of UDP ports for RTP, RTCP, and T.38. The UDP port range
can be configured using media realms in the Media Realm table, allowing you to assign
different port ranges (media realms) to different interfaces. However, if you do not use
media realms, you can configure the lower boundary of the UDP port used for RTP, RTCP
(RTP port + 1) and T.38 (RTP port + 2), using the 'RTP Base UDP Port' (BaseUDPport)
parameter. For example, if the BaseUDPPort is set to 6000, then one channel may use the
ports RTP 6000, RTCP 6001, and T.38 6002, while another channel may use RTP 6010,
RTCP 6011, and T.38 6012.
The range of possible UDP ports is 6,000 to 64,000 (default base UDP port is 6000). The
port range is calculated using the BaseUDPport parameter as follows: BaseUDPPort to
(BaseUDPPort + <channels -1> * 10)
The default local UDP ports for audio and fax media streams is calculated using the
following formula: BaseUDPPort + (Channel ID * 10) + Port Offset
Where the port offsets are as follows:
 Audio RTP: 0
 Audio RTCP: 1
 Fax T.38: 2
For example, the local T.38 UDP port for channel 30 is calculated as follows: 6000 +
(30*10) + 2 = 6302
The maximum (when all channels are required) UDP port range is calculated as follows:
 MP-112/MP-114: BaseUDPport to (BaseUDPport + 3*10) - for example, if the
BaseUDPPort is set to 6,000, then the UDP port range is 6,000 to 6,030
 MP-118: BaseUDPport to (BaseUDPport + 7*10) - for example, if the BaseUDPPort is
set to 6,000, then the UDP port range is 6,000 to 6,070
 MP-124: BaseUDPport to (BaseUDPport + 23*10) - for example, if the BaseUDPPort
is set to 6,000, then the UDP port range is 6,000 to 6,230
Notes:
• The device allocates the UDP ports randomly to the channels.
• To configure the device to use the same port for both RTP and T.38 packets, set
the T38UseRTPPort parameter to 1.
• If you are using Media Realms (see Configuring Media Realms on page 188), the
port range configured for the Media Realm must be within this range defined by
the BaseUDPPort parameter.
The procedure below describes how to configure the RTP base UDP port using the Web
interface.
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 To configure the RTP base UDP port:

submenu > RTP/RTCP Settings). The relevant parameter is listed under the 'General
Settings' group, as shown below:
Figure 14-7: RTP Based UDP Port in RTP/RTCP Settings Page
2. Set the 'RTP Base UDP Port' parameter to the required value.
3. Click Submit.
4. Reset the device for the settings to take effect.
14.4 Configuring Analog Settings

The Analog Settings page allows you to configure various analog parameters. For a
detailed description of the parameters appearing on this page, see 'Configuration
This page also selects the type (USA or Europe) of FXS and/or FXO coefficient
information. The FXS coefficient contains the analog telephony interface characteristics
such as DC and AC impedance, feeding current, and ringing voltage.
 To configure the analog parameters:

1. Open the Analog Settings page (Configuration tab > VoIP menu > Media submenu >
Analog Settings).
Figure 14-8: Analog Settings Page


14.5 Configuring DSP Templates

The DSP Template determines the coders that can be used by the device and various
other functionalities. For a list of DSP templates and the maximum number of channels
supported by each coder, see 'DSP Templates' on page 649. You can select a single DSP
Template.
Notes:
• If no entries are defined, the device uses the default DSP template (i.e., Template
0).
• A single DSP Template can also be configured using the ini file parameter,
DSPVersionTemplateNumber.
 To select a DSP Template(s):

1. Open the General Settings page (Configuration tab > VoIP menu > Media > General
Media Settings).
Figure 14-9: Defining Single DSP Template in General Settings Page
2. In the 'DSP Version Template Number' field, enter the required DSP Template
number.
3. Click Submit.
4. Reset the device with a flash burn for the settings to take effect (see 'Saving
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14.6 Configuring Media Security

The device supports Secured RTP (SRTP) according to RFC 3711. SRTP is used to
encrypt RTP and RTCP transport for protecting VoIP traffic. SRTP requires a key
exchange mechanism that is performed according to RFC 4568 – “Session Description
Protocol (SDP) Security Descriptions for Media Streams”. The key exchange is done by
adding a 'crypto' attribute to the SDP. This attribute is used (by both sides) to declare the
various supported cipher suites and to attach the encryption key. If negotiation of the
encryption data is successful, the call is established.
SRTP supports the following cipher suites (all other suites are ignored):
 AES_CM_128_HMAC_SHA1_32
 AES_CM_128_HMAC_SHA1_80
When the device is the offering side, it generates an MKI of a size configured by the
'Master Key Identifier (MKI) Size' parameter. The length of the MKI is limited to four bytes.
If the remote side sends a longer MKI, the key is ignored. The key lifetime field is not
supported. However, if it is included in the key it is ignored and the call does not fail.
The device supports the following session parameters (as defined in RFC 4568, SDP
Security Descriptions for Media Streams):
 UNENCRYPTED_SRTP
 UNENCRYPTED_SRTCP
 UNAUTHENTICATED_SRTP
Session parameters should be the same for the local and remote sides. When the device is
the offering side, the session parameters are configured by the following parameter -
'Authentication On Transmitted RTP Packets', 'Encryption On Transmitted RTP Packets,
and 'Encryption On Transmitted RTCP Packets'. When the device is the answering side,
the device adjusts these parameters according to the remote offering. Unsupported
session parameters are ignored, and do not cause a call failure.
Below is an example of crypto attributes usage:
a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:PsKoMpHlCg+b5X0YLuSvNrImEh/dAe
inline:IsPtLoGkBf9a+c6XVzRuMqHlDnEiAd
The device also supports symmetric MKI negotiation, whereby it can be configured to
forward the MKI size received in the SDP offer crypto line in the SDP answer crypto line.
To configure the device's mode of operation if negotiation of the cipher suite fails, use the
'Media Security Behavior' parameter. This parameter can be set to enforce SRTP, whereby
incoming calls that don’t include encryption information are rejected.
Notes:
• For a detailed description of the SRTP parameters, see SRTP Parameters on
page 507.
• When SRTP is used, the channel capacity may be reduced.

 To configure media security:

1. Open the Media Security page (Configuration tab > VoIP menu > Media submenu >
Media Security).

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14.7 Configuring Media Realms

The Media Realm Table page allows you to define a pool of up to 64 SIP media interfaces,
termed Media Realms. Media Realms allow you to divide a Media-type interface, which is
configured in the Multiple Interface table, into several realms, where each realm is
specified by a UDP port range. You can also define the maximum number of sessions per
Media Realm. Once configured, Media Realms can be assigned to IP Groups (see
'Configuring IP Groups' on page 205).
Once you have configured a Media Realm, you can configure it with the following:
 Quality of Experience parameters for reporting to AudioCodes SEM server used for
monitoring the quality of calls (see Configuring Quality of Experience Parameters per
Media Realm on page 190)
Notes:
• For this setting to take effect, a device reset is required.
• The Media Realm table can also be configured using the table ini file parameter,
CpMediaRealm.
 To define a Media Realm:

1. Open the Media Realm Table page (Configuration tab > VoIP menu > Media
submenu > Media Realm Configuration).
2. Click the Add button; the following appears:
Figure 14-10: Media Realm Page - Add Record Dialog Box
3. Configure the parameters as required. See the table below for a description of each
parameter
5. Reset the device to save the changes to flash memory (see 'Saving Configuration' on
page 366).
Table 14-1: Media Realm Table Parameter Descriptions
Index Defines the required table index number.

[CpMediaRealm_Index]

Media Realm Name Defines an arbitrary, identifiable name for the Media Realm.
[CpMediaRealm_MediaRealm The valid value is a string of up to 40 characters.
Name] Notes:
 This parameter is mandatory.
 The name assigned to the Media Realm must be unique.
 This Media Realm name is used in the IP Groups table.
IPv4 Interface Name Assigns an IPv4 interface to the Media Realm. This is name of the
[CpMediaRealm_IPv4IF] interface as configured for the Interface Name field in the Multiple
Interface table.
IPv6 Interface Name Assigns an IPv6 interface to the Media Realm. This is name of the
[CpMediaRealm_IPv6IF] interface as configured for the Interface Name field in the Multiple
Interface table.
Port Range Start Defines the starting port for the range of Media interface UDP
[CpMediaRealm_PortRangeSt ports.
art] Notes:
 You must either configure all media realms with port ranges or
all without; not some with and some without.
 The available UDP port range is calculated using the
BaseUDPport parameter:
 BaseUDPport to BaseUDPport + 4030*10
 Port ranges over 60,000 must not be used.
Number of Media Session Legs Defines the number of media sessions associated with the range of
[CpMediaRealm_MediaSessio ports. This is the number of media sessions available in the port
nLeg] range. For example, 100 ports correspond to 10 media sessions,
since ports are allocated in chunks of 10.
Port Range End Read-only field displaying the ending port for the range of Media
[CpMediaRealm_PortRangeE interface UDP ports. This field is calculated by adding the 'Media
nd] Session Leg' field (multiplied by the port chunk size) to the 'Port
Range Start' field. A value appears once a row has been
successfully added to the table.
Is Default Defines the Media Realm as the default Media Realm. This default
[CpMediaRealm_IsDefault] Media Realm is used when no Media Realm is configured for an IP
Group for a specific call.
 [0] No (default)
 [1] Yes
Notes:
 This parameter can be set to Yes for only one defined Media
Realm.
 If this parameter is not configured, then the first Media Realm in
the table is used as the default.
 If the table is not configured, then the default Media Realm
includes all the configured media interfaces.
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14.7.1 Configuring Quality of Experience per Media Realm

You can configure Quality of Experience (QoE) per Media Realm. This enables you to
monitor and analyze media and signaling traffic, allowing you to detect problems causing
service degradation. The device can save call information and statistics at call start, at call
end, or at specific changes in the call. The information is stored as call records on an
external server. The device connects, as a client, to the server using TLS over TCP.
You can specify the call parameters to monitor and configure their upper and lower
thresholds. If these thresholds are exceeded, the device can be configured to do the
following:
 Reports the change in the monitored parameter to the monitoring server (default).
 Sends RFC 2198 RTP redundancy packets on the call leg that crossed the threshold.
This enables the device to adapt to the changed network status. In this option, you
can also configure the redundancy depth. The channel configuration is unchanged if
the change requires channel reopening. Currently, this option is applicable only when
the monitored parameter is remote packet loss.
The device can be configured to monitor the following parameters on the local (i.e., at the
device) or remote side:
 Packet loss
 Mean Opinion Score (MOS)
 Jitter
 Packet delay
 Residual Echo Return Loss (RERL)
At any given time during a call, each of these parameters can be in one of the following
states according to its value in the last RTCP / RTCP XR packet:
 Gray - indicates that the value is unknown
 Green - indicates good call quality
 Yellow - indicates medium call quality
 Red - indicates poor call quality
The mapping between the values of the parameters and the color is according to the
configured threshold of these parameters, per Media Realm. The call itself also has a state
(color), which is the worst-state color of all the monitored parameters. Each time a color of
a parameter changes, the device sends a report to the external server. A report is also sent
at the end of each call.
Notes:
• The QoE feature is available only if the device is installed with the relevant
Software License Key.
• To configure the address of the AudioCodes Session Experience Manager (SEM)
server to where the device reports the QoE, see 'Configuring SEM Server for
Media Quality of Experience' on page 193.
• You can also configure QoE per Media Realm using the table ini file parameter
QOERules.
 To configure QoE per Media Realm:

1. Open the Media Realm Table page (Configuration tab > VoIP menu > Media
submenu > Media Realm Configuration).
2. Select the Media Realm for which you want to configure Quality of Experience, and
then click the Quality Of Experience link; the Quality Of Experience page appears.


Figure 14-11: Quality of Experience Page - Add Record Dialog Box
The figure above shows value thresholds for the MOS parameter, which are assigned
using pre-configured values of the Low Sensitivity profile. In this example setting, if the
MOS value changes by 0.1 (hysteresis) to 3.3 or 3.5, the device sends a report to the
SEM indicating this change. If the value changes to 3.3, it sends a yellow state (i.e.,
medium quality); if the value changes to 3.5, it sends a green state.
4. Configure the parameters as required. See the table below for a description of each
parameter.
Table 14-2: Quality of Experience Parameter Descriptions
Index Defines the table index entry. Up to four table row entries can be
[QOERules_RuleIndex] configured per Media Realm.
Monitored Parameter Defines the parameter to monitor and report.
[QOERules_MonitoredParam]  [0] MOS (default)
 [1] Delay
 [2] Packet Loss
 [3] Jitter
 [4] RERL
Direction Defines the monitoring direction.
[QOERules_Direction]  [0] Device Side (default)
 [1] Remote Side
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Profile Defines the pre-configured threshold profile to use.

[QOERules_Profile]  [0] No Profile = No profile is used and you need to define the
thresholds in the parameters described below.
 [1] Low Sensitivity = Automatically sets the thresholds to low
sensitivity values. Therefore, reporting is done only if changes in
parameters' values is significant.
 [2] Default Sensitivity = Automatically sets the thresholds to a
medium sensitivity.
 [3] High Sensitivity = Automatically sets the thresholds to high
sensitivity values. Therefore, reporting is done for small
fluctuations in parameters' values.
Green Yellow Threshold Defines the parameter threshold values between green (good
[QOERules_GreenYellowThre quality) and yellow (medium quality) states.
shold]
Green Yellow Hysteresis Defines the hysteresis (fluctuation) for the green-yellow threshold.
[QOERules_GreenYellowHyst When the threshold is exceeded by this hysteresis value, the
ersis] device sends a report to the SEM indicating this change.
Yellow Red Threshold Defines the parameter threshold values between yellow (medium
[QOERules_YellowRedThresh quality) and red (poor quality). When this threshold is exceeded,
old] the device sends a report to the SEM indicating this change.
Yellow Red Hysteresis Defines the hysteresis (fluctuation) for the yellow-red threshold.
[QOERules_YellowRedHyster When the threshold is exceeded by this hystersis value, the device
sis] sends a report to the SEM indicating this change.
Green Yellow Operation Defines the action that is done if the green-yellow threshold is
[QOERules_GreenYellowOper crossed.
ation]  [1] Notify = (Default) Device sends a report to the SEM server.
 [2] Change Redundancy Depth= RTP redundancy packets are
sent to the relevant call leg.
Note: This field is applicable only if the monitored parameter is
remote packet loss.
Green Yellow Operation Details Note: This field is currently not supported.
[QOERules_GreenYellowOper Defines the desired RTP redundancy depth. The actual
ationDetails] redundancy depth on the relevant call leg is the minimum between
the desired depth and the maximum supported depth on that call
leg.
Note: This field is applicable only if the 'Green Yellow Operation'
field is set to Change Redundancy Depth.
Yellow Red Operation Note: This field is currently not supported.
[QOERules_YellowRedOperat Defines the action that is done if the yellow-red threshold is
ion] crossed.
 [1] Notify = (Default) Device sends a report to the SEM server.
 [2] Change Redundancy Depth = RTP redundancy packets are
sent to the relevant call leg.
 Note: This field is applicable only if the monitored parameter is
remote packet loss.

Yellow Red Operation Details Note: This field is currently not supported.
[QOERules_YellowRedOperat Defines the desired RTP redundancy depth. The actual
ionDetails] redundancy depth on the relevant call leg is the minimum between
the desired depth and the maximum supported depth on that call
leg.
Note: This field is applicable only if the 'Yellow Red Operation' field
is set to Change Redundancy Depth.
14.8 Quality of Experience

This chapter describes how to configure the Quality of Experience feature.
14.8.1 Reporting Voice Quality of Experience to SEM

The device can be configured to report voice (media) Quality of Experience (QoE) to
AudioCodes' Session Experience Manager (SEM) server, a plug-in for AudioCodes EMS.
The reports include real-time metrics of the quality of the actual call experience, which are
then processed by the SEM.
SEM is a VoIP-quality monitoring and analysis tool. SEM provides comprehensive details
on voice traffic quality, allowing system administrators to quickly identify, fix and prevent
issues that could affect the voice calling experience in enterprise and service provider VoIP
networks. IT managers and administrators can employ SEM in their VoIP networks to
guarantee effective utilization, smooth performance, reliable QoS levels, and SLA
fulfillment.
Note: For information on the SEM server, refer to the SEM User's Manual.
14.8.1.1 Configuring the SEM Server

The device can report QoE voice metrics to AudioCodes SEM server.
You can also configure at what stage of the call the device must send the report to the
SEM server. The report can be sent during the call or only at the end of the call. Reporting
at the end of the call may be beneficial when network congestion occurs, as this reduces
bandwidth usage over time.
Notes:
• To support this feature, the device must be installed with the relevant Software
License Key.
• To configure the parameters to report and their thresholds per Media Realm, see
'Configuring Quality of Experience per Media Realm' on page 190.
• For information on the SEM server, refer to the EMS User's Manual.
For a detailed description of the SEM parameters, see ''SIP Media Realm Parameters'' on
page 518.
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 To configure the SEM server address and other related features:

1. Open the Media Quality of Experience page (Configuration tab > VoIP menu >
Media submenu > Media Quality of Experience.
Figure 14-12: Configuring Session Experience Manager
2. In the 'Server IP Address' field, enter the SEM server's IP address.

3. In the 'Port' field, enter the SEM server's port.
4. In the 'Interface Name' field, enter the device's IP network interface from which the
device sends the reports to the SEM server.
5. Click Submit.
14.8.1.2 Configuring Clock Synchronization between Device and SEM

To ensure accurate call quality statistics and analysis by the SEM server, you must
configure the device and the SEM server with the same clock source for clock
synchronization. In other words, you need to configure them with the same NTP server.
The NTP server can be one of the following:
 AudioCodes EMS server (also acting as an NTP server)
 Third-party, external NTP server
Once you have determined the NTP server, all the elements--device, SEM, and EMS--must
be configured with the same NTP server address.
To configure, the NTP server's address on the device, see ''Configuring Automatic Date
and Time using SNTP'' on page 119.
14.8.1.3 Enabling RTCP XR Reporting to SEM

In order for the device to be able to send voice metric reports to the SEM, you need to
enable the RTP Control Protocol Extended Reports (RTCP XR) VoIP management
protocol. RTCP XR defines a set of voice metrics that contain information for assessing
VoIP call quality and diagnosing problems. Enabling RTCP XR means that the device can
send RTCP XR messages, containing the call-quality metrics, to the SEM server.
For enabling RTCP XR reporting, see ''Configuring RTCP XR'' on page 423. To configure
what to report to the SEM, see ''Configuring Quality of Experience Profiles'' on page 190.

User's Manual 15. Services
15 Services
This section describes configuration for various supported services.
15.1 Least Cost Routing

This section provides a description of the device's least cost routing (LCR) feature and how
to configure it.
15.1.1 Overview
The LCR feature enables the device to choose the outbound IP destination routing rule
based on lowest call cost. This is useful in that it enables service providers to optimize
routing costs for customers. For example, you may wish to define different call costs for
local and international calls, or different call costs for weekends and weekdays (specifying
even the time of call). The device sends the calculated cost of the call to a Syslog server
(as Information messages), thereby enabling billing by third-party vendors.
LCR is implemented by defining Cost Groups and assigning them to routing rules in the
Outbound IP Routing table. The device searches this routing table for matching routing
rules, and then selects the rule with the lowest call cost. If two routing rules have identical
costs, then the rule appearing higher up in the table is used (i.e., first-matched rule). If a
selected route is unavailable, the device selects the next least-cost routing rule. However,
even if a matched rule is not assigned a Cost Group, the device can select it as the
preferred route over other matched rules with Cost Groups. This is determined according to
the settings of the Default Cost parameter in the Routing Rule Groups table.
The Cost Group defines a fixed connection cost (connection cost) and a charge per minute
(minute cost). Cost Groups can also be configured with time segments (time bands), which
define connection cost and minute cost based on specific days of the week and time of day
(e.g., from Saturday through Sunday, between 6:00 and 18:00). If multiple time bands are
configured per Cost Group and a call spans multiple time bands, the call cost is calculated
using only the time band in which the call was initially established.
In addition to Cost Groups, the device can calculate the call cost using an optional, user-
defined average call duration value. The logic in using this option is that a Cost Group may
be cheap if the call duration is short, but due to its high minute cost, may prove very
expensive if the duration is lengthy. Thus, together with Cost Groups, the device can use
this option to determine least cost routing. The device calculates the Cost Group call cost
as follows: Total Call Cost = Connection Cost + (Minute Cost * Average Call Duration).
The below table shows an example of call cost when taking into consideration call duration.
This example shows four defined Cost Groups and the total call cost if the average call
duration is 10 minutes:
Table 15-1: Call Cost Comparison between Cost Groups for different Call Durations
Total Call Cost per Duration

Connection
Cost Group Minute Cost
Cost
1 Minute 10 Minutes
A 1 6 7 61
B 0 10 10 100
C 0.3 8 8.3 80.3
D 6 1 7 16
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If four matching routing rules are located in the routing table and each one is assigned a
different Cost Group as listed in the table above, then the rule assigned Cost Group "D" is
selected. Note that for one minute, Cost Groups "A" and "D" are identical, but due to the
average call duration, Cost Group "D" is cheaper. Therefore, average call duration is an
important factor in determining the cheapest routing role.
Below are a few examples of how you can implement LCR:
 Example 1: This example uses two different Cost Groups for routing local calls and
international calls:
Two Cost Groups are configured as shown below:
Cost Group Connection Cost Minute Cost
1. "Local Calls" 2 1
2. "International Calls" 6 3
The Cost Groups are assigned to routing rules for local and international calls in the
Outbound IP Routing table:
Routing Index Dest Phone Prefix Destination IP Cost Group ID
1 2000 x.x.x.x 1 "Local Calls"
2 00 x.x.x.x 2 "International Calls"
 Example 2: This example shows how the device determines the cheapest routing rule
in the Outbound IP Routing table:
The Default Cost parameter (global) in the Routing Rule Groups table is set to Min,
meaning that if the device locates other matching LCR routing rules (with Cost Groups
assigned), the routing rule without a Cost Group is considered the lowest cost route.
• The following Cost Groups are configured:
Cost Group Connection Cost Minute Cost
1. "A" 2 1
2. "B" 6 3
• The Cost Groups are assigned to routing rules in the Outbound IP Routing table:
Routing Index Dest Phone Prefix Destination IP Cost Group ID
1 201 x.x.x.x "A'
2 201 x.x.x.x "B"
3 201 x.x.x.x 0
4 201 x.x.x.x "B"
The device calculates the optimal route in the following index order: 3, 1, 2, and then
4, due to the following logic:
• Index 1 - Cost Group "A" has the lowest connection cost and minute cost
• Index 2 - Cost Group "B" takes precedence over Index 4 entry based on the first-
matched method rule
• Index 3 - no Cost Group is assigned, but as the Default Cost parameter is set to
Min, it is selected as the cheapest route
• Index 4 - Cost Group "B" is only second-matched rule (Index 1 is the first)

 Example 3: This example shows how the cost of a call is calculated if the call spans
over multiple time bands:
Assume a Cost Group, "CG Local" is configured with two time bands, as shown below:
Connection
Cost Group Time Band Start Time End Time Minute Cost
Cost
TB1 16:00 17:00 2 1
CG Local
TB2 17:00 18:00 7 2
Assume that the call duration is 10 minutes, occurring between 16:55 and 17:05. In
other words, the first 5 minutes occurs in time band "TB1" and the next 5 minutes
occurs in "TB2", as shown below:
Figure 15-1: LCR using Multiple Time Bands (Example)
The device calculates the call using the time band in which the call was initially
established, regardless of whether the call spans over additional time bands:
Total call cost = "TB1" Connection Cost + ("TB1" Minute Cost x call duration) = 2 + 1
x 10 min = 12
15.1.2 Configuring LCR

The following main steps need to be done to configure LCR:
1. Enable the LCR feature and configure the average call duration and default call
connection cost - see 'Enabling LCR and Configuring Default LCR' on page 197.
2. Configure Cost Groups - see 'Configuring Cost Groups' on page 199.
3. Configure Time Bands for a Cost Group - see 'Configuring Time Bands for Cost
Groups' on page 200.
4. Assign Cost Groups to outbound IP routing rules - see 'Assigning Cost Groups to
Routing Rules' on page 201.
15.1.2.1 Enabling the LCR Feature

The procedure below describes how to enable the LCR feature. This also includes
configuring the average call duration and default call cost for routing rules that are not
assigned Cost Groups in the Outbound IP Routing table.
Note: The Routing Rule Groups table can also be configured using the table ini file
parameter, RoutingRuleGroups.
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 To enable LCR:
1. Open the Routing Rule Groups Table page (Configuration tab > VoIP menu >
Services submenu > Least Cost Routing > Routing Rule Groups Table).
Figure 15-2: Routing Rule Groups Table - Add Record
3. Configure the parameters as required. For a description of the parameters, see the
table below.
4. Click Submit; the entry is added to the Routing Rule Groups table.
Table 15-2: Routing Rule Groups Table Description
Index Defines the table index entry.

[RoutingRuleGroups_Index] Note: Only one index entry can be configured.
LCR Enable Enables the LCR feature:
[RoutingRuleGroups_LCREnab  [0] Disabled (default)
le]  [1] Enabled
LCR Call Length Defines the average call duration (in minutes) and is used to
[RoutingRuleGroups_LCRAver calculate the variable portion of the call cost. This is useful, for
ageCallLength] example, when the average call duration spans over multiple time
bands. The LCR is calculated as follows: cost = call connect cost
+ (minute cost * average call duration)
The valid value range is 0-65533. The default is 1.
For example, assume the following Cost Groups:
 "Weekend A": call connection cost is 1 and charge per minute
is 6. Therefore, a call of 1 minute cost 7 units.
 "Weekend_ B": call connection cost is 6 and charge per
minute is 1. Therefore, a call of 1 minute cost 7 units.
Therefore, for calls under one minute, "Weekend A" carries the
lower cost. However, if the average call duration is more than one
minute, then "Weekend B" carries the lower cost.

Default Cost Determines whether routing rules in the Outbound IP Routing

[RoutingRuleGroups_LCRDefa table without an assigned Cost Group are considered a higher
ultCost] cost or lower cost route compared to other matched routing rules
that are assigned Cost Groups.
 [0] Lowest Cost = If the device locates other matching LCR
routing rules, this routing rule is considered the lowest cost
route and therefore, it is selected as the route to use (default.)
 [1] Highest Cost = If the device locates other matching LCR
routing rules, this routing rule is considered as the highest cost
route and therefore, is not used or used only if the other
cheaper routes are unavailable.
Note: If more than one valid routing rule without a defined Cost
Group exists, the device selects the first-matched rule.
15.1.2.2 Configuring Cost Groups

The procedure below describes how to configure Cost Groups. Cost Groups are defined
with a fixed call connection cost and a call rate (charge per minute). Once configured, you
can configure Time Bands for each Cost Group. Up to 10 Cost Groups can be configured.
Note: The Cost Group table can also be configured using the table ini file parameter,
CostGroupTable.
 To configure Cost Groups:

1. Open the Cost Group Table page (Configuration tab > VoIP menu > Services
submenu > Least Cost Routing > Cost Group Table).
table below.
4. Click Submit; the entry is added to the Cost Group table.
Table 15-3: Cost Group Table Description

[CostGroupTable_Index]
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Cost Group Name Defines an arbitrary name for the Cost Group.
[CostGroupTable_CostGroupNam The valid value is a string of up to 30 characters.
e]
Note: Each Cost Group must have a unique name.
Default Connect Cost Defines the call connection cost (added as a fixed charge to
[CostGroupTable_DefaultConnecti the call) for a call outside the time bands.
onCost] The valid value range is 0-65533. The default is 0.
Note: When calculating the cost of a call, if the current time of
the call is not within a time band configured for the Cost
Group, then this default connection cost is used.
Default Time Cost Defines the call charge per minute for a call outside the time
[CostGroupTable_DefaultMinuteC bands.
ost] The valid value range is 0-65533. The default is 0.
Note: When calculating the cost of a call, if the current time of
the call is not within a time band configured for the Cost
Group, then this default charge per minute is used.
15.1.2.3 Configuring Time Bands for Cost Groups

The procedure below describes how to configure Time Bands for a Cost Group. The time
band defines the day and time range for which the time band is applicable (e.g., from
Saturday 05:00 to Sunday 24:00) as well as the fixed call connection charge and call rate
per minute for this interval. Up to 70 time bands can be configured, and up to 21 time
bands can be assigned to each Cost Group.
Notes:
• You cannot define overlapping time bands.
• The Time Band table can also be configured using the table ini file parameter,
CostGroupTimebands.
 To configure Time Bands for a Cost Group:

1. Open the Cost Group Table page (Configuration tab > VoIP menu > Services
submenu > Least Cost Routing > Cost Group Table).
2. Select a Cost Group for which you want to assign Time Bands, and then click the
Time Band link located below the table; the Time Band table for the selected Cost
Group appears.

table below.
5. Click Submit; the entry is added to the Time Band table for the relevant Cost Group.
Table 15-4: Time Band Table Description

[CostGroupTimebands_TimebandI
ndex]
Start Time Defines the day and time of day from when this time band is
[CostGroupTimebands_StartTime] applicable. The format is DDD:hh:mm (e.g., SUN:06:00),
where:
 DDD is the day in upper case (i.e., SUN, MON, TUE, WED,
THU, FRI, or SAT)
 hh and mm denote the time of day, where hh is the hour
(00-23) and mm the minutes (00-59)
End Time Defines the day and time of day until when this time band is
[CostGroupTimebands_EndTime] applicable. For a description of the valid values, see the
parameter above.
Connection Cost Defines the call connection cost during this time band. This is
[CostGroupTimebands_Connectio added as a fixed charge to the call.
nCost] The valid value range is 0-65533. The default is 0.
Note: The entered value must be a whole number (i.e., not a
decimal).
Minute Cost Defines the call cost per minute charge during this timeband.
[CostGroupTimebands_MinuteCo The valid value range is 0-65533. The default is 0.
st]
Note: The entered value must be a whole number (i.e., not a
decimal).
15.1.2.4 Assigning Cost Groups to Routing Rules

Once you have configured your Cost Groups, you need to assign them to routing rules in
the Outbound IP Routing table - see Configuring Tel to IP Routing on page 256.
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User's Manual 16. Enabling Applications
16 Enabling Applications
In addition to the Gateway application (i.e., IP-to-Tel and Tel-to-IP calling), the device
supports the following main application:
 Stand-Alone Survivability (SAS) application
The procedure below describes how to enable these applications. Once an application is
enabled, the Web GUI provides menus and parameter fields relevant to the application.
Notes:
• For configuring the SAS application, see 'Stand-Alone Survivability (SAS)
Application' on page 327.
• For enabling an application, a device reset is required.
 To enable an application:
1. Open the Applications Enabling page (Configuration tab > VoIP menu >
Applications Enabling submenu > Applications Enabling).
2. From the relevant application drop-down list, select Enable.

3. Save (burn) the changes to the device's flash memory with a device reset (see 'Saving
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User's Manual 17. Control Network
17 Control Network
This section describes configuration of the network at the SIP control level.
17.1 Configuring IP Groups

The IP Group Table page allows you to create up to nine logical IP entities called IP
Groups. An IP Group is an entity with a set of definitions such as a Proxy Set ID (see
'Configuring Proxy Sets Table' on page 208), which represents the IP address of the IP
Group.
IP Groups are used for the following:
 SIP dialog registration and authentication (digest user/password) of a specific IP
Group (Served IP Group, e.g., corporate IP-PBX) with another IP Group (Serving IP
Group, e.g., ITSP). This is configured in the Account table (see Configuring Account
Table on page 213).
 Call routing rules:
• Outgoing IP calls (Tel-to-IP): The IP Group identifies the source of the call and is
used as the destination of the outgoing IP call (defined in the Tel to IP Routing).
For Tel-to-IP calls, the IP Group (Serving IP Group) can be used as the IP
destination to where all SIP dialogs that are initiated from a Hunt Group are sent
(defined in Configuring Hunt Group Settings on page 237).
• Incoming IP calls (IP-to-Tel): The IP Group identifies the source of the IP call.
• Number Manipulation rules to IP: The IP Group is used to associate the rule with
specific calls identified by IP Group.
Notes:
• IP Group ID 0 cannot be used. This IP Group is set to default values and is used
by the device when IP Groups are not implemented.
• When operating with multiple IP Groups, the default Proxy server must not be
used (i.e., the parameter IsProxyUsed must be set to 0).
• You can also configure the IP Groups table using the table ini file parameter,
IPGroup (see 'Configuration Parameters Reference' on page 475).
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 To configure IP Groups:
1. Open the IP Group Table page (Configuration tab > VoIP menu > Control Network
submenu > IP Group Table).
2. Click the Add button: the following dialog box appears:
3. Configure the IP Group parameters according to the table below.

4. Click Submit.
Table 17-1: IP Group Parameters
Common Parameters
Description Defines a brief description for the IP Group.
[IPGroup_Description] The valid value is a string of up to 29 characters. The default is
an empty field.
Proxy Set ID Assigns a Proxy Set ID to the IP Group. All INVITE messages
[IPGroup_ProxySetId] destined to this IP Group are sent to the IP address configured
for the Proxy Set.
Notes:
 Proxy Set ID 0 must not be used; this is the device's default
Proxy.
 To configure Proxy Sets, see 'Configuring Proxy Sets Table'
on page 208.
SIP Group Name Defines the SIP Request-URI host name used in INVITE and
[IPGroup_SIPGroupName] REGISTER messages sent to this IP Group, or the host name in
the From header of INVITE messages received from this IP
Group.
The valid value is a string of up to 49 characters. The default is
an empty field.
Note: If this parameter is not configured, the value of the global
parameter, ProxyName is used instead (see 'Configuring Proxy
and Registration Parameters' on page 216).

Contact User Defines the user part of the From, To, and Contact headers of
[IPGroup_ContactUser] SIP REGISTER messages, and the user part of the Contact
header of INVITE messages received from this IP Group and
forwarded by the device to another IP Group.
Note: This parameter is overridden by the ‘Contact User’
parameter in the ‘Account’ table (see 'Configuring Account
Table' on page 213).
Local Host Name Defines the host name (string) that the device uses in the SIP
[IPGroup_ContactName] message's Via and Contact headers. This is typically used to
define an FQDN as the host name. The device uses this string
for Via and Contact headers in outgoing INVITE messages to a
specific IP Group, and the Contact header in SIP 18x and 200
OK responses for incoming INVITE messages from a specific IP
Group. The Inbound IP Routing table can be used to identify the
source IP Group from where the INVITE message was received.
If this parameter is not configured (default), these headers are
populated with the device's dotted-decimal IP address of the
network interface on which the message is sent.
Note: To ensure proper device handling, this parameter should
be a valid FQDN.
Media Realm Name Assigns a Media Realm to the IP Group. The string value must
[IPGroup_MediaRealm] be identical (including case-sensitive) to the Media Realm name
defined in the Media Realm table.
Notes:
 For this parameter to take effect, a device reset is required.
 If the Media Realm is later deleted from the Media Realm
table, then this value becomes invalid.
 For configuring Media Realms, see Configuring Media
Realms on page 188.
IP Profile ID Assigns an IP Profile to the IP Group.
[IPGroup_ProfileId] The default is 0.
Note: To configure IP Profiles, see 'Configuring IP Profiles' on
page 225.
Gateway Parameters
Always Use Route Table Defines the Request-URI host name in outgoing INVITE
[IPGroup_AlwaysUseRouteTable] messages.
 [0] No (default).
 [1] Yes = The device uses the IP address (or domain name)
defined in the Tel to IP Routing (see Configuring the Tel to IP
Routing on page 256) as the Request-URI host name in
outgoing INVITE messages, instead of the value configured
in the 'SIP Group Name' field.
SIP Re-Routing Mode Defines the routing mode after a call redirection (i.e., a 3xx SIP
[IPGroup_SIPReRoutingMode] response is received) or transfer (i.e., a SIP REFER request is
received).
 [-1] Not Configured (Default)
 [0] Standard = INVITE messages that are generated as a
result of Transfer or Redirect are sent directly to the URI,
according to the Refer-To header in the REFER message or
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Contact header in the 3xx response.
 [1] Proxy = Sends a new INVITE to the Proxy. This is
applicable only if a Proxy server is used and the parameter
AlwaysSendtoProxy is set to 0.
 [2] Routing Table = Uses the Routing table to locate the
destination and then sends a new INVITE to this destination.
Notes:
 When this parameter is set to [1] and the INVITE sent to the
Proxy fails, the device re-routes the call according to the
Standard mode [0].
 When this parameter is set to [2] and the INVITE fails, the
device re-routes the call according to the Standard mode [0].
If DNS resolution fails, the device attempts to route the call to
the Proxy. If routing to the Proxy also fails, the Redirect /
Transfer request is rejected.
 When this parameter is set to [2], the XferPrefix parameter
can be used to define different routing rules for redirected
calls.
 This parameter is ignored if the parameter
AlwaysSendToProxy is set to 1.
17.2 Configuring Proxy Sets Table

The Proxy Sets Table page allows you to define Proxy Sets. A Proxy Set is a group of
Proxy servers defined by IP address or fully qualified domain name (FQDN). You can
define up to 10 Proxy Sets, each with up to five Proxy server addresses. For each Proxy
server address you can define the transport type (i.e., UDP, TCP, or TLS). The total
number of IP addresses that can be resolved from a DNS query is 15. In addition, Proxy
load balancing and redundancy mechanisms can be applied per Proxy Set if it contains
more than one Proxy address.
Proxy Sets can later be assigned to Server-type IP Groups (see 'Configuring IP Groups' on
page 205). When the device sends an INVITE message to an IP Group, it is sent to the IP
address or domain name defined for the Proxy Set that is associated with the IP Group. In
other words, the Proxy Set represents the destination of the call.
Notes:
• Proxy Sets can be assigned only to Server-type IP Groups.
• The Proxy Set table can also be configured using two complementary tables:
- Proxy Set ID with IP addresses: Table ini file parameter, ProxyIP.
- Attributes for the Proxy Set: Table ini file parameter, ProxySet.

 To configure Proxy Sets:

1. Open the Proxy Sets Table page (Configuration tab > VoIP menu > Control
Network submenu > Proxy Sets Table).
Figure 17-1: Proxy Sets Table Page
2. From the 'Proxy Set ID' drop-down list, select an ID for the desired group.
3. Configure the Proxy parameters, as required. For a description of the parameters, see
the table below.
4. Click Submit.
Table 17-2: Proxy Sets Table Parameters
Web: Proxy Set ID Defines the Proxy Set identification number.

EMS: Index The valid value is 0 to 9. Proxy Set ID 0 is used as the default Proxy Set.
[ProxySet_Index]
Note: Although not recommended, you can use both default Proxy Set
(ID 0) and IP Groups for call routing. For example, in the Hunt Group
Settings page (see Configuring Hunt Group Settings on page 237) you
can configure a Serving IP Group to where you want to route specific
Hunt Group endpoints, and all other device endpoints then use the default
Proxy Set. You can also use IP Groups in the Tel to IP Routing (see
Configuring the Tel to IP Routing on page 256) to configure the default
Proxy Set if the parameter PreferRouteTable is set to 1.
To summarize, if the default Proxy Set is used, the INVITE message is
sent according to the following preferences:
 To the Hunt Group's Serving IP Group ID, as defined in the Hunt
Group Settings table.
 According to the Tel to IP Routing if the parameter PreferRouteTable
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is set to 1.
 To the default Proxy.
Typically, when IP Groups are used, there is no need to use the default
Proxy and all routing and registration rules can be configured using IP
Groups and the Account tables (see 'Configuring Account Table' on page
213).
Proxy Address Defines the address (and optionally, port number) of the Proxy server. Up
[ProxyIp_IpAddress] to five addresses can be configured per Proxy Set.
The address can be defined as an IP address in dotted-decimal notation
(e.g., 201.10.8.1) or as an FQDN. You can also specify the selected port
in the format, <IP address>:<port>.
If you enable Proxy Redundancy (by setting the parameter
EnableProxyKeepAlive to 1 or 2), the device can operate with multiple
Proxy servers. If there is no response from the first (primary) Proxy
defined in the list, the device attempts to communicate with the other
(redundant) Proxies in the list. When a redundant Proxy is located, the
device either continues operating with it until the next failure occurs or
reverts to the primary Proxy (refer to the parameter
ProxyRedundancyMode). If none of the Proxy servers respond, the
device goes over the list again.
The device also provides real-time switching (Hot-Swap mode) between
the primary and redundant proxies (refer to the parameter
IsProxyHotSwap). If the first Proxy doesn't respond to the INVITE
message, the same INVITE message is immediately sent to the next
Proxy in the list. The same logic applies to REGISTER messages (if
RegistrarIP is not defined).
Notes:
 If EnableProxyKeepAlive is set to 1 or 2, the device monitors the
connection with the Proxies by using keep-alive messages (OPTIONS
or REGISTER).
 To use Proxy Redundancy, you must specify one or more redundant
Proxies.
 When a port number is specified (e.g., domain.com:5080), DNS
NAPTR/SRV queries aren't performed, even if ProxyDNSQueryType is
set to 1 or 2.
Transport Type Defines the transport type of the proxy server.
[ProxyIp_TransportTyp  [0] UDP
e]  [1] TCP
 [2] TLS
 [-1] = Undefined
Note: If no transport type is selected, the value of the global parameter
SIPTransportType is used.
Web/EMS: Enable Proxy Enables the Keep-Alive mechanism with the Proxy server(s).
Keep Alive  [0] Disable (default).
[ProxySet_EnableProxy  [1] Using Options = Enables Keep-Alive with Proxy using SIP
KeepAlive] OPTIONS messages.
 [2] Using Register = Enables Keep-Alive with Proxy using SIP
REGISTER messages.
If set to 'Using Options', the SIP OPTIONS message is sent every user-
defined interval (configured by the parameter ProxyKeepAliveTime). If set
to 'Using Register', the SIP REGISTER message is sent every user-
defined interval (configured by the RegistrationTime parameter). Any

response from the Proxy, either success (200 OK) or failure (4xx
response) is considered as if the Proxy is communicating correctly.
Notes:
 This parameter must be set to 'Using Options' when Proxy redundancy
is used.
 When this parameter is set to 'Using Register', the homing redundancy
mode is disabled.
 When the active proxy doesn't respond to INVITE messages sent by
the device, the proxy is tagged as 'offline'. The behavior is similar to a
Keep-Alive (OPTIONS or REGISTER) failure.
 If this parameter is enabled and the proxy uses the TCP/TLS transport
type, you can enable CRLF Keep-Alive mechanism, using the
UsePingPongKeepAlive parameter.
Web: Proxy Keep Alive Defines the Proxy keep-alive time interval (in seconds) between Keep-
Time Alive messages.
EMS: Keep Alive Time The valid range is 5 to 2,000,000. The default is 60.
[ProxySet_ProxyKeepAl
Note: This parameter is applicable only if the parameter
iveTime]
EnableProxyKeepAlive is set to 1 (OPTIONS). When the parameter
EnableProxyKeepAlive is set to 2 (REGISTER), the time interval between
Keep-Alive messages is determined by the RegistrationTime parameter.
Web: Proxy Load Enables the Proxy Load Balancing mechanism per Proxy Set ID.
Balancing Method  [0] Disable = Load Balancing is disabled (default)
EMS: Load Balancing  [1] Round Robin
Method
 [2] Random Weights
[ProxySet_ProxyLoadB
alancingMethod] When the Round Robin algorithm is used, a list of all possible Proxy IP
addresses is compiled. This list includes all IP addresses per Proxy Set,
after necessary DNS resolutions (including NAPTR and SRV, if
configured). After this list is compiled, the Proxy Keep-Alive mechanism
(according to parameters EnableProxyKeepAlive and
ProxyKeepAliveTime) tags each entry as 'offline' or 'online'. Load
balancing is only performed on Proxy servers that are tagged as 'online'.
All outgoing messages are equally distributed across the list of IP
addresses. REGISTER messages are also distributed unless a
RegistrarIP is configured.
The IP addresses list is refreshed according to ProxyIPListRefreshTime.
If a change in the order of the entries in the list occurs, all load statistics
are erased and balancing starts over again.
When the Random Weights algorithm is used, the outgoing requests are
not distributed equally among the Proxies. The weights are received from
the DNS server by using SRV records. The device sends the requests in
such a fashion that each Proxy receives a percentage of the requests
according to its' assigned weight. A single FQDN should be configured as
a Proxy IP address. The Random Weights Load Balancing is not used in
the following scenarios:
 The Proxy Set includes more than one Proxy IP address.
 The only Proxy defined is an IP address and not an FQDN.
 SRV is not enabled (DNSQueryType).
 The SRV response includes several records with a different Priority
value.
Web/EMS: Is Proxy Hot- Enables the Proxy Hot-Swap redundancy mode.
Swap
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[ProxySet_IsProxyHotS  [0] No (default)
wap]  [1] Yes
If Proxy Hot-Swap is enabled, the SIP INVITE/REGISTER message is
initially sent to the first Proxy/Registrar server. If there is no response
from the first Proxy/Registrar server after a specific number of
retransmissions (configured by the parameter HotSwapRtx), the message
is resent to the next redundant Proxy/Registrar server.
Web/EMS: Proxy Determines whether the device switches back to the primary Proxy after
Redundancy Mode using a redundant Proxy.
[ProxySet_ProxyRedun  [-1] Not Configured = (Default) The global parameter,
dancyMode] ProxyRedundancyMode applies.
 [0] Parking = The device continues operating with a redundant (now
active) Proxy until the next failure, after which it operates with the next
redundant Proxy.
 [1] Homing = The device always attempts to operate with the primary
Proxy server (i.e., switches back to the primary Proxy whenever it's
available).
Notes:
 To use the Proxy Redundancy mechanism, you need to enable the
keep-alive with Proxy option, by setting the parameter
EnableProxyKeepAlive to 1 or 2.
 If this parameter is configured, then the global parameter is ignored.
Main Proxy Success Defines the number of consecutive, successful keep-alive (using
Detection Retries OPTIONS method) responses from the primary proxy that are required
[ProxySet_HomingSucc before the device switches to the proxy after it was offline. This is used
essDetectionRetries] when the Proxy Set is configured for homing (i.e., 'Proxy Redundancy
Mode' parameter set to Homing).
The valid value range is 1 to 300 (default 1).
Note: The parameter is applicable only if 'Proxy Redundancy Mode' is
configured to Homing and 'Enable Proxy Keep Alive' is configured to
Using Options.

User's Manual 18. SIP Definitions
18 SIP Definitions
This section describes configuration of SIP parameters.
18.1 Configuring SIP Parameters

Many of the stand-alone SIP parameters associated with various features can be
configured in the following pages:
 SIP General Parameters page: Provides SIP parameters for configuring general SIP
features. To access this page, use the following path: Configuration tab > VoIP menu
> SIP Definitions > General Parameters.
 SIP Advanced Parameters page: Provides SIP parameters for configuring advanced
SIP features. To access this page, use the following path: Configuration tab > VoIP
menu > SIP Definitions > Advanced Parameters.
For a description of these parameters, refer to the section corresponding to the feature or
see 'Configuration Parameters Reference' on page 475.
18.2 Configuring Account Table

The Account Table page lets you define up to 24 Accounts per ("served") Hunt Group.
Accounts are used to register and/or digest authenticate a Hunt Group, using a username
and password, to a destination ("serving") IP Group. For example, the device can use the
Account table to register a PBX, which is connected to the device, to an ITSP. The device
sends the registration requests to the Proxy Set ID (see 'Configuring Proxy Sets Table' on
page 208) that is associated with the serving IP Group.
A Hunt Group or served IP Group can register to more than one serving IP Group (e.g.,
multiple ITSPs). This is done by configuring multiple entries in the Account table for the
same Hunt Group or served IP Group, but with different serving IP Groups, user
name/password, host name, and contact user values.
When using the Account table to register a Hunt Group, if all channels belonging to the
Hunt Group are down, the device un-registers the channels. If any channel belonging to the
Hunt Group is returned to service, the device registers them again. This ensures, for
example, that the Proxy does not send INVITEs to trunks that are out of service.
Notes:
• For viewing Account registration status, see Viewing Endpoint Registration Status
on page 419.
• The Account table can also be configured using the table ini file parameter,
Account.
 To configure Accounts:
1. Open the Account Table page (Configuration tab > VoIP menu > SIP Definitions
submenu > Account Table).
2. In the 'Add' field, enter the desired table row index, and then click Add. A new row
appears.
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3. Configure the Account parameters according to the table below.

4. Click the Apply button to save your changes.
6. To perform registration, click the Register button; to unregister, click Unregister. The
registration method for each Trunk Group is according to the setting of the
'Registration Mode' parameter in the Trunk Group Settings page.
Table 18-1: Account Table Parameters Description
Served Trunk Group Defines the Hunt Group ID that you want to register and/or
CLI: served-trunk-group authenticate to a destination IP Group (i.e., Serving IP Group).
[Account_ServedTrunkGroup]  For Tel-to-IP calls, the Served Hunt Group is the source Hunt
Group from where the call originated.
 For IP-to-Tel calls, the Served Hunt Group is the Hunt Group ID
to which the call is sent.
Serving IP Group Defines the destination IP Group ID to where the SIP REGISTER
[Account_ServingIPGroup] requests, if enabled, are sent and authentication is done. The actual
destination to where the REGISTER requests are sent is the IP
address configured for the Proxy Set ID that is associated with the IP
Group.
Registration occurs only if:
 The 'Registration Mode' parameter is set to 'Per Account' in the
Hunt Group Settings table (see Configuring Hunt Group Settings
on page 237).
 The 'Register' parameter in this Account table is set to Yes.
In addition, for a SIP call that is identified by both the Served Hunt
Group and Serving IP Group, the username and password for digest
authentication defined in this table is used.
For Tel-to-IP calls, the Serving IP Group is the destination IP Group
defined in the Hunt Group Settings table or Tel to IP Routing (see
Configuring the Tel to IP Routing on page 256). For IP-to-Tel calls,
the Serving IP Group is the 'Source IP Group ID' defined in the IP to
Hunt Group Routing Table (see Configuring the IP to Hunt Group
Routing Table on page 263).
Note: If no match is found in this table for incoming or outgoing calls,
the username and password defined in the Authentication table (see
Configuring Authentication on page 304) or by the global
parameters, UserName and Password (in the Proxy & Registration
page) are used.
Username Defines the digest MD5 Authentication user name.
[Account_Username] The valid value is a string of up to 50 characters.
Password Defines the digest MD5 Authentication password.
[Account_Password] The valid value is a string of up to 50 characters.
Note: After you click the Apply button, this password is displayed as
an asterisk (*).

Host Name Defines the Address of Record (AOR) host name. It appears in
[Account_HostName] REGISTER From/To headers as ContactUser@HostName. For
successful registrations, this host name is also included in the
INVITE request's From header URI.
This parameter can be up to 49 characters.
Note: If this parameter is not configured or if registration fails, the
'SIP Group Name' parameter configured in the IP Group table is
used instead.
Register Enables registration.
[Account_Register]  [0] No (Default)
 [1] Yes
When enabled, the device sends REGISTER requests to the Serving
IP Group. The host name (i.e., host name in SIP From/To headers)
and Contact User (user in From/To and Contact headers) are taken
from this table upon successful registration. See the example below:
REGISTER sip:xyz SIP/2.0
Via: SIP/2.0/UDP
10.33.37.78;branch=z9hG4bKac1397582418
From:
<sip:ContactUser@HostName>;tag=1c1397576231
To: <sip: ContactUser@HostName >
CSeq: 1 REGISTER
Contact:
<sip:[email protected]>;expires=3600
Expires: 3600
User-Agent: Sip-Gateway/v.6.00A.008.002
Content-Length: 0
Notes:
 To activate registration, you also need to set the parameter
'Registration Mode' to 'Per Account' in the Hunt Group Settings
table for the specific Hunt Group.
 The Hunt Group account registration is not affected by the
parameter IsRegisterNeeded.
Contact User Defines the AOR user name. This appears in REGISTER From/To
[Account_ContactUser] headers as ContactUser@HostName, and in INVITE/200 OK
Contact headers as ContactUser@<device's IP address>.
Notes:
 If this parameter is not configured, the 'Contact User' parameter
in the IP Group table is used instead.
 If registration fails, then the user part in the INVITE Contact
header contains the source party number.
Application Type Defines the application type:
[Account_ApplicationType]  [0] GW/IP2IP = (Default) Gateway application.
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18.3 Configuring Proxy and Registration Parameters

The Proxy & Registration page allows you to configure the Proxy server and registration
parameters. For a description of the parameters appearing on this page, see 'Configuration
Note: To view the registration status of endpoints with a SIP Registrar/Proxy server,
see Viewing Endpoint Registration Status on page 419.
 To configure the Proxy and registration parameters:

1. Open the Proxy & Registration page (Configuration tab > VoIP menu > SIP
Definitions submenu > Proxy & Registration).

 To register or un-register the device to a Proxy/Registrar:

 Click the Register button to register.
 Click Un-Register button to un-register.
Instead of registering the entire device, you can register specific entities as listed below by
using the Register button located on the page in which these entities are configured:

 FXS/FXO endpoints - Endpoint Phone Number Table page (see Configuring Endpoint
Phone Numbers on page 235)
 Accounts - Account table (see 'Configuring Account Table' on page 213)
Click the Proxy Set Table button to Open the Proxy Sets Table page to configure
groups of proxy addresses. Alternatively, you can open this page from the Proxy Sets
Table page item (see 'Configuring Proxy Sets Table' on page 208 for a description of this
page).
18.3.1 SIP Message Authentication Example

The device supports basic and digest (MD5) authentication types, according to SIP RFC
3261 standard. A proxy server might require authentication before forwarding an INVITE
message. A Registrar/Proxy server may also require authentication for client registration. A
proxy replies to an unauthenticated INVITE with a 407 Proxy Authorization Required
response, containing a Proxy-Authenticate header with the form of the challenge. After
sending an ACK for the 407, the user agent can then re-send the INVITE with a Proxy-
Authorization header containing the credentials.
User agents, Redirect or Registrar servers typically use the SIP 401 Unauthorized
response to challenge authentication containing a WWW-Authenticate header, and expect
the re-INVITE to contain an Authorization header.
The following example shows the Digest Authentication procedure, including computation
of user agent credentials:
1. The REGISTER request is sent to a Registrar/Proxy server for registration:
REGISTER sip:10.2.2.222 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.200
From: <sip: [email protected]>;tag=1c17940
To: <sip: [email protected]>
User-Agent: Sip-Gateway/MediaPack/v.6.60.010.006
CSeq: 1 REGISTER
Contact: sip:[email protected]:
Expires:3600
2. Upon receipt of this request, the Registrar/Proxy returns a 401 Unauthorized
response:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.2.1.200
From: <sip:[email protected] >;tag=1c17940
To: <sip:[email protected] >
Cseq: 1 REGISTER
Date: Mon, 30 Jul 2012 15:33:54 GMT
Server: Columbia-SIP-Server/1.17
Content-Length: 0
WWW-Authenticate: Digest realm="audiocodes.com",
nonce="11432d6bce58ddf02e3b5e1c77c010d2",
stale=FALSE,
algorithm=MD5
3. According to the sub-header present in the WWW-Authenticate header, the correct
REGISTER request is created.
4. Since the algorithm is MD5:
• The username is equal to the endpoint phone number "122".
• The realm return by the proxy is "audiocodes.com".
• The password from the ini file is "AudioCodes".
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• The equation to be evaluated is "122:audiocodes.com:AudioCodes". According to

the RFC, this part is called A1.
• The MD5 algorithm is run on this equation and stored for future usage.
• The result is "a8f17d4b41ab8dab6c95d3c14e34a9e1".
5. The par called A2 needs to be evaluated:
• The method type is "REGISTER".
• Using SIP protocol "sip".
• Proxy IP from ini file is "10.2.2.222".
• The equation to be evaluated is "REGISTER:sip:10.2.2.222".
• The MD5 algorithm is run on this equation and stored for future usage.
• The result is "a9a031cfddcb10d91c8e7b4926086f7e".
6. Final stage:
• A1 result: The nonce from the proxy response is
"11432d6bce58ddf02e3b5e1c77c010d2".
• A2 result: The equation to be evaluated is
"A1:11432d6bce58ddf02e3b5e1c77c010d2:A2".
• The MD5 algorithm is run on this equation. The outcome of the calculation is the
response needed by the device to register with the Proxy.
• The response is "b9c45d0234a5abf5ddf5c704029b38cf".
At this time, a new REGISTER request is issued with the following response:
Via: SIP/2.0/UDP 10.1.1.200
CSeq: 1 REGISTER
Contact: sip:[email protected]:
Expires:3600
Authorization: Digest, username: 122,
realm="audiocodes.com”,
nonce="11432d6bce58ddf02e3b5e1c77c010d2",
uri=”10.2.2.222”,
response=“b9c45d0234a5abf5ddf5c704029b38cf”
7. Upon receiving this request and if accepted by the Proxy, the Proxy returns a 200 OK
response, completing the registration transaction:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.200
Cseq: 1 REGISTER
Date: Thu, 26 Jul 2012 09:34:42 GMT
Server: Columbia-SIP-Server/1.17
Content-Length: 0
Contact: <sip:[email protected]>; expires="Thu, 26 Jul 2012
10:34:42 GMT"; action=proxy; q=1.00
Contact: <[email protected]:>; expires="Tue, 19 Jan 2038 03:14:07
GMT"; action=proxy; q=0.00
Expires: Thu, 26 Jul 2012 10:34:42 GMT

User's Manual 19. Coders and Profiles
19 Coders and Profiles

This section describes configuration of the coders and SIP profiles parameters.
19.1 Configuring Coders

The Coders page allows you to configure up to 10 voice coders for the device. Each coder
can be configured with packetization time (ptime), bit rate, payload type, and silence
suppression. The first coder configured in the table has the highest priority and is used by
the device whenever possible. If the remote side cannot use the first coder, the device
attempts to use the next coder in the table, and so on.
Notes:
• A specific coder can only be configured once in the table.
• If packetization time and/or rate are not specified, the default is applied.
• Only the packetization time of the first coder in the coder list is declared in
INVITE/200 OK SDP, even if multiple coders are defined.
• The device always uses the packetization time requested by the remote side for
sending RTP packets. If not specified, the packetization time is assigned the
default value.
• The value of several fields is hard-coded according to common standards (e.g.,
payload type of G.711 U-law is always 0). Other values can be set dynamically. If
no value is specified for a dynamic field, a default is assigned. If a value is
specified for a hard-coded field, the value is ignored.
• The G.722 coder provides Packet Loss Concealment (PLC) capabilities, ensuring
higher voice quality.
• For G.729, it's also possible to select silence suppression without adaptations.
• If G.729 is selected and silence suppression is disabled, the device includes
'annexb=no' in the SDP of the relevant SIP messages. If silence suppression is
enabled or set to 'Enable w/o Adaptations', 'annexb=yes' is included. An exception
to this logic is when the remote gateway is a Cisco device (IsCiscoSCEMode).
• The G.727 coder is currently not supported by MP-124 Rev. E.
• For defining groups of coders, which can be assigned to Tel and IP Profiles, see
'Configuring Coder Groups' on page 222.
• For information on V.152 and implementation of T.38 and VBD coders, see
'Supporting V.152 Implementation' on page 177.
• The Coders table can also be configured using the table ini file parameter,
CodersGroup.
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 To configure the device's coders:

1. Open the Coders page (Configuration tab > VoIP menu > Coders and Profiles
submenu > Coders).
Figure 19-1: Coders Table Page
2. From the 'Coder Name' drop-down list, select the required coder.
3. From the 'Packetization Time' drop-down list, select the packetization time (in msec)
for the selected coder. The packetization time determines how many coder payloads
are combined into a single RTP packet.
4. From the 'Rate' drop-down list, select the bit rate (in kbps) for the selected coder.
5. In the 'Payload Type' field, if the payload type (i.e., format of the RTP payload) for the
selected coder is dynamic, enter a value from 0 to 120 (payload types of 'well-known'
coders cannot be modified).
6. From the 'Silence Suppression' drop-down list, enable or disable the silence
suppression option for the selected coder.
7. Repeat steps 2 through 6 for the next optional coders.
8. Click Submit.
The table below lists the supported coders:
Table 19-1: Supported Coders
Coder Name Packetization Time Rate (kbps) Payload Silence

(msec) Type Suppression
G.711 A-law 10, 20 (default), 30, 40, 64 8  [0] Disable

[g711Alaw64k] 50, 60, 80, 100, 120  [1] Enable
G.711 U-law 10, 20 (default), 30, 40, 64 0  [0] Disable
[g711Ulaw64k] 50, 60, 80, 100, 120  [1] Enable
G.711A-law_VBD 10, 20 (default), 30, 40, 64 Dynamic (0- N/A
[g711AlawVbd] 50, 60, 80, 100, 120 127)
Default is
180
G.711U-law_VBD 10, 20 (default), 30, 40, 64 Dynamic (0- N/A
[g711UlawVbd] 50, 60, 80, 100, 120 127)
Default is
120
G.722 20 (default), 40, 60, 80, 64 (default) 9 N/A
[g722] 100, 120

Coder Name Packetization Time Rate (kbps) Payload Silence

(msec) Type Suppression
G.723.1 30 (default), 60, 90, 120,  [0] 5.3 4  [0] Disable

[g7231] 150 (default)  [1] Enable
 [1] 6.3
G.726 10, 20 (default), 30, 40,  [0] 16 Dynamic (0-  [0] Disable
[g726] 50, 60, 80, 100, 120  [1] 24 127)  [1] Enable
 [2] 32 Default is 23
(default)
 [3] 40
G.727 ADPCM 10, 20 (default), 30, 40, 16, 24, 32, 40 Dynamic (0-  [0] Disable
50, 60, 80, 100, 120 127)  [1] Enable
G.729 10, 20 (default), 30, 40, 8 18  [0] Disable
[g729] 50, 60, 80, 100  [1] Enable
 [2] Enable
w/o
Adaptation
s
T.38 N/A N/A N/A N/A
[t38fax]
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19.2 Configuring Coder Groups

The Coder Group Settings page allows you to define up to four groups of coders (termed
Coder Groups). For each Coder Group, you can define up to 10 coders configured with
packetization time (ptime), rate, payload type, and silence suppression. The first coder in
the Coder Group table has the highest priority and is used by the device whenever
possible. If the remote side cannot use the first coder, the device attempts to use the next
coder, and so on.
Coder Groups can be used as follows:
 Assigned to Tel Profiles in the Tel Profiles table (see Configuring Tel Profiles on page
223).
 Assigned to IP Profiles in the IP Profiles table (see 'Configuring IP Profiles' on page
225).
Notes:
• A specific coder can be selected only once per Coder Group.
• For a list of supported coders, see 'Configuring Coders' on page 219.
• The Coder Group Settings table can also be configured using the table ini file
parameter, CodersGroup.
 To configure Coder Groups:

1. Open the Coder Group Settings page (Configuration tab > VoIP menu > Coders and
Profiles submenu > Coders Group Settings).
Figure 19-2: Coder Group Settings Page
2. From the 'Coder Group ID' drop-down list, select a Coder Group ID.
3. From the 'Coder Name' drop-down list, select the first coder for the Coder Group.
4. From the 'Packetization Time' drop-down list, select the packetization time (in msec)
for the coder. The packetization time determines how many coder payloads are
combined into a single RTP packet.
5. From the 'Rate' drop-down list, select the bit rate (in kbps) for the coder you selected.
6. In the 'Payload Type' field, if the payload type (i.e., format of the RTP payload) for the
coder you selected is dynamic, enter a value from 0 to 120 (payload types of common
coders cannot be modified).

7. From the 'Silence Suppression' drop-down list, enable or disable the silence
suppression option for the coder you selected.
8. Repeat steps 3 through 7 for the next coders (optional).
9. Repeat steps 2 through 8 for the next coder group (optional).
19.3 Configuring Tel Profile

The Tel Profile Settings table allows you to define up to nine configuration profiles for Tel
calls. These profiles are termed Tel Profiles. The Tel Profile Settings table contains a list of
parameters, which can also be configured globally for all calls using their corresponding
"global" parameters. The only difference between the Tel Profile parameters and the global
parameters regarding description may be their default values.
Tel Profiles provide high-level adaptation when the device interworks between different
equipment and protocols (at both the Tel and IP sides), each of which may require different
handling by the device. Once configured, Tel Profiles can be assigned to specific channels
(endpoints). Therefore, Tel Profiles enable you to assign special configuration settings for
device handling of specific calls. For example, if specific channels require the use of the
G.711 coder, you can configure a Tel Profile with this coder and assign it to these
channels. Tel Profiles are assigned to channels in the <Endpoint Phone Number Table
(see Configuring Endpoint Phone Numbers on page 235).
The procedure below describes how to configure Tel Profiles using the Web interface.
Note: Tel Profiles can also be configured using the table ini file parameter, TelProfile
(see 'Configuration Parameters Reference' on page 475)
 To configure Tel Profiles:

1. Open the Tel Profile Settings page (Configuration tab > VoIP menu > Coders and
Profiles submenu > Tel Profile Settings).
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Figure 19-3: Tel Profile Settings
2. From the 'Profile ID' drop-down list, select the Tel Profile index.
3. In the 'Profile Name' field, enter an arbitrary name that enables you to easily identify
the Tel Profile.
4. From the 'Profile Preference' drop-down list, select the priority of the Tel Profile, where
1 is the lowest priority and 20 the highest. If both IP and Tel profiles apply to the same
call, the coders and other common parameters (noted by an asterisk in the description
of the parameter TelProfile) of the preferred Profile are applied to that call. If the
Preference of the Tel and IP Profiles is identical, the Tel Profile parameters are
applied.
Note: If the coder lists of both IP and Tel Profiles apply to the same call, only the
coders common to both are used. The order of the coders is determined by the
preference.
5. Configure the parameters as required. For a description of each parameter, refer to
the corresponding "global" parameter.

19.4 Configuring IP Profiles

The IP Profile Settings table allows you to define up to nine IP Profiles. An IP Profile is a
set of special call configuration behaviors relating to signaling and media (e.g., coder used)
applied to specific IP calls (inbound and/or outbound). Therefore, IP Profiles provide high-
level adaptation when the device interworks between different IP entities (for Tel and IP
sides), each of which may require different handling by the device. For example, if a
specific IP entity uses the G.711 coder only, you can configure an IP Profile with G.711 for
this IP entity.
Many of the parameters in the IP Profile Settings table have a corresponding "global"
parameter. If an IP Profile is not associated with specific calls, the settings of the global
parameters are applied to these calls.
IP Profiles can be assigned to the following configuration elements:
 IP Groups - see Configuring IP Groups on page 205
 Tel-to-IP routing rules (for Gateway / IP-to-IP application) - see Configuring Tel-to-IP
Routing Table on page 256
 IP-to-Tel routing rules (for Gateway / IP-to-IP application) - see Configuring IP-to-Tel
Routing Table on page 263
The device selects the IP Profile as follows:
 If different IP Profiles (not default) are assigned to the same specific calls in all these
tables, the device uses the IP Profile that has the highest preference level (as set in
the 'Profile Preference' field). If they have the same preference level, the device uses
the IP Profile assigned in the IP Group table.
 If different IP Profiles are assigned to these tables and one table is set to the default
IP Profile, the device uses the IP Profile that is not the default.
Note:
• IP Profiles can also be implemented when using a Proxy server (when the
AlwaysUseRouteTable parameter is set to 1).
• RxDTMFOption configures the received DTMF negotiation method: [-1] not
configured, use the global parameter; [0] don’t declare RFC 2833; [1] declare RFC
2833 payload type is SDP.
• You can also configure IP Profiles using the table ini file parameter, IPProfile (see
Configuration Parameters Reference on page 475).
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 To configure IP Profiles:
1. Open the IP Profile Settings page (Configuration tab > VoIP menu > Coders and
Profiles submenu > IP Profile Settings).
Figure 19-4: IP Profile Settings
2. From the 'Profile ID' drop-down list, select the IP Profile index.
3. In the 'Profile Name' field, enter an arbitrary name that allows you to easily identify the
IP Profile.
4. From the 'Profile Preference' drop-down list, select the priority of the IP Profile, where
'1' is the lowest priority and '20' is the highest. If both IP and Tel profiles apply to the
same call, the coders and other common parameters (noted by an asterisk) of the
preferred Profile are applied to that call. If the Preference of the Tel and IP Profiles is
identical, the Tel Profile parameters are applied.
Note: If the coder lists of both IP and Tel Profiles apply to the same call, only the
coders common to both are used. The order of the coders is determined by the
preference.

Table 19-2: IP Profile Parameters Description
Web: Profile ID Defines a unique index number for the IP Profile.

[IpProfile_Index]
Web: Profile Name (Optional) Defines a descriptive name for the IP Profile.
[IpProfile_ProfileName]
Common Parameters
Web: RTP IP DiffServ For a description, see the global parameter
[IpProfile_IPDiffServ] PremiumServiceClassMediaDiffServ.
Web: Signaling DiffServ For a description, see the global parameter
[IpProfile_SigIPDiffServ] PremiumServiceClassControlDiffServ.
Web: Disconnect on Broken For a description, see the global parameter
Connection DisconnectOnBrokenConnection.
[IpProfile_DisconnectOnBroken
Connection]
Web: Media IP Version For a description, see the global parameter
Preference MediaIPVersionPreference.
[IpProfile_MediaIPVersionPrefer
ence]
Web: Dynamic Jitter Buffer For a description, see the global parameter DJBufMinDelay.
Minimum Delay
[IpProfile_JitterBufMinDelay]
Web: Dynamic Jitter Buffer For a description, see the global parameter DJBufOptFactor.
Optimization Factor
[IpProfile_JitterBufOptFactor]
Web: RTP Redundancy Depth For a description, see the global parameter
[IpProfile_RTPRedundancyDept RTPRedundancyDepth.
h]
Web: Echo Canceler For a description, see the global parameter
[IpProfile_EnableEchoCanceller EnableEchoCanceller.
]
Web: Input Gain For a description, see the global parameter InputGain.
[IpProfile_InputGain]
Web: Voice Volume For a description, see the global parameter VoiceVolume.
[IpProfile_VoiceVolume]
Web: Symmetric MKI Negotiation For a description, see the global parameter
[IpProfile_EnableSymmetricMKI EnableSymmetricMKI.
]
Web: MKI Size For a description, see the global parameter
[IpProfile_MKISize] SRTPTxPacketMKISize.
Gateway Parameters
Web: Fax Signaling Method For a description, see the global parameter IsFaxUsed.
[IpProfile_IsFaxUsed]
Web: Play Ringback Tone to IP For a description, see the global parameter PlayRBTone2IP.
[IpProfile_PlayRBTone2IP]
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Web: Enable Early Media For a description, see the global parameter EnableEarlyMedia.
[IpProfile_EnableEarlyMedia]
Web: Copy Destination Number to For a description, see the global parameter
Redirect Number CopyDest2RedirectNumber.
[IpProfile_CopyDest2RedirectN
umber]
Web: Media Security Behavior For a description, see the global parameter
[IpProfile_MediaSecurityBehavi MediaSecurityBehaviour.
our]
Web: CNG Detector Mode For a description, see the global parameter CNGDetectorMode.
[IpProfile_CNGmode]
Web: Modems Transport Type For a description, see the global parameters
[IpProfile_VxxTransportType] V21ModemTransportType, V22ModemTransportType,
V23ModemTransportType, V32ModemTransportType, and
V34ModemTransportType.
Web: NSE Mode For a description, see the global parameter NSEMode.
[IpProfile_NSEMode]
Web: Number of Calls Limit Defines the maximum number of concurrent calls (incoming and
[IpProfile_CallLimit] outgoing). If the number of concurrent calls reaches this limit,
the device rejects any new incoming and outgoing calls
belonging to this IP Profile.
This parameter can also be set to the following:
 [-1] = (Default) No limitation on calls.
 [0] = Calls are rejected.
Note: For IP-to-IP calls, you can configure the device to route
calls to an alternative IP Group when this maximum number of
concurrent calls is reached. To do so, you need to add an
alternative routing rule in the Outbound IP Routing table that
reroutes the call to an alternative IP Group. You also need to
add a rule to the Reason for Alternative Routing table to initiate
an alternative rule for Tel-to-IP calls using cause 805.
Web: Progress Indicator to IP For a description, see the global parameter
[IpProfile_ProgressIndicator2IP] ProgressIndicator2IP.
Web: Profile Preference Defines the priority of the IP Profile, where "1" is the lowest and
[IpProfile_IpPreference] "20" the highest. If both IP and Tel Profiles apply to the same
call, the coders and other common parameters of the preferred
profile are applied to the call. If the preference of the Tel and IP
Profiles is identical, the Tel Profile parameters are applied.
Note: If the coder lists of both IP and Tel Profiles apply to the
same call, only the coders common to both are used. The order
of the coders is determined by the preference.
Web: Coder Group For a description, see the global parameter CodersGroup.
[IpProfile_CodersGroupID]
Web: Remote RTP Base UDP For a description, see the global parameter
Port RemoteBaseUDPPort.
[IpProfile_RemoteBaseUDPPort
]
Web: First Tx DTMF Option For a description, see the global parameter TxDTMFOption.
[IpProfile_FirstTxDtmfOption]

Web: Second Tx DTMF Option For a description, see the global parameter TxDTMFOption.
[IpProfile_SecondTxDtmfOption
]
Web: Declare RFC 2833 in SDP For a description, see the global parameter RxDTMFOption.
[IpProfile_RxDTMFOption]
Web: Enable Hold For a description, see the global parameter EnableHold.
[IpProfile_EnableHold]
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Part V
Gateway Application
User's Manual 20. Introduction
20 Introduction
This section describes configuration of the Gateway applications. The Gateway application
refers to IP-to-Tel call routing and vice versa.
Notes:
• In some areas of the Web interface, the term "GW" application refers to the
Gateway applications, respectively.
• The terms IP-to-Tel and Tel-to-IP refer to the direction of the call relative to the
device. IP-to-Tel refers to calls received from the IP network and destined to the
PBX (i.e., telephone connected directly or indirectly to the device); Tel-to-IP refers
to calls received from telephones connected directly to the device's FXS ports or
from the PBX, and destined for the IP network.
• FXO (Foreign Exchange Office) is the interface replacing the analog telephone
and connects to a Public Switched Telephone Network (PSTN) line from the
Central Office (CO) or to a Private Branch Exchange (PBX). The FXO is designed
to receive line voltage and ringing current, supplied from the CO or the PBX (just
like an analog telephone). An FXO VoIP device interfaces between the CO/PBX
line and the Internet.
• FXS (Foreign Exchange Station) is the interface replacing the Exchange (i.e., the
CO or the PBX) and connects to analog telephones, dial-up modems, and fax
machines. The FXS is designed to supply line voltage and ringing current to these
telephone devices. An FXS VoIP device interfaces between the analog telephone
devices and the Internet.
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User's Manual 21. Hunt Group
21 Hunt Group
This section describes the configuration of the device's channels, which entails assigning
them to Hunt Groups.
21.1 Configuring Endpoint Phone Numbers

The Endpoint Phone Number Table page allows you to activate the device's ports
(channels or endpoints), by defining telephone numbers for the endpoints and assigning
them to Hunt Groups and Tel Profiles.
Notes:
• Each endpoint must be assigned a unique phone number. In other words, no two
endpoints can have the same phone number.
• The number of endpoints depends on the MediaPack model (e.g., MP-118
displays 8 endpoints).
• You can also configure the endpoint phone numbers using the table ini file
parameter TrunkGroup (see 'Number Manipulation Parameters' on page 633).
 To configure the Endpoint Phone Number table:

1. Open the Endpoint Phone Number Table page (Configuration tab > VoIP menu >
GW and IP to IP submenu > Hunt Group submenu > Endpoint Phone Number).
Figure 21-1: Endpoint Phone Number Table Page
2. Configure the endpoint phone numbers according to the table below. You must enter a
number in the 'Phone Number' fields for each port that you want to use.
4. To save the changes to the flash memory, see 'Saving Configuration' on page 366.
To register an endpoint to a Proxy/Registrar server, click the Register button; to un-
register an endpoint, click Un-Register.
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Table 21-1: Endpoint Phone Number Table Parameters
Channel(s) Defines the device's channels (or ports) that you want to
[TrunkGroup_FirstBChannel] activate. Enter the channel numbers as labeled on the device's
[TrunkGroup_LastBChannel] rear panel. You can enter a range of channels, by using the
syntax n-m, where n represents the lower channel number and
m the higher channel number. For example, "1-4" specifies
channels 1 through 4.
Phone Number Defines the telephone number for the channel. For a range of
[TrunkGroup_FirstPhoneNumber] channels, enter only the first telephone number. Subsequent
channels are assigned the next consecutive telephone number.
For example, if you enter 400 for channels 1 to 4, then channel
1 is assigned phone number 400, channel 2 is assigned phone
number 401, and so on.
These phone numbers are also used for channel allocation for
IP-to-Tel calls if the Hunt Group’s 'Channel Select Mode'
parameter is set to By Dest Phone Number.
This value can include up to 50 characters.
Notes:
 If this field includes alphabetical characters and the phone
number is defined for a range of channels (e.g., 1-4), then
the phone number must end with a number (e.g., 'user1').
 Phone number must be entered only as digits, without any
other characters. For example, if you wish to enter the
phone number 555-1212, it must be entered as 5551212
without the hyphen (-). If the hyphen is entered, the entry is
invalid.
Hunt Group ID Defines a Hunt Group ID (1-99) to the channels. The same
[TrunkGroup_TrunkGroupNum] Hunt Group ID can be assigned to more than one group of
channels. The Hunt Group ID is used to define a group of
common channel behaviors that are used for routing IP-to-Tel
calls. If an IP-to-Tel call is assigned to a Hunt Group, the call is
routed to the channel(s) pertaining to that Hunt Group ID.
Notes:
 Once you have defined a Hunt Group, you must configure
the parameter PSTNPrefix (IP to Hunt Group Routing Table)
to assign incoming IP calls to the appropriate Hunt Group. If
you do not configure this table, calls cannot be established.
 You can define the method for which calls are assigned to
channels within the Hunt Groups, using the parameter
TrunkGroupSettings.
Tel Profile ID Defines a Tel Profile ID to the channels.
[TrunkGroup_ProfileId] Note: For configuring Tel Profiles, see 'Configuring Tel Profiles'
on page 223.

21.2 Configuring Hunt Group Settings

The Hunt Group Settings allows you to configure the following per Hunt Group:
 Channel select method by which IP-to-Tel calls are assigned to the Hunt Group's
channels.
 Registration method for registering Hunt Groups to selected Serving IP Group IDs.
Notes:
• For configuring Hunt Groups, see Configuring Endpoint Phone Numbers on page
235.
• The Hunt Group Settings table can also be configured using the table ini file
parameter, TrunkGroupSettings (see 'Number Manipulation Parameters' on page
633).
 To configure the Hunt Group Settings table:

1. Open the Hunt Group Settings page (Configuration tab > VoIP menu > GW and IP
to IP submenu > Hunt Group > Hunt Group Settings).
Figure 21-2: Hunt Group Settings Page
2. From the 'Index' drop-down list, select the range of entries that you want to edit.
3. Configure the Hunt Group as required. For a description of the parameters, see the
table below.
Table 21-2: Hunt Group Settings Parameters Description
Hunt Group ID Defines the Hunt Group ID that you want to configure.
[TrunkGroupSettings_TrunkGro
upId]
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Channel Select Mode Defines the method by which IP-to-Tel calls are assigned to the
[TrunkGroupSettings_ChannelS channels of the Hunt Group.
electMode]  [0] By Dest Phone Number = (Default) The channel is
selected according to the called (destination) number. If the
number is not located, the call is released. If the channel is
unavailable (e.g., busy), the call is put on call waiting (if call
waiting is enabled and no other call is on call waiting);
otherwise, the call is released.
 [1] Cyclic Ascending = The next available channel in the
Hunt Group, in ascending cyclic order is selected. After the
device reaches the highest channel number in the Hunt
Group, it selects the lowest channel number in the Hunt
Group, and then starts ascending again.
 [2] Ascending = The lowest available channel in the Hunt
Group is selected, and if unavailable, the next higher
channel is selected.
 [3] Cyclic Descending = The next available channel in
descending cyclic order is selected. The next lower channel
number in the Hunt Group is always selected. When the
device reaches the lowest channel number in the Hunt
Group, it selects the highest channel number in the Hunt
Group, and then starts descending again.
 [4] Descending = The highest available channel in the Hunt
Group is selected, and if unavailable, the next lower channel
is selected.
 [5] Dest Number + Cyclic Ascending = The channel is
selected according to the called number. If the called number
isn't found, the next available channel in ascending cyclic
order is selected.
Note: If the called number is located, but the port associated
with the number is busy, the call is released.
 [6] By Source Phone Number = The channel is selected
according to the calling number.
 [9] Ring to Hunt Group = The device allocates IP-to-Tel calls
to all the FXS ports (channels) in the Hunt Group. When a
call is received for the Hunt Group, all telephones connected
to the FXS ports belonging to the Hunt Group start ringing.
The call is eventually received by whichever telephone first
answers the call (after which the other phones stop ringing).
This option is applicable only to FXS interfaces.
 [11] Dest Number + Ascending = The device allocates a
channels to incoming IP-to-Tel calls as follows:
a. The device attempts to route the call to the channel that
is associated with the destination (called) number. If
located, the call is sent to that channel.
b. If the number is not located or the channel is unavailable
(e.g., busy), the device searches in ascending order for
the next available channel in the Trunk Group. If located,
the call is sent to that channel.
c. If all the channels are unavailable, the call is released.
Note: If this parameter is not configured for the Hunt Group,
then its channel select method is according to the global
parameter, ChannelSelectMode.

Registration Mode Defines the registration method for the Hunt Group:
[TrunkGroupSettings_Registrati  [1] Per Gateway = (Default) Single registration for the entire
onMode] device. This is applicable only if a default Proxy or Registrar
IP is configured and Registration is enabled (i.e., parameter
IsRegisterUsed is set to 1). In this mode, the SIP URI user
part in the From, To, and Contact headers is set to the value
of the global registration parameter, GWRegistrationName or
username if GWRegistrationName is not configured.
 [0] Per Endpoint = Each channel in the Hunt Group registers
individually. The registrations are sent to the 'Serving IP
Group ID' if defined in the table, otherwise, it is sent to the
default Proxy, and if no default Proxy, then to the Registrar
IP.
 [4] Don't Register = No registrations are sent by endpoints
pertaining to the Hunt Group. For example, if the device is
configured globally to register all its endpoints (using the
parameter ChannelSelectMode), you can exclude some
endpoints from being registered by assigning them to a Hunt
Group and configuring the Hunt Group registration mode to
'Don't Register'.
 [5] Per Account = Registrations are sent (or not) to an IP
Group, according to the settings in the Account table (see
'Configuring Account Table' on page 213).
An example is shown below of a REGISTER message for
registering endpoint "101" using the registration Per Endpoint
mode:
REGISTER sip:SipGroupName SIP/2.0
Via: SIP/2.0/UDP
10.33.37.78;branch=z9hG4bKac862428454
From: <sip:101@GatewayName>;tag=1c862422082
To: <sip:101@GatewayName>
CSeq: 3 REGISTER
Contact: <sip:[email protected]>;expires=3600
Expires: 3600
User-Agent: Sip-Gateway/v.6.60A.011.002
Content-Length: 0
The "SipGroupName" in the Request-URI is configured in the IP
Group table (see 'Configuring IP Groups' on page 205).
Notes:
 If this parameter is not configured, the registration is
performed according to the global registration parameter,
ChannelSelectMode.
 To enable Hunt Group registration, set the global parameter,
IsRegisterNeeded to 1. This is unnecessary for 'Per Account'
registration mode.
 If the device is configured globally to register Per Endpoint
and an endpoint group includes four FXO endpoints to
register Per Gateway, the device registers all endpoints
except the first four endpoints. The group of these four
endpoints sends a single registration request.
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Serving IP Group ID Assigns an IP Group to where INVITE messages received from

[TrunkGroupSettings_ServingIP this Hunt Group are sent. The actual destination to where these
Group] INVITE messages are sent is according to the Proxy Set ID
associated with the IP Group. The Request-URI host name in
the INVITE and REGISTER messages (except for 'Per Account'
registration modes) is set to the value of the 'SIP Group Name'
parameter configured in the IP Group table (see 'Configuring IP
Groups' on page 205).
Notes:
 If this parameter is not configured, the INVITE messages are
sent to the default Proxy or according to the Tel to IP
Routing (see 'Configuring Tel to IP Routing' on page 256).
 If the PreferRouteTable parameter is set to 1 (see
'Configuring Proxy and Registration Parameters' on page
216), the routing rules in the Outbound IP Routing table take
precedence over the selected Serving IP Group ID.
Gateway Name Defines the host name for the SIP From header in INVITE
[TrunkGroupSettings_GatewayN messages and for the From/To headers in REGISTER requests.
ame] Note: If this parameter is not configured, the global parameter,
SIPGatewayName is used.
Contact User Defines the user part for the SIP Contact URI in INVITE
[TrunkGroupSettings_ContactUs messages and for the From, To, and Contact headers in
er] REGISTER requests.
Notes:
 This parameter is applicable only if the 'Registration Mode'
parameter is set to 'Per Account' and registration through the
Account table is successful.
 If registration fails, the user part in the INVITE Contact
header is set to the source party number.
 The 'Contact User' parameter in the Account table overrides
this parameter (see 'Configuring Account Table' on page
213).
Trunk Group Name Defines a name for the Trunk Group. This name represents the
[TrunkGroupSettings_TrunkGro Trunk Group in the SIP 'tgrp' parameter of the outgoing INVITE
upName] messages (according to RFC 4904). For example:
sip:+16305550100;tgrp=TG-1;trunk-context=+1-
[email protected];user=phone
The valid value can be a string of up to 20 characters. By
default, no name is configured.
Notes:
 If this parameter is not configured, the Trunk Group decimal
number is used in the SIP 'tgrp' parameter.
 This feature is enabled by any of the following parameters:
 UseSIPtgrp
 UseBroadsoftDTG
 Currently, this parameter can only be configured using the ini
file.

User's Manual 22. Manipulation
22 Manipulation
This section describes the configuration of various manipulation processes.
22.1 Configuring General Settings

The General Settings page allows you to configure general manipulation parameters. For a
description of the parameters appearing on this page, see 'Configuration Parameters
 To configure the general manipulation parameters:

1. Open the General Settings page (Configuration tab > VoIP menu > GW and IP to IP
submenu > Manipulations submenu >General Settings).

22.2 Configuring Source/Destination Number

Manipulation Rules
You can configure rules for manipulating destination and/or source telephone numbers for
IP-to-Tel and Tel-to-IP calls. The following number manipulation tables are used for this:
 Tel-to-IP calls:
• Destination Phone Number Manipulation Table for Tel > IP Calls table (up to 120
entries)
• Source Phone Number Manipulation Table for Tel > IP Calls table (up to 20
entries)
 IP-to-Tel calls:
• Destination Phone Number Manipulation Table for IP > Tel Calls table (up to 120
entries)
• Source Phone Number Manipulation Table for IP > Tel Calls table (up to 20
entries)
The number manipulation tables provide two configuration areas:
 Matching characteristics (Rule) of incoming call, for example, prefix of destination
number.
 Manipulation operation (Action), for example, remove user-defined number of digits
from the left of the number.
If the incoming call matches the characteristics of a rule, then its manipulation action is
applied.
The device searches a matching manipulation rule starting from the first entry (i.e., top of
the table). In other words, a rule at the top of the table takes precedence over a rule
defined lower down in the table. Therefore, define more specific rules above more generic
rules. For example, if you enter 551 in Index 1 and 55 in Index 2, the device applies rule 1
to numbers that start with 551 and applies rule 2 to numbers that start with 550, 552, 553,
and so on until 559. However, if you enter 55 in Index 1 and 551 in Index 2, the device
applies rule 1 to all numbers that start with 55, including numbers that start with 551.
You can perform a second "round" (additional) of destination (NumberMapIP2Tel
parameter) and source (SourceNumberMapIP2Tel parameter) number manipulations for
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IP-to-Tel calls on an already manipulated number. The initial and additional number
manipulation rules are both configured in these tables. The additional manipulation is
performed on the initially manipulated number. Therefore, for complex number
manipulation schemes, you only need to configure relatively few manipulation rules in
these tables (that would otherwise require many rules). This feature is enabled using the
following parameters:
 PerformAdditionalIP2TELSourceManipulation for source number manipulation
 PerformAdditionalIP2TELDestinationManipulation for destination number manipulation
Telephone number manipulation can be useful, for example, for the following:
 Stripping or adding dialing plan digits from or to the number, respectively. For
example, a user may need to first dial 9 before dialing the phone number to indicate
an external line. This number 9 can then be removed by number manipulation before
the call is setup.
 Allowing or blocking Caller ID information according to destination or source prefixes.
For more information on Caller ID, see Configuring Caller Display Information on page
307.
Notes:
• Number manipulation can occur before or after a routing decision is made. For
example, you can route a call to a specific Hunt Group according to its original
number, and then you can remove or add a prefix to that number before it is
routed. To determine when number manipulation is performed, configure the 'IP to
Tel Routing Mode' parameter (RouteModeIP2Tel) described in 'Configuring IP to
Hunt Group Routing Table' on page 263, and 'Tel to IP Routing Mode' parameter
(RouteModeTel2IP) described in 'Configuring Tel to IP Routing' on page 256.
• The device manipulates the number in the following order: 1) strips digits from the
left of the number, 2) strips digits from the right of the number, 3) retains the
defined number of digits, 4) adds the defined prefix, and then 5) adds the defined
suffix.
• The source/destination number manipulation tables can also be configured using
the ini file:
1) Destination Phone Number Manipulation Table for IP > Tel Calls table:
NumberMapIP2Tel (ini)
2) Destination Phone Number Manipulation Table for Tel > IP Calls table:
NumberMapTel2IP (ini)
3) Source Phone Number Manipulation Table for IP > Tel Calls table:
SourceNumberMapIP2Tel (ini)
4) Source Phone Number Manipulation Table for Tel > IP Calls table:
SourceNumberMapTel2IP (ini)

 To configure number manipulation rules:

1. Open the required Number Manipulation page (Configuration tab > VoIP menu > GW
and IP to IP submenu > Manipulations submenu > Dest Number IP->Tel, Dest
Number Tel->IP, Source Number IP->Tel, or Source Number Tel->IP); the relevant
Manipulation table page is displayed.
Figure 22-1: Number Manipulation Table - Add Dialog Box
3. Click the Rule tab, and then configure the matching characteristics. For a description
of the parameters, see the table below.
4. Click the Action tab, and then configure the manipulation operation. For a description
The table below shows configuration examples of Tel-to-IP source phone number
manipulation rules, where:
 Rule 1: When the destination number has the prefix 03 (e.g., 035000), source number
prefix 201 (e.g., 20155), and from source IP Group ID 2, the source number is
changed to, for example, 97120155.
 Rule 2: When the source number has prefix 1001 (e.g., 1001876), it is changed to
587623.
 Rule 3: When the source number has prefix 123451001 (e.g., 1234510012001), it is
changed to 20018.
 Rule 4: When the source number has prefix from 30 to 40 and a digit (e.g., 3122), it is
changed to 2312.
 Rule 5: When the destination number has the prefix 6, 7, or 8 (e.g., 85262146),
source number prefix 2001, it is changed to 3146.
Parameter Rule 1 Rule 2 Rule 3 Rule 4 Rule 5
Source IP Group 2 0 - - -
Destination 03 * * [6,7,8]
Prefix
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Parameter Rule 1 Rule 2 Rule 3 Rule 4 Rule 5
Source Prefix 201 1001 123451001# [30-40]x 2001

Stripped Digits - 4 - - 5
from Left
Stripped Digits - - - 1 -
from Right
Prefix to Add 971 5 - 2 3
Suffix to Add - 23 8 - -
Number of - - 4 - -
Digits to Leave
Presentation Allowed Restricted - - -
Table 22-1: Number Manipulation Parameters Description
Matching Characteristics (Rule)

Web: Destination Prefix Defines the destination (called) telephone number prefix and/or suffix.
EMS: Prefix You can use special notations for denoting the prefix. For example,
[DestinationPrefix] [100-199](100,101,105) denotes a number that starts with 100 to 199
and ends with 100, 101 or 105. You can also use the $ sign to denote
calls without a called number. For a description of available notations,
see 'Dialing Plan Notation for Routing and Manipulation Tables' on
page 473.
Web/EMS: Source Prefix Defines the source (calling) telephone number prefix and/or suffix. You
[SourcePrefix] can use special notations for denoting the prefix. For example, [100-
199](100,101,105) denotes a number that starts with 100 to 199 and
ends with 100, 101 or 105. You can also use the $ sign to denote calls
without a calling number. For a description of available notations, see
'Dialing Plan Notation for Routing and Manipulation Tables' on page
473.
Web/EMS: Source IP Defines the source IP address of the caller. This is obtained from the
Address Contact header in the INVITE message.
[SourceAddress] Notes:
 This parameter is applicable only to the number manipulation tables
for IP-to-Tel calls.
 The source IP address can include the 'x' wildcard to represent
single digits. For example: 10.8.8.xx represents all IP addresses
between 10.8.8.10 to 10.8.8.99.
 The source IP address can include the asterisk (*) wildcard to
represent any number between 0 and 255. For example, 10.8.8.*
represents all IP addresses between 10.8.8.0 and 10.8.8.255.

Web: Source Host Prefix Defines the URI host name prefix of the incoming SIP INVITE message
[SrcHost] in the From header.
Notes:
 The asterisk (*) wildcard can be used to denote any prefix.
 If the P-Asserted-Identity header is present in the incoming INVITE
message, then the value of this parameter is compared to the P-
Asserted-Identity URI host name (instead of the From header).
Web: Destination Host Defines the Request-URI host name prefix of the incoming SIP INVITE
Prefix message.
[DestHost] Notes:
Web: Source Trunk Group Defines the source Hunt Group ID for Tel-to-IP calls. To denote all Hunt
[SrcTrunkGroupID] Groups, leave this field empty.
Notes:
 The value -1 indicates that this field is ignored in the rule.
for Tel-to-IP calls.
Web: Source IP Group Defines the IP Group from where the IP call originated. Typically, the IP
[SrcIPGroupID] Group of an incoming INVITE is determined or classified using the IP to
Hunt Group Routing Table. If not used (i.e., any IP Group), leave the
field empty.
Notes:
 The value -1 indicates that this field is ignored.
for Tel-to-IP calls.
Web: Destination IP Group Defines the IP Group to where the call is sent.
[DestIPGroupID] Notes:
 The value -1 indicates that this field is ignored.
 This parameter is applicable only to the Destination Phone Number
Manipulation Table for Tel -> IP Calls.
Operation (Action)
Web: Stripped Digits From Defines the number of digits to remove from the left of the telephone
Left number prefix. For example, if you enter 3 and the phone number is
EMS: Number Of Stripped 5551234, the new phone number is 1234.
Digits
[RemoveFromLeft]
Web: Stripped Digits From Defines the number of digits to remove from the right of the telephone
Right number prefix. For example, if you enter 3 and the phone number is
EMS: Number Of Stripped 5551234, the new phone number is 5551.
Digits
[RemoveFromRight]
Web: Prefix to Add Defines the number or string that you want added to the front of the
EMS: Prefix/Suffix To Add telephone number. For example, if you enter 9 and the phone number
[Prefix2Add] is 1234, the new number is 91234.
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Web: Suffix to Add Defines the number or string that you want added to the end of the
EMS: Prefix/Suffix To Add telephone number. For example, if you enter 00 and the phone number
[Suffix2Add] is 1234, the new number is 123400.
Web/EMS: Number of Defines the number of digits that you want to keep from the right of the
Digits to Leave phone number. For example, if you enter 4 and the phone number is
[LeaveFromRight] 00165751234, then the new number is 1234.
Web: Presentation Enables caller ID.
EMS: Is Presentation  Not Configured = Privacy is determined according to the Caller ID
Restricted table (see Configuring Caller Display Information on page 307).
[IsPresentationRestricted]  [0] Allowed = Sends Caller ID information when a call is made using
these destination/source prefixes.
 [1] Restricted = Restricts Caller ID information for these prefixes.
Notes:
 This field is applicable only to number manipulation tables for source
phone number manipulation.
 If this field is set to Restricted and the 'Asserted Identity Mode'
(AssertedIdMode) parameter is set to Add P-Asserted-Identity, the
From header in the INVITE message includes the following: From:
'anonymous' <sip: [email protected]> and 'privacy:
id' header.
22.3 Manipulating Number Prefix

The device supports a notation for adding a prefix where part of the prefix is first extracted
from a user-defined location in the original destination or source number. This notation is
entered in the 'Prefix to Add' field in the Number Manipulation tables (see 'Configuring
Source/Destination Number Manipulation' on page 241): x[n,l]y...
where,
 x = any number of characters/digits to add at the beginning of the number (i.e. first
digits in the prefix).
 [n,l] = defines the location in the original destination or source number where the digits
y are added:
• n = location (number of digits counted from the left of the number) of a specific
string in the original destination or source number.
• l = number of digits that this string includes.
 y = prefix to add at the specified location.
For example, assume that you want to manipulate an incoming IP call with destination
number +5492028888888 (area code 202 and phone number 8888888) to the number
0202158888888. To perform such a manipulation, the following configuration is required in
the Number Manipulation table:
1. The following notation is used in the 'Prefix to Add' field:
0[5,3]15
where,
• 0 is the number to add at the beginning of the original destination number.
• [5,3] denotes a string that is located after (and including) the fifth character (i.e.,
the first '2' in the example) of the original destination number, and its length being
three digits (i.e., the area code 202, in the example).

• 15 is the number to add immediately after the string denoted by [5,3] - in other
words, 15 is added after (i.e. to the right of) the digits 202.
2. The first seven digits from the left are removed from the original number, by entering
"7" in the 'Stripped Digits From Left' field.
Table 22-2: Example of Configured Rule for Manipulating Prefix using Special Notation
Parameter Rule 1
Destination Prefix +5492028888888
Source Prefix *
Source IP Address *
Stripped Digits from Left 7
Prefix to Add 0[5,3]15
In this configuration example, the following manipulation process occurs:

1. The prefix is calculated as 020215.
2. The first seven digits from the left are removed from the original number, thereby
changing the number to 8888888.
3. The prefix that was previously calculated is then added.
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22.4 SIP Calling Name Manipulations

The Calling Name Manipulations Tel2IP and Calling Name Manipulations IP2Tel tables
allow you to configure up to 20 manipulation rules for manipulating the calling name (i.e.,
caller ID) in SIP messages. This can include modifying or removing the calling name. SIP
calling name manipulation is applicable to Tel-to-IP and IP-to-Tel calls.
For example, assume that an incoming SIP INVITE message includes the following
header:
P-Asserted-Identity: "company:john" sip:[email protected]
Using the Calling Name Manipulations IP2Tel table, the text "company" can be changed to
"worker" in the outgoing INVITE, as shown below:
P-Asserted-Identity: "worker:john" sip:[email protected]
The calling name manipulation tables provide two configuration areas:
 Matching characteristics (Rule) of incoming call, for example, prefix of destination
number.
from the left of the calling name.
applied.
Notes:
• The Calling Name Manipulations Tel2IP table can also be configured using the
table ini file parameter, CallingNameMapTel2Ip.
• The Calling Name Manipulations IP2Tel table can also be configured using the
table ini file parameter, CallingNameMapIp2Tel.
 To configure calling name manipulation rules:

1. Open the required Calling Name Manipulations page (Configuration tab > VoIP menu
> GW and IP to IP submenu > Manipulations > Calling Name IP->Tel or Calling
Name Tel->IP).
Figure 22-2: Calling Name Manipulation IP2Tel - Rule Tab

5. Click the Submit button to save your changes.

Table 22-3: Calling Name Manipulation Parameters Description

Web: Destination Prefix Defines the destination (called) telephone number prefix and/or suffix.
You can use special notations for denoting the prefix. For example,
[100-199](100,101,105) denotes a number that starts with 100 to 199
and ends with 100, 101 or 105. You can also use the $ sign to denote
calls without a called number. For a description of available notations,
see 'Dialing Plan Notation for Routing and Manipulation Tables' on
page 473.
Web/EMS: Source Prefix Defines the source (calling) telephone number prefix and/or suffix. You
can use special notations for denoting the prefix. For example, [100-
199](100,101,105) denotes a number that starts with 100 to 199 and
ends with 100, 101 or 105. You can also use the $ sign to denote calls
without a calling number. For a description of available notations, see
'Dialing Plan Notation for Routing and Manipulation Tables' on page
473.
Web: Calling Name Prefix Defines the caller name (i.e., caller ID) prefix. You can use special
notations for denoting the prefix. For example, to denote any prefix, use
the asterisk (*) symbol or to denote calls without a calling name, use
the $ sign. For a description of available notations, see 'Dialing Plan
Notation for Routing and Manipulation Tables' on page 473.
Web: Source Trunk Group Defines the source Hunt Group ID for Tel-to-IP calls. To denote all Hunt
ID Groups, leave this field empty.
Notes:
 This parameter is applicable only to the Calling Name Manipulations
Tel2IP table.
Web: Source IP Group ID Defines the IP Group from where the IP call originated.
Notes:
Tel2IP table.
Web/EMS: Source IP Defines the source IP address of the caller, obtained from the Contact
Address header in the INVITE message.
Notes:
IP2Tel table.
 The source IP address can include the 'x' wildcard to represent
single digits. For example: 10.8.8.xx represents all IP addresses
between 10.8.8.10 to 10.8.8.99.
 The source IP address can include the asterisk (*) wildcard to
represent any number between 0 and 255. For example, 10.8.8.*
represents all IP addresses between 10.8.8.0 and 10.8.8.255.
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Web: Source Host Prefix Defines the URI host name prefix of the incoming SIP INVITE message
in the From header.
Notes:
IP2Tel table.
Asserted-Identity URI host name (instead of the From header).
Web: Destination Host Defines the Request-URI host name prefix of the incoming SIP INVITE
Prefix message.
Notes:
IP2Tel table.
Operation (Action)
Web: Stripped Digits From Defines the number of characters to remove from the left of the calling
Left name. For example, if you enter 3 and the calling name is
EMS: Number Of Stripped "company:john", the new calling name is "pany:john".
Digits
Web: Stripped Digits From Defines the number of characters to remove from the right of the calling
Right name. For example, if you enter 3 and the calling name is
EMS: Number Of Stripped "company:name", the new name is "company:n".
Digits
Web/EMS: Number of Defines the number of characters that you want to keep from the right
Digits to Leave of the calling name. For example, if you enter 4 and the calling name is
"company:name", the new name is "name".
Web: Prefix to Add Defines the number or string to add at the front of the calling name. For
EMS: Prefix/Suffix To Add example, if you enter ITSP and the calling name is "company:name",
the new name is ITSPcompany:john".
Web: Suffix to Add Defines the number or string to add at the end of the calling name. For
EMS: Prefix/Suffix To Add example, if you enter 00 and calling name is "company:name", the new
name is "company:name00".

22.5 Configuring Redirect Number IP to Tel

You can configure rules for manipulating the redirect number received in the incoming
message:
 Tel-to-IP redirect number manipulation: You can manipulate the prefix of the redirect
number, received from the Tel side, in the outgoing SIP Diversion, Resource-Priority,
or History-Info headers sent to the IP side. This is configured in the Redirect Number
Tel > IP table.
The redirect number manipulation tables provide two configuration areas:
 Matching characteristics (Rule) of incoming call, for example, prefix of redirect
number.
from the left of the redirect number.
applied.
Notes:
• If the device copies the received destination number to the outgoing SIP redirect
number (enabled by the CopyDest2RedirectNumber parameter), then no redirect
number Tel-to-IP manipulation is done.
• The manipulation rules are done in the following order: Stripped Digits From Left,
Stripped Digits From Right, Number of Digits to Leave, Prefix to Add, and then
Suffix to Add.
• The Redirect Prefix parameter is used before it is manipulated.
• The redirect number manipulation tables can also be configured using the ini file:
Redirect Number Tel to IP table - RedirectNumberMapTel2Ip (ini)
 To configure redirect number manipulation rules:

1. Open the redirect number manipulation table (Configuration tab > VoIP menu > GW
and IP to IP submenu > Manipulations > Redirect Number Tel > IP).
2. Click the Add button; the following dialog box appears (e.g., Redirect Number Tel > IP
table):
Figure 22-3: Redirect Number Manipulation (e.g., Tel to IP)
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Table 22-4: Redirect Number Manipulation Parameters Description

Web/EMS: Redirect Prefix Defines the redirect telephone number prefix. To denote any number,
[RedirectPrefix] use the wildcard asterisk (*) symbol.
Web/EMS: Destination Defines the destination (called) telephone number prefix. To denote
Prefix any number, use the wildcard asterisk (*) symbol.
[DestinationPrefix]
Web: Source Trunk Group Defines the Hunt Group from where the Tel call is received. To denote
ID any Hunt Group, leave this field empty.
[SrcTrunkGroupID] Notes:
 This parameter is applicable only to the Redirect Number Tel > IP
table.
Operation (Action)
Web: Stripped Digits From Defines the number of digits to remove from the left of the redirect
Left number prefix. For example, if you enter 3 and the redirect number is
EMS: Remove From Left 5551234, the new number is 1234.
[RemoveFromLeft]
Web: Stripped Digits From Defines the number of digits to remove from the right of the redirect
Right number prefix. For example, if you enter 3 and the redirect number is
EMS: Remove From Right 5551234, the new number is 5551.
[RemoveFromRight]
Web/EMS: Number of Defines the number of digits that you want to retain from the right of the
Digits to Leave redirect number.
[LeaveFromRight]
Web/EMS: Prefix to Add Defines the number or string that you want added to the front of the
[Prefix2Add] redirect number. For example, if you enter 9 and the redirect number is
1234, the new number is 91234.
Web/EMS: Suffix to Add Defines the number or string that you want added to the end of the
[Suffix2Add] redirect number. For example, if you enter 00 and the redirect number
is 1234, the new number is 123400.
Web: Presentation Enables caller ID.
EMS: Is Presentation  Not Configured = Privacy is determined according to the Caller ID
Restricted table (see Configuring Caller Display Information on page 307).
[IsPresentationRestricted]  [0] Allowed = Sends Caller ID information when a call is made using
these destination / source prefixes.
 [1] Restricted = Restricts Caller ID information for these prefixes.
Note: If 'Presentation' is set to 'Restricted' and the AssertedIdMode
parameter is set to Add P-Asserted-Identity, the From header in the
INVITE message includes the following: From: 'anonymous' <sip:
[email protected]> and 'privacy: id' header.

22.6 Mapping NPI/TON to SIP Phone-Context

The Phone-Context table page allows you to map Numbering Plan Indication (NPI) and
Type of Number (TON) to the SIP 'phone-context' parameter. The 'phone-context'
parameter appears in the standard SIP headers where a phone number is used (i.e.,
Request-URI, To, From, and Diversion). When a call is received from the Tel side, the NPI
and TON are compared against the table and the matching 'phone-context' value is used in
the outgoing SIP INVITE message. The same mapping occurs when an INVITE with a
'phone-context' parameter is received.
For example, for a Tel-to-IP call with NPI/TON set as E164 National (values 1/2), the
device sends the following SIP INVITE URI:
sip:12365432;phone-context= na.e.164.nt.com
This is configured for entry 3 in the figure below. In the opposite direction (IP-to-Tel call), if
the incoming INVITE contains this 'phone-context' (e.g. "phone-context= na.e.164.nt.com"),
the NPI/TON of the called number in the outgoing Setup message is changed to E164
National.
 To configure NPI/TON to SIP phone-context rules:

1. Open the Phone Context Table page (Configuration tab > VoIP menu > GW and IP
to IP submenu > Manipulations > Phone Context).
Figure 22-4: Phone Context Table Page
table below.
Notes:
• You can configure multiple rows with the same NPI/TON or same SIP 'phone-
context'. In such a configuration, a Tel-to-IP call uses the first matching rule in the
table.
• The Phone Context table can also be configured using the table ini file parameter,
PhoneContext (see 'Number Manipulation Parameters' on page 633).
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Table 22-5: Phone-Context Parameters Description
Add Phone Context As Prefix Determines whether the received SIP 'phone-context' parameter is
[AddPhoneContextAsPrefix] added as a prefix to the outgoing called and calling numbers.
 [1] Enable
NPI Defines the Number Plan Indicator (NPI).
[PhoneContext_Npi]  [0] Unknown (default)
 [1] E.164 Public
 [9] Private
TON Defines the Type of Number (TON).
[PhoneContext_Ton]  If you selected Unknown as the NPI, you can select Unknown [0].
 If you selected Private as the NPI, you can select one of the
following:
 [0] Unknown
 [1] Level 2 Regional
 [2] Level 1 Regional
 [3] PSTN Specific
 [4] Level 0 Regional (Local)
 If you selected E.164 Public as the NPI, you can select one of the
following:
 [0] Unknown
 [1] International
 [2] National
 [3] Network Specific
 [4] Subscriber
 [6] Abbreviated
Phone Context Defines the SIP 'phone-context' URI parameter.
[PhoneContext_Context]

User's Manual 23. Routing
23 Routing
This section describes the configuration of call routing rules.
23.1 Configuring General Routing Parameters

The Routing General Parameters page allows you to configure general routing parameters.
For a description of these parameters, see 'Configuration Parameters Reference' on page
475.
 To configure general routing parameters:

1. Open the Routing General Parameters page (Configuration tab > VoIP menu > GW
and IP to IP submenu > Routing submenu > General Parameters).
Figure 23-1: General Parameters

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23.2 Configuring Tel to IP Routing

The Tel to IP Routing page allows you to configure up to 50 Tel-to-IP call routing rules. The
device uses these rules to route calls from the Tel to a user-defined IP destination.
The Tel to IP Routing table provides two configuration areas:
 Matching Characteristics: Characteristics of the incoming call. If the call
characteristics match a table entry, the routing rule is used to route the call to the
specified destination. One or more characteristics can be defined for the rule:
• Source and destination Request-URI host name prefix
• Source Hunt Group (from where the call is received)
• Source (calling) and destination (called) telephone number prefix and suffix
 Destination: If the call matches the configured characteristics, the device routes the
call to an IP destination. If no characteristics match is found in the table, the call is
rejected. The destination can be any of the following:
• IP address in dotted-decimal notation.
• Fully Qualified Domain Name (FQDN).
• E.164 Telephone Number Mapping (ENUM service).
• IP Group, where the call is routed to the IP address configured for the Proxy Set
associated with the IP Group (configured in 'Configuring IP Groups' on page 205).
Notes: When using a proxy server, you do not need to configure this table, unless
you require one of the following:
• Fallback (alternative) routing if communication is lost with the proxy server.
• IP security, whereby the device routes only received calls whose source IP
addresses are defined in this table. IP security is enabled using the
SecureCallsFromIP parameter.
• Filter Calls to IP feature: the device checks this table before a call is routed to the
proxy server. However, if the number is not allowed, i.e., the number does not
exist in the table or a Call Restriction (see below) routing rule is applied, the call is
released.
• Obtain different SIP URI host names (per called number).
• Assign IP Profiles to calls.
• For this table to take precedence over a proxy for routing calls, you need to set the
parameter PreferRouteTable to 1. The device checks the 'Destination IP Address'
field in this table for a match with the outgoing call; a proxy is used only if a match
is not found.
In addition to basic outbound IP routing, this table supports the following features:
 Least Cost Routing (LCR): If the LCR feature is enabled, the device searches the
routing table for matching routing rules and then selects the one with the lowest call
cost. The call cost of the routing rule is done by assigning it a Cost Group. For
configuring Cost Groups, see 'Least Cost Routing' on page 195. If two routing rules
have identical costs, then the rule appearing higher up in the table (i.e., first-matched
rule) is used. If a selected route is unavailable, the device uses the next least-cost
routing rule. However, even if a matched rule is not assigned a Cost Group, the device
can select it as the preferred route over other matched routing rules with Cost Groups,
according to the settings of the LCR parameter, LCRDefaultCost (see 'Enabling LCR
and Configuring Default LCR' on page 197).

 Call Forking: If the Tel-to-IP Call Forking feature is enabled, the device can send a
Tel call to multiple IP destinations. An incoming Tel call with multiple matched routing
rules (e.g., all with the same source prefix numbers) can be sent (forked) to multiple IP
destinations if the rules are defined with a Forking Group in the table. The call is
established with the first IP destination that answers the call.
 Call Restriction: Rejects calls whose matching routing rule is configured with the
destination IP address of 0.0.0.0.
 Always Use Routing Table: Even if a proxy server is used, the SIP Request-URI
host name in the outgoing INVITE message is obtained from this table. Using this
feature, you can assign a different SIP URI host name for different called and/or
calling numbers. This feature is enabled using the AlwaysUseRouteTable parameter.
 IP Profiles: IP Profiles can be assigned to destination addresses (also when a proxy
is used).
 Alternative Routing (when a proxy isn't used): An alternative IP destination can be
configured for a specific call. To associate an alternative IP address to a called
telephone number prefix, assign it with an additional entry with a different IP address,
or use an FQDN that resolves into two IP addresses. For more information on
alternative routing, see 'Alternative Routing for Tel-to-IP Calls' on page 268.
Notes:
• The maximum number of alternative routing rules that can be configured for each
routing rule in the table is three.
• Outbound IP routing can be performed before or after number manipulation. This
is configured using the RouteModeTel2IP parameter, as described below.
• The Tel to IP Routing can also be configured using the table ini file parameter,
Prefix.
 To configure Tel-to-IP routing rules:

1. Open the Tel to IP Routing page (Configuration tab > VoIP menu > GW and IP to IP
submenu > Routing > Tel to IP Routing).
2. From the 'Routing Index' drop-down list, select the range of entries that you want to
add.
3. Configure the routing rule as required. For a description of the parameters, see the
table below.
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The table below shows configuration examples of Tel-to-IP routing rules, where:
 Rule 1 and 2 (Least Cost Routing rule): For both rules, the called (destination)
phone number prefix is 10, the caller's (source) phone number prefix is 100, and the
call is assigned IP Profile ID 1. However, Rule 1 is assigned a cheaper Cost Group
than Rule 2, and therefore, the call is sent to the destination IP address (10.33.45.63)
associated with Rule 1.
 Rule 3 (IP Group destination rule): For all callers (*), if the called phone number
prefix is 20, the call is sent to IP Group 1 (whose destination is the IP address
configured for its associated Proxy Set ID).
 Rule 4 (domain name destination rule): If the called phone number prefix is 5, 7, 8,
or 9 and the caller belongs to Hunt Group ID 1, the call is sent to domain.com.
 Rule 5 (block rule): For all callers (*), if the called phone number prefix is 00, the call
is rejected (discarded).
 Rule 6, Rule 7, and Rule 8 (Forking Group rule): For all callers (*), if the called
phone number prefix is 100, the call is sent to Rule 7 and 9 (belonging to Forking
Group "1"). If their destinations are unavailable and alternative routing is enabled, the
call is sent to Rule 8 (Forking Group "2").

Table 23-1: Example of Tel-to-IP Source Phone Number Manipulation Rules
Parameter Rule 1 Rule 2 Rule Rule 4 Rule 5 Rule 6 Rule 7 Rule 8

3
Src. Trunk * 0 1 - - - - -
Group ID
Src. Trunk - - * 1 - * * *
Group ID
Dest. Phone 10 10 20 [5,7-9] 00 100 100 100
Prefix
Source 100 100 * * * * * *
Phone Prefix
Dest. IP 10.33.45.63 10.33.45.50 - domain.com 0.0.0.0 10.33.45.68 10.33.45.67 domain.com
Address
Dest IP - - 1 - - - - -
Group ID
IP Profile ID 1 1 - - - - - -
Cost Group Weekend Weekend_B - - - - - -
ID
Forking - - - 1 2 1
Group
Table 23-2: Tel-to-IP Routing Table Parameters
Matching Call Characteristics

Web/EMS: Tel to IP Routing Determines whether to route received calls to an IP destination before
Mode or after manipulation of the destination number.
[RouteModeTel2IP]  [0] Route calls before manipulation = Calls are routed before the
number manipulation rules are applied (default).
 [1] Route calls after manipulation = Calls are routed after the
number manipulation rules are applied.
Notes:
 This parameter is not applicable if outbound proxy routing is used.
 For number manipulation, see 'Configuring Source/Destination
Number Manipulation' on page 241.
Web: Src. Trunk Group ID Defines the Hunt Group from where the call is received.
EMS: Source Trunk Group ID Note: To denote any Hunt Group, use the asterisk (*) symbol.
[PREFIX_SrcTrunkGroupID]
Web: Dest. Phone Prefix Defines the prefix and/or suffix of the called (destination) telephone
EMS: Destination Phone number. The suffix is enclosed in parenthesis after the suffix value.
Prefix You can use special notations for denoting the prefix. For example,
[PREFIX_DestinationPrefix] [100-199](100,101,105) denotes a number that starts with 100 to 199
and ends with 100, 101 or 105. To denote any prefix, use the asterisk
(*) symbol or to denote calls without a called number, use the $ sign.
For a description of available notations, see 'Dialing Plan Notation for
Routing and Manipulation Tables' on page 473.
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The number can include up to 50 digits.
Web/EMS: Source Phone Defines the prefix and/or suffix of the calling (source) telephone
Prefix number. You can use special notations for denoting the prefix. For
[PREFIX_SourcePrefix] example, [100-199](100,101,105) denotes a number that starts with
100 to 199 and ends with 100, 101 or 105. To denote any prefix, use
the asterisk (*) symbol or to denote calls without a calling number,
use the $ sign. For a description of available notations, see 'Dialing
Plan Notation for Routing and Manipulation Tables' on page 473.
The number can include up to 50 digits.
Operation (IP Destination)
Web: Dest. IP Address Defines the IP address (in dotted-decimal notation or FQDN) to where
EMS: Address the call is sent. If an FQDN is used (e.g., domain.com), DNS
[PREFIX_DestAddress] resolution is done according to the DNSQueryType parameter.
For ENUM-based routing, enter the string value "ENUM". The device
sends an ENUM query containing the destination phone number to an
external DNS server, configured in the Multiple Interface table. The
ENUM reply includes a SIP URI which is used as the Request-URI in
the subsequent outgoing INVITE and for routing (if a proxy is not
used). To configure the type of ENUM service (e.g., e164.arpa), use
the EnumService parameter.
Notes:
 This field and any value assigned to it is ignored if you have
configured a destination IP Group for this routing rule (in the 'Dest
IP Group ID' field).
 To reject calls, enter the IP address 0.0.0.0. For example, if you
want to prohibit international calls, then in the 'Dest Phone Prefix'
field, enter 00 and in the 'Dest IP Address' field, enter 0.0.0.0.
 For routing calls between phones connected to the device (i.e.,
local routing), enter the device's IP address.
 When the device's IP address is unknown (e.g., when DHCP is
used), enter IP address 127.0.0.1.
 When using domain names, enter the DNS server's IP address or
alternatively, configure these names in the Internal DNS table (see
'Configuring the Internal DNS Table' on page 142).
 The IP address can include the following wildcards:
 "x": represents single digits. For example, 10.8.8.xx denotes
all addresses between 10.8.8.10 and 10.8.8.99.
 "*": represents any number between 0 and 255. For example,
10.8.8.* denotes all addresses between 10.8.8.0 and
10.8.8.255.
Web: Port Defines the destination port to where you want to route the call.
EMS: Destination Port
[PREFIX_DestPort]
Web/EMS: Transport Type Defines the transport layer type for sending the IP call:
[PREFIX_TransportType]  [-1] Not Configured
 [0] UDP
 [1] TCP
 [2] TLS
Note: When set to Not Configured (-1), the transport type defined by
the SIPTransportType parameter is used.

Web: Dest IP Group ID Defines the IP Group to where you want to route the call. The SIP
EMS: Destination IP Group INVITE message is sent to the IP address defined for the Proxy Set
ID ID associated with the IP Group.
[PREFIX_DestIPGroupID] Notes:
 If you select an IP Group, you do not need to configure a
destination IP address. However, if both parameters are
configured in this table, the INVITE message is sent only to the IP
Group (and not the defined IP address).
 If the parameter AlwaysUseRouteTable is set to 1 (see
'Configuring IP Groups' on page 205), then the Request-URI host
name in the INVITE message is set to the value defined for the
parameter 'Dest. IP Address' (above); otherwise, if no IP address
is defined, it is set to the value of the parameter 'SIP Group Name'
(defined in the IP Group table).
 This parameter is used as the 'Serving IP Group' in the Account
table for acquiring authentication user/password for this call (see
'Configuring Account Table' on page 213).
 For defining Proxy Set ID's, see 'Configuring Proxy Sets Table' on
page 208.
IP Profile ID Assigns an IP Profile ID to this IP destination call. This allows you to
[PREFIX_ProfileId] assign numerous configuration attributes (e.g., voice codes) per
routing rule. To configure IP Profiles, see 'Configuring IP Profiles' on
page 225.
Status Displays the connectivity status of the routing rule's IP destination. If
there is connectivity with the destination, this field displays "OK" and
the device uses this routing rule if required.
The routing rule is not used if any of the following is displayed:
 "n/a" = The destination IP Group is unavailable
 "No Connectivity" = No connection with the destination (no
response to the ping or SIP OPTIONS).
 "QoS Low" = Poor Quality of Service (QoS) of the destination.
 "DNS Error" = No DNS resolution. This status is applicable only
when a domain name is used (instead of an IP address).
 "Unavailable" = The destination is unreachable due to networking
issues.
Web/EMS: Charge Code Assigns a Charge Code to the routing rule. To configure Charge
[PREFIX_MeteringCode] Codes, see Configuring Charge Codes Table on page 302.
Note: This parameter is applicable only to FXS interfaces.
Cost Group ID Assigns a Cost Group with the routing rule for determining the cost of
[PREFIX_CostGroup] the call. To configure Cost Groups, see 'Configuring Cost Groups' on
page 199.
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Forking Group Defines a forking group ID for the routing rule. This enables forking of
[PREFIX_ForkingGroup] incoming Tel calls to two or more IP destinations. The device sends
simultaneous INVITE messages and handles multiple SIP dialogs
until one of the calls is answered. When a call is answered, the other
calls are dropped.
If all matched routing rules belong to the same Forking Group
number, the device sends an INVITE to all the destinations belonging
to this group and according to the following logic:
 If matched routing rules belong to different Forking Groups, the
device sends the call to the Forking Group of the first matched
routing rule. If the call cannot be established with any of the
destinations associated with this Forking Group and alternative
routing is enabled, the device forks the call to the Forking Group of
the next matched routing rules as long as the Forking Group is
defined with a higher number than the previous Forking Group.
For example:
 Table index entries 1 and 2 are defined with Forking Group "1",
and index entries 3 and 4 with Forking Group "2": The device first
sends the call according to index entries 1 and 2, and if
unavailable and alternative routing is enabled, sends the call
according to index entries 3 and 4.
 Table index entry 1 is defined with Forking Group "2", and index
entries 2, 3, and 4 with Forking Group "1": The device sends the
call according to index entry 1 only and ignores the other index
entries even if the destination is unavailable and alternative routing
is enabled. This is because the subsequent index entries are
defined with a Forking Group number that is lower than that of
index entry 1.
 Table index entry 1 is defined with Forking Group "1", index entry
2 with Forking Group "2", and index entries 3 and 4 with Forking
Group "1": The device first sends the call according to index
entries 1, 3, and 4 (all belonging to Forking Group "1"), and if the
destination is unavailable and alternative routing is enabled, the
device sends the call according to index entry 2.
 Table index entry 1 is defined with Forking Group "1", index entry
2 with Forking Group "3", index entry 3 with Forking Group "2",
and index entry 4 with Forking Group "1": The device first sends
the call according to index entries 1 and 4 (all belonging to Forking
Group "1"), and if the destination is unavailable and alternative
routing is enabled, the device sends the call according to index
entry 2 (Forking Group "3"). Even if index entry 2 is unavailable
and alternative routing is enabled, the device ignores index entry 3
because it belongs to a Forking Group that is lower than index
entry 2.
Notes:
 To enable Tel-to-IP call forking, set the 'Tel2IP Call Forking Mode'
(Tel2IPCallForkingMode) parameter to Enable.
 When the UseDifferentRTPportAfterHold parameter is enabled,
every forked call is sent with a different RTP port. Thus, ensure
that the device has available RTP ports for these forked calls.

23.3 Configuring IP to Hunt Group Routing Table

The IP to Hunt Group Routing Table page allows you to configure up to 24 inbound call
routing rules:
 For IP-to-Tel routing: This table is used to route incoming IP calls to Hunt Groups. The
specific channel pertaining to the Hunt Group to which the call is routed is determined
according to the Hunt Group's channel selection mode. The channel selection mode
can be defined per Hunt Group (see 'Configuring Hunt Group Settings' on page 237)
or for all Hunt Groups using the global parameter ChannelSelectMode.
The IP to Hunt Group Routing Table provides two configuration areas:
 Matching characteristics of incoming IP call, for example, prefix of destination number.
 Operation (destination), for example, sends to a specific Hunt Group.
If the incoming call matches the characteristics of a rule, then the call is sent to the
destination configured for that rule.
The device also supports alternative routing if the Hunt Group is unavailable:
 If a call release reason is received for a specific IP-to-Tel call and this reason is
configured for alternative IP-to-Tel routing, then the device re-routes the call to an
alternative Hunt Group. The alternative route is configured in this table as an
additional row (below the main routing rule) with the same call characteristics, but with
a destination to a different Hunt Group. For more information on IP-to-Tel alternative
routing, see 'Alternative Routing to Trunk upon Q.931 Call Release Cause Code' on
page 271.
 The device can re-route (i.e., call redirection) IP-to-Tel calls to an alternative IP
destination using SIP 3xx responses. For more information, see 'Alternative Routing to
IP Destinations upon Busy Trunk' on page 272.
The device automatically re-routes an IP-to-Tel call to a different physical FXO port if the
initially destined FXO port within the same Hunt Group is detected as out of service (e.g.,
physically disconnected). When the physical FXO port is disconnected, the device sends
the SNMP trap, GWAPP_TRAP_BUSYOUT_LINK notifying of the out-of-service state for
the specific FXO line. When the FXO port is physically reconnected, this trap is sent
notifying of the back-to-service state.
Note: You can also configure the IP to Hunt Group Routing Table using the table ini
file parameter, PSTNPrefix (see 'Number Manipulation Parameters' on page 633).
 To configure IP-to-Tel routing rules:

1. Open the IP to Hunt Group Routing Table page (Configuration tab > VoIP menu >
GW and IP to IP submenu > Routing > IP to Hunt Group Routing).
Figure 23-2: Inbound IP Routing Table Page
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The previous figure displays the following configured routing rules:

• Rule 1: If the incoming IP call destination phone prefix is between 10 and 19, the
call is assigned settings configured for IP Profile ID 2 and routed to Hunt Group
ID 1.
• Rule 2: If the incoming IP call destination phone prefix is between 501 and 502
and source phone prefix is 101, the call is assigned settings configured for IP
Profile ID 1 and routed to Hunt Group ID 2.
• Rule 3: If the incoming IP call has a From URI host prefix as domain.com, the call
is routed to Hunt Group ID 3.
2. Configure the routing rule, as required. For a description of the parameters, see the
table below.
Table 23-3: IP-to-Tel Routing Table Description
IP to Tel Routing Mode Determines whether to route the incoming IP call before or after
[RouteModeIP2Tel] manipulation of destination number, configured in 'Configuring
Source/Destination Number Manipulation' on page 241.
 [0] Route calls before manipulation = (Default) Incoming IP calls are
routed before number manipulation.
 [1] Route calls after manipulation = Incoming IP calls are routed after
number manipulation.
Matching Characteristics
Web: Dest. Host Prefix Defines the Request-URI host name prefix of the incoming SIP INVITE
[DestPrefix] message. If this routing rule is not required, leave the field empty.
Note: The asterisk (*) wildcard can be used to depict any prefix.
Web: Source Host Prefix Defines the From URI host name prefix of the incoming SIP INVITE
[SrcHostPrefix] message. If this routing rule is not required, leave the field empty.
Notes:
 The asterisk (*) wildcard can be used to depict any prefix.
Asserted-Identity URI host name (and not the From header).
Web: Dest. Phone Prefix Defines the prefix or suffix of the called (destined) telephone number. You
[DestHostPrefix] can use special notations for denoting the prefix. For example, [100-
199](100,101,105) denotes a number that starts with 100 to 199 and ends
with 100, 101 or 105. To denote any prefix, use the asterisk (*) symbol or
to denote calls without a called number, use the $ sign. For a description
of available notations, see 'Dialing Plan Notation for Routing and
Manipulation Tables' on page 473.
The prefix can include up to 49 digits.
Web: Source Phone Defines the prefix or suffix of the calling (source) telephone number. You
Prefix can use special notations for denoting the prefix. For example, [100-
[SourcePrefix] 199](100,101,105) denotes a number that starts with 100 to 199 and ends
with 100, 101 or 105. To denote any prefix, use the asterisk (*) symbol or
to denote calls without a calling number, use the $ sign. For a description
of available notations, see 'Dialing Plan Notation for Routing and
Manipulation Tables' on page 473.
The prefix can include up to 49 digits.

Web: Source IP Address Defines the source IP address of the incoming IP call that can be used for
[SourceAddress] routing decisions.
The IP address must be configured in dotted-decimal notation (e.g.,
10.8.8.5); not as an FQDN.
Notes:
 The source IP address is obtained from the Contact header in the
INVITE message.
 You can configure from where the source IP address is obtained,
using the SourceIPAddressInput parameter.
 The source IP address can include the following wildcards:
 "x": denotes single digits. For example, 10.8.8.xx represents all
the addresses between 10.8.8.10 and 10.8.8.99.
 "*": denotes any number between 0 and 255. For example,
10.8.8.* represents all addresses between 10.8.8.0 and
10.8.8.255.
Operation (Destination)
Web: Hunt Group ID Defines the Hunt Group to where the incoming SIP call is sent.
[TrunkGroupId]
Web: IP Profile ID Assigns an IP Profile (configured in 'Configuring IP Profiles' on page 225)

[ProfileId] to the call.
Web: Source IP Group ID Defines the IP Group associated with the incoming IP call. This is the IP
[SrcIPGroupID] Group that sent the INVITE message. This IP Group can later be used as
the 'Serving IP Group' in the Account table for obtaining authentication
user name/password for this call (see 'Configuring Account Table' on
page 213).
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23.4 IP Destinations Connectivity Feature

The device can be configured to check the integrity of the connectivity to IP destinations of
Tel-to-IP routing rules in the Outbound IP Routing table. The IP Connectivity feature can be
used for the Alternative Routing feature, whereby the device attempts to re-route calls from
unavailable Tel-to-IP routing destinations to available ones (see 'Alternative Routing Based
on IP Connectivity' on page 268).
The device supports the following methods for checking the connectivity of IP destinations:
 Network Connectivity: The device checks the network connectivity of the IP
destination using one of the following methods configured by the 'Alt Routing Tel to IP
Connectivity Method' parameter:
• Ping: The device periodically (every seven seconds) pings the IP destination.
• SIP OPTIONS: The device sends "keep-alive" SIP OPTIONS messages to the IP
destination. If the device receives a SIP 200 OK in response, it considers the
destination as available. If the destination does not respond to the OPTIONS
message, then it is considered unavailable. You can configure the time interval
for sending these OPTIONS messages, using the 'Alt Routing Tel to IP Keep
Alive Time' parameter.
These parameters are configured in the Routing General Parameters page
(Configuration tab > VoIP menu > GW and IP to IP > Routing > General
Parameters), as shown below:
Figure 23-3: IP Connectivity Method in Routing General Parameters Page
 Quality of Service (QoS): You can enable the device to check the QoS of IP
destinations. The device measures the QoS according to RTCP statistics of previously
established calls with the IP destination. The RTCP includes packet delay (in
milliseconds) and packet loss (in percentage). If these measured statistics exceed a
user-defined threshold, the destination is considered unavailable. Note that if call
statistics is not received within two minutes, the QoS data is reset. These thresholds
are configured using the following parameters:
• 'Max Allowed Packet Loss for Alt Routing' (IPConnQoSMaxAllowedPL): defines
the threshold value for packet loss after which the IP destination is considered
unavailable.
• 'Max Allowed Delay for Alt Routing' (IPConnQoSMaxAllowedDelay): defines the
threshold value for packet delay after which the IP destination is considered
unavailable
These parameters are configured in the Routing General Parameters page, as shown
below:
Figure 23-4: IP QoS Thresholds in Routing General Parameters Page
 DNS Resolution: When a host name (FQDN) is used (instead of an IP address) for
the IP destination, it is resolved into an IP address by a DNS server. The device
checks network connectivity and QoS of the resolved IP address. If the DNS host
name is unresolved, the device considers the connectivity of the IP destination as
unavailable.

You can view the connectivity status of IP destinations in the following Web interface
pages:
 Outbound IP Routing Table: The connectivity status of the IP destination per routing
rule is displayed in the 'Status' column. For more information, see 'Configuring Tel to
IP Routing' on page 256.
 IP Connectivity: This page displays a more informative connectivity status of the IP
destinations used in Tel-to-IP routing rules in the Outbound IP Routing table. For
viewing this page, see 'Viewing IP Connectivity' on page 421.
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23.5 Alternative Routing for Tel-to-IP Calls

The device supports various alternative Tel-to-IP call routing methods, as described in this
section.
23.5.1 Alternative Routing Based on IP Connectivity

You can configure the device to do alternative Tel-to-IP call routing based on IP
connectivity. When the connectivity state of an IP destination is unavailable, the device
attempts to re-route the Tel-to-IP call to an alternative IP destination. It does this by
searching for the next call matching rule (e.g., phone number prefix) in the Outbound IP
Routing table.
Notes:
• Alternative routing based on IP connectivity is applicable only when a proxy server
is not used.
• As the device searches the Outbound IP Routing table for a matching rule starting
from the top, you must configure the main routing rule above the alternative
routing rules.
• The maximum number of alternative routing rules that can be configured for each
routing rule in the table is three.
• For configuring Tel-to-IP routing rules in the Outbound IP Routing table, see
'Configuring Tel to IP Routing' on page 256.
The device searches for an alternative IP destination when any of the following connectivity
states are detected with the IP destination of the initial Tel-to-IP routing rule:
 No response received from a ping or from SIP OPTIONS messages. This depends on
the chosen method for checking IP connectivity.
 Poor QoS according to the configured thresholds for packet loss and delay.
 Unresolved DNS, if the configured IP destination is a domain name (or FQDN). If the
domain name is resolved into two IP addresses, the timeout for INVITE re-
transmissions can be configured using the HotSwapRtx parameter. For example, if
you set this parameter to 3, the device attempts up to three times to route the call to
the first IP address and if unsuccessful, it attempts up to three times to re-route it to
the second resolved IP address.
The connectivity status of the IP destination is displayed in the 'Status' column of the
Outbound IP Routing table per routing rule. If it displays a status other than "ok", then the
device considers the IP destination as unavailable and attempts to re-route the call to an
alternative destination. For more information on the IP connectivity methods and on
viewing IP connectivity status, see 'IP Destinations Connectivity Feature' on page 266.
The table below shows an example of alternative routing where the device uses an
available alternative routing rule in the Outbound IP Routing table to re-route the initial Tel-
to-IP call.
Table 23-4: Alternative Routing based on IP Connectivity Example
Destination IP Connectivity
IP Destination Rule Used?
Phone Prefix Status
Main Route 40 10.33.45.68 "No Connectivity" No

Alternative Route #1 40 10.33.45.70 "QoS Low" No

Destination IP Connectivity
IP Destination Rule Used?
Phone Prefix Status
Alternative Route #2 40 10.33.45.72 "ok" Yes

The steps for configuring alternative Tel-to-IP routing based on IP connectivity are
summarized below.
 To configure alternative Tel-to-IP routing based on IP connectivity:

1. In the Outbound IP Routing table, add alternative Tel-to-IP routing rules for specific
calls.
2. In the Routing General Parameters page (Configuration tab > VoIP menu > GW and
IP to IP > Routing > General Parameters), do the following:
a. Enable alternative routing based on IP connectivity, by setting the 'Enable Alt
Routing Tel to IP' (AltRoutingTel2IPEnable) parameter to Enable.
b. Configure the IP connectivity reason for triggering alternative routing, by setting
the 'Alt Routing Tel to IP Mode' parameter (AltRoutingTel2IPMode) to one of the
following:
♦ Ping or SIP OPTIONS failure
♦ Poor QoS
♦ Ping or SIP OPTIONS failure, poor QoS, or unresolved DNS
c. The device plays a tone to the Tel endpoint (for analog interfaces) whenever an
alternative route is used. This tone is played for a user-defined time configured by
the 'Alternative Routing Tone Duration' parameter.
23.5.2 Alternative Routing Based on SIP Responses

You can configure the device to do alternative routing based on the received SIP response.
If the SIP response code reflects an error (i.e., 4xx, 5xx, or 6xx) and you have configured
this specific response code as a trigger for alternative routing, then the device attempts to
re-route the call to an alternative destination.
You can configure up to five SIP response codes for triggering alternative routing. This is
done in the Reasons for Alternative Routing table, explained in this section.
Typically, the device performs alternative routing when there is no response at all to an
INVITE message after a user-defined number of INVITE re-transmissions, configured using
the SIPMaxRtx parameter. In such a scenario, the device issues itself the SIP response
code 408 "Request Timeout". If this release code is defined in the Reasons for Alternative
Routing table, then alternative routing is done.
Note: The device also plays a tone to the endpoint whenever an alternative route is
used. This tone is played for a user-defined time, configured by the
AltRoutingToneDuration parameter.
Depending on configuration, the alternative routing is done using one of the following
configuration entities:
 Outbound IP Routing Rules: You can configure up to two alternative routing rules in
the table. If the initial, main routing rule destination is unavailable, the device searches
the table (starting from the top) for the next call matching rule (e.g., destination phone
number), and if available attempts to re-route the call to the IP destination configured
for this alternative routing rule. The table below shows an example of alternative
routing where the device uses the first available alternative routing rule to re-route the
initial, unsuccessful Tel-to-IP call destination.
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Table 23-5: Alternative Routing based on SIP Response Code Example
Destination
IP Destination SIP Response Rule Used?
Phone Prefix
408 Request No
Main Route 40 10.33.45.68
Timeout
Alternative Route #1 40 10.33.45.70 486 Busy Here No
Alternative Route #2 40 10.33.45.72 200 OK Yes
 Proxy Sets: Proxy Sets are used for Server-type IP Groups (e.g., an IP PBX) and
define the actual IP destination (IP address or FQDN) of the server. As you can define
up to five IP destinations per Proxy Set, the device supports proxy redundancy, which
works together with the alternative routing feature. If the destination of a routing rule in
the Outbound IP Routing table is an IP Group, the device routes the call to the IP
destination configured for the Proxy Set associated with the IP Group. If the first IP
destination of the Proxy Set is unavailable, the device attempts to re-route the call to
the next proxy destination, and so on until an available IP destination is located. To
enable the Proxy Redundancy feature, set the IsProxyHotSwap parameter to 1 (per
Proxy Set) and set the EnableProxyKeepAlive to 1.
When the Proxy Redundancy feature is enabled, the device continually monitors the
connection with the proxies by using keep-alive messages (SIP OPTIONS). The
device sends these messages every user-defined interval (ProxyKeepAliveTime
parameter). Any response from the proxy, either success (200 OK) or failure (4xx
response) is considered as if the proxy is communicating. If there is no response from
the first (primary) proxy after a user-defined number of re-transmissions (re-INVITEs)
configured using the HotSwapRtx parameter, the device attempts to communicate
(using the same INVITE) with the next configured (redundant) proxy in the list, and so
on until an available redundant proxy is located. The device’s behavior can then be
one of the following, depending on the ProxyRedundancyMode parameter setting:
• The device continues operating with the redundant proxy (now active) until the
next failure occurs, after which it switches to the next redundant proxy. This is
referred to as Parking mode.
• The device always attempts to operate with the primary proxy. In other words, it
switches back to the primary proxy whenever it's available again. This is referred
to as Homing mode.
If none of the proxy servers respond, the device goes over the list again.
The steps for configuring alternative Tel-to-IP routing based on SIP response codes are
summarized below.
 To configure alternative Tel-to-IP routing based on SIP response codes:

1. Enable alternative routing based on SIP responses, by setting the 'Redundant Routing
Mode' parameter to one of the following:
• Routing Table for using the Outbound IP Routing table for alternative routing.
• Proxy for using the Proxy Set redundancy feature for alternative routing.
2. If you are using the Outbound IP Routing table, configure alternative routing rules with
identical call matching characteristics, but with different IP destinations. If you are
using the Proxy Set, configure redundant proxies.

3. Define SIP response codes (call failure reasons) that invoke alternative Tel-to-IP
routing:
a. Open the Reasons for Alternative Routing page (Configuration tab > VoIP menu
> GW and IP to IP submenu > Routing submenu > Alternative Routing
Reasons).
Figure 23-5: Tel to IP Reasons - Reasons for Alternative Routing Page
b. Under the 'Tel to IP Reasons' group, select up to five different SIP response
codes (call failure reasons) that invoke alternative Tel-to-IP routing.
c. Click Submit.
23.6 Alternative Routing for IP-to-Tel Calls

The device supports alternative IP-to-Tel call routing, as described in this section.
23.6.1 Alternative Routing to Trunk upon Q.931 Call Release Cause

Code
You can configure the device to do alternative IP-to-Tel call routing based on the received
ISDN Q.931 cause code. If an IP-to-Tel call is rejected or disconnected on the Tel side as a
result of a specific ISDN Q.931 release cause code that is listed in the Reasons for
Alternative Routing table, the device searches for an alternative IP-to-Tel routing rule in the
Inbound IP Routing table and sends it to the alternative Hunt Group. For example, you can
enable alternative IP-to-Tel routing for scenarios where the initial Tel destination is busy
and a Q.931 Cause Code No. 17 is received (or for other call releases that issue the
default Cause Code No. 3).
You can also configure a default release cause code that the device issues itself upon the
following scenarios:
 The device initiates a call release whose cause is unknown.
 No free channels (i.e., busy) in the Hunt Group.
 No appropriate routing rule located in the Inbound IP Routing table to the Hunt Group.
 Phone number is not found in the Inbound IP Routing table.
By default, it is set to Cause Code No. 3 (No Route to Destination). This default cause
code can be changed using the 'Default Release Cause' parameter located in the
Advanced Parameters page (Configuration tab > VoIP menu > SIP Definitions >
Advanced Parameters). To enable alternative routing based on Q.931 cause code, you
need to define this cause code in the Reasons for Alternative Routing table.
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 To configure alternative Hunt Group routing based on Q.931 cause codes:

1. In the Proxy & Registration page, set the 'Redundant Routing Mode' parameter to
Routing Table so that the device uses the Inbound IP Routing table for alternative
routing.
2. In the Inbound IP Routing table, configure alternative routing rules with the same call
matching characteristics, but with different Hunt Group destinations.
3. Configure up to five Q.931 cause codes that invoke alternative IP-to-Tel routing:
a. Open the Reasons for Alternative Routing page (Configuration tab > VoIP menu
> GW and IP to IP > Routing > Alternative Routing Reasons).
Figure 23-6: IP to Tel Reasons - Reasons for Alternative Routing Page
b. Under the 'IP to Tel Reasons' group, select the desired Q.931 cause codes.
c. Click Submit to apply your changes.
Notes:
• You can configure up to two alternative routing rules in the Inbound IP Routing
table.
• The default release cause is described in the Q.931 notation and is translated to
corresponding SIP 40x or 50x values (e.g., Cause Code No. 3 to SIP 404, and
Cause Code No. 34 to SIP 503).
• For information on mapping PSTN release causes to SIP responses, see PSTN
Release Cause to SIP Response Mapping.
• For configuring IP-to-Tel routing rules in the Inbound IP Routing table, see
'Configuring IP to Hunt Group Routing Table' on page 263.
• The Reasons for Alternative Routing IP to Tel table can also be configured using
the table ini file parameter, AltRouteCauseIP2Tel.
23.6.2 Alternative Routing to an IP Destination upon a Busy Trunk

You can configure the device to forward (i.e., call redirection) IP-to-Tel calls to an
alternative IP destination using SIP 3xx responses. This can be done upon the following
scenario:
 Unavailable FXS / FXO Hunt Group. This feature can be used, for example, to
forward the call to another FXS / FXO device.
This feature is configured per Hunt Group and is configured in the Forward on Busy Trunk
Destination table, as described in this section.
The alternative destination can be defined as a host name or as a SIP Request-URI user
name and host part (i.e., user@host). For example, the below configuration forwards IP-to-
Tel calls to destination user “112” at host IP address 10.13.4.12, port 5060, using transport
protocol TCP, if Trunk Group ID 2 is unavailable:
ForwardOnBusyTrunkDest 1 = 2, [email protected]:5060;transport=tcp;
When configured with user@host, the original destination number is replaced by the user
part.

The device forwards calls using this table only if no alternative IP-to-Tel routing rule has
been configured in the Inbound IP Routing table or alternative routing fails and the
following reason in the SIP Diversion header of 3xx messages exists:
 "unavailable": All FXS / FXO lines pertaining to a Hunt Group are busy or unavailable
Note: You can also configure the Forward on Busy Trunk Destination table using the
table ini file parameter, ForwardOnBusyTrunkDest.
 To configure Forward on Busy Trunk Destination rules:

1. Open the Forward on Busy Trunk Destination page (Configuration tab > VoIP menu
> GW and IP to IP > Routing > Forward on Busy Trunk).
Figure 23-7: Forward on Busy Trunk Destination Page
The figure above displays a configuration that forwards IP-to-Tel calls destined for
Hunt Group ID 1 to destination IP address 10.13.5.67 if the conditions mentioned
earlier exist.
2. Configure the table as required, and then click Submit to apply your changes.
3. Save the changes to the device's flash memory with a device reset (see 'Saving
Table 23-6: Forward on Busy Trunk Destination Description Parameters
Trunk Group ID Defines the Trunk Group ID to which the IP call is destined
[ForwardOnBusyTrunkDest_Trunk to.
GroupId]
Forward Destination Defines the alternative IP destination for the call used if the
[ForwardOnBusyTrunkDest_Forwar Trunk Group is busy or unavailable.
dDestination] The valid value can be an IP address in dotted-decimal
notation, an FQDN, or a SIP Request-URI user name and
host part (i.e., user@host). The following syntax can also be
used: host:port;transport=xxx (i.e., IP address, port and
transport type).
Note: When configured with a user@host, the original
destination number is replaced by the user part.
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User's Manual 24. Configuring DTMF and Dialing
24 Configuring DTMF and Dialing

The DTMF & Dialing page is used to configure parameters associated with dual-tone multi-
frequency (DTMF) and dialing. For a description of the parameters appearing on this page,
see 'Configuration Parameters Reference' on page 475.
 To configure the DTMF and dialing parameters:

1. Open the DTMF & Dialing page (Configuration tab > VoIP menu > GW and IP to IP
submenu > DTMF & Supplementary submenu > DTMF & Dialing).
Figure 24-1: DTMF & Dialing

24.1 Dialing Plan Features

This section describes various dialing plan features supported by the device.
24.1.1 Digit Mapping

The device collects digits until a match is found in the user-defined digit pattern (e.g., for
closed numbering schemes). The device stops collecting digits and starts sending the
digits (collected number) when any one of the following scenarios occur:
 Maximum number of digits is received. You can define (using the MaxDigits
parameter) the maximum number of collected destination number digits that can be
received (i.e., dialed) from the Tel side by the device. When the number of collected
digits reaches the maximum (or a digit map pattern is matched), the device uses these
digits for the called destination number.
 Inter-digit timeout expires (e.g., for open numbering schemes). This is defined using
the TimeBetweenDigits parameter. This is the time that the device waits between each
received digit. When this inter-digit timeout expires, the device uses the collected
digits to dial the called destination number.
 The phone's pound (#) key is pressed.
 Digit string (i.e., dialed number) matches one of the patterns defined in the digit map.
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Digit map (pattern) rules are defined using the DigitMapping parameter. The digit map
pattern can contain up to 52 options (rules), each separated by a vertical bar ("|"). The
maximum length of the entire digit pattern is 152 characters. The available notations are
described in the table below:
Table 24-1: Digit Map Pattern Notations
Notation Description
[n-m] Range of numbers (not letters).

. (single dot) Repeat digits until next notation (e.g., T).
x Any single digit.
Note: This notation does not apply in some scenarios when using the star (*)
or hash (#) key. For example, the key sequence of ** must be presented in the
dial plan as *x.s (instead of xx).
T Dial timeout (configured by the TimeBetweenDigits parameter).
S Short timer (configured by the TimeBetweenDigits parameter; default is two
seconds) that can be used when a specific rule is defined after a more general
rule. For example, if the digit map is 99|998, then the digit collection is
terminated after the first two 9 digits are received. Therefore, the second rule
of 998 can never be matched. But when the digit map is 99s|998, then after
dialing the first two 9 digits, the device waits another two seconds within which
the caller can enter the digit 8.
Below is an example of a digit map pattern containing eight rules:

DigitMapping = 11xS|00[1-
7]xxx|8xxxxxxx|#xxxxxxx|*xx|91xxxxxxxxxx|9011x|xx.T
In the example, the rule "00[1-7]xxx" denotes dialed numbers that begin with 00, and then
any digit from 1 through 7, followed by three digits (of any number). Once the device
receives these digits, it does not wait for additional digits, but starts sending the collected
digits (dialed number) immediately.
Notes:
• If you want the device to accept/dial any number, ensure that the digit map
contains the rule "xx.T"; otherwise, dialed numbers not defined in the digit map are
rejected.
• If you are using an external Dial Plan file for dialing plans (see 'Dialing Plans for
Digit Collection' on page 376), the device first attempts to locate a matching digit
pattern in the Dial Plan file, and if not found, then attempts to locate a matching
digit pattern in the Digit Map (configured by the DigitMapping parameter).
• It may be useful to configure both Dial Plan file and Digit Maps. For example, the
Digit Map can be used for complex digit patterns (which are not supported by the
Dial Plan) and the Dial Plan can be used for long lists of relatively simple digit
patterns. In addition, as timeout between digits is not supported by the Dial Plan,
the Digit Map can be used to define digit patterns (MaxDigits parameter) that are
shorter than those defined in the Dial Plan, or left at default. For example, “xx.T”
Digit Map instructs the device to use the Dial Plan and if no matching digit pattern,
it waits for two more digits and then after a timeout (TimeBetweenDigits
parameter), it sends the collected digits. Therefore, this ensures that calls are not
rejected as a result of their digit pattern not been completed in the Dial Plan.

User's Manual 24. Configuring DTMF and Dialing
24.1.2 External Dial Plan File

The device can be loaded with a Dial Plan file with user-defined dialing plans. For more
information, see 'Dial Plan File' on page 376.
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User's Manual 25. Configuring Supplementary Services
25 Configuring Supplementary Services

This section describes SIP supplementary services that can enhance your telephone
service.
Notes:
• All call participants must support the specific supplementary service that is used.
• When working with certain application servers (such as BroadSoft’s BroadWorks)
in client server mode (the application server controls all supplementary services
and keypad features by itself), the device's supplementary services must be
disabled.
The Supplementary Services page is used to configure many of the discussed

supplementary services parameters. For a description of the parameters appearing on this
page, see 'Configuration Parameters Reference' on page 475.
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 To configure supplementary services parameters:

1. Open the Supplementary Services page (Configuration tab > VoIP menu > GW and
IP to IP submenu > DTMF & Supplementary submenu > Supplementary Services).
Figure 25-1: Supplementary Services

3. Click Submit to apply your changes, or click the Subscribe to MWI or Unsubscribe
to MWI buttons to save your changes and to subscribe / unsubscribe to the MWI
server.

25.1 Call Hold and Retrieve

Initiating Call Hold and Retrieve:
 Active calls can be put on-hold by pressing the phone's hook-flash button.
 The party that initiates the hold is called the holding party; the other party is called the
held party.
 After a successful Hold, the holding party hears a dial tone (HELD_TONE defined in
the device's Call Progress Tones file).
 Call retrieve can be performed only by the holding party while the call is held and
active.
 The holding party performs the retrieve by pressing the telephone's hook-flash button.
 After a successful retrieve, the voice is connected again.
 Hold is performed by sending a Re-INVITE message with IP address 0.0.0.0 or
a=sendonly in the SDP according to the parameter HoldFormat.
Receiving Hold/Retrieve:
 When an active call receives a re-INVITE message with either the IP address 0.0.0.0
or the ‘inactive’ string in SDP, the device stops sending RTP and plays a local held
tone.
 When an active call receives a re-INVITE message with the ‘sendonly’ string in SDP,
the device stops sending RTP and listens to the remote party. In this mode, it is
expected that on-hold music (or any other hold tone) is played (over IP) by the remote
party.
You can also configure the device to keep a call on-hold for a user-defined time after which
the call is disconnected, using the HeldTimeout parameter.
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The device also supports "double call hold" for FXS interfaces where the called party,
which has been placed on-hold by the calling party, can then place the calling party on hold
as well and make a call to another destination. The flowchart below provides an example of
this type of call hold:
Figure 25-2: Double Hold SIP Call Flow
The flowchart above describes the following "double" call-hold scenario:

1. A calls B and establishes a voice path.
2. A places B on hold; A hears a dial tone and B hears a held tone.
3. A calls C and establishes a voice path.
4. B places A on hold; B hears a dial tone.
5. B calls D and establishes a voice path.
6. A ends call with C; A hears a held tone.
7. B ends call with D.
8. B retrieves call with A.

Notes:
• If a party that is placed on hold (e.g., B in the above example) is called by another
party (e.g., D), then the on-hold party receives a call waiting tone instead of the
held tone.
• While in a Double Hold state, placing the phone on-hook disconnects both calls
(i.e. call transfer is not performed).
• You can enable the device to handle incoming re-INVITE messages with
"a=sendonly" in the SDP, in the same way as if "a=inactive" is received in the
SDP. This is configured using the SIPHoldBehavior parameter. When enabled, the
device plays a held tone to the Tel phone and responds with a 200 OK containing
"a=recvonly" in the SDP.
25.2 Call Pickup

The device supports the Call Pick-Up feature, whereby the FXS user can answer someone
else's telephone call by pressing a user-defined sequence of phone keys. When the user
dials the user-defined digits (e.g., #77), the incoming call from the other phone is forwarded
to the FXS user's phone. This feature is configured using the parameter KeyCallPickup.
Note: The Call Pick-Up feature is supported only for FXS endpoints pertaining to the
same Hunt Group ID.
25.3 Consultation Feature

The device's Consultation feature allows you to place one number on hold and make a
second call to another party.
 After holding a call (by pressing hook-flash), the holding party hears a dial tone and
can then initiate a new call, which is called a Consultation call.
 While hearing a dial tone, or when dialing to the new destination (before dialing is
complete), the user can retrieve the held call by pressing hook-flash.
 The held call can’t be retrieved while ringback tone is heard.
 After the Consultation call is connected, the user can toggle between the held and
active call by pressing the hook-flash key.
Note: The Consultation feature is applicable only to FXS interfaces.
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25.4 Call Transfer

This section describes the device's support for call transfer types.
25.4.1 Consultation Call Transfer

The device supports Consultation Call Transfer using the SIP REFER message and
Replaces header. The common method to perform a consultation transfer is described in
the following example, which assumes three call parties:
 Party A = transferring
 Party B = transferred
 Party C = transferred to
1. A Calls B.
2. B answers.
3. A presses the hook-flash button and places B on-hold (party B hears a hold tone).
4. A dials C.
5. After A completes dialing C, A can perform the transfer by on-hooking the A phone.
6. After the transfer is complete, B and C parties are engaged in a call.
The transfer can be initiated at any of the following stages of the call between A and C:
 Just after completing dialing C phone number - transfer from setup
 While hearing ringback – transfer from alert
 While speaking to C - transfer from active
25.4.2 Blind Call Transfer

Blind call transfer is done (using SIP REFER messages) after a call is established between
call parties A and B, and party A decides to immediately transfer the call to C without first
speaking to C. The result of the transfer is a call between B and C (similar to consultation
transfer, but skipping the consultation stage).
25.5 Call Forward

The following methods of call forwarding are supported:
 Immediate: incoming call is forwarded immediately and unconditionally.
 Busy: incoming call is forwarded if the endpoint is busy.
 No Reply: incoming call is forwarded if it isn't answered for a specified time.
 On Busy or No Reply: incoming call is forwarded if the port is busy or when calls are
not answered after a specified time.
 Do Not Disturb: immediately reject incoming calls. Upon receiving a call for a Do Not
Disturb, the 603 Decline SIP response code is sent.
Three forms of forwarding parties are available:
 Served party: party configured to forward the call (FXS device).
 Originating party: party that initiates the first call (FXS or FXO device).
 Diverted party: new destination of the forwarded call (FXS or FXO device).

The served party (FXS interface) can be configured through the Web interface (see
Configuring Call Forward on page 309) or ini file to activate one of the call forward modes.
These modes are configurable per endpoint.
Notes:
• When call forward is initiated, the device sends a SIP 302 response with a contact
that contains the phone number from the forward table and its corresponding IP
address from the routing table (or when a proxy is used, the proxy’s IP address).
• For receiving call forward, the device handles SIP 3xx responses for redirecting
calls with a new contact.
25.5.1 Call Forward Reminder Ring

The device supports the Call Forward Reminder Ring feature for FXS interfaces, whereby
the device's FXS endpoint emits a short ring burst, only in onhook state, when a third-party
Application Server (e.g., softswitch) forwards an incoming call to another destination. This
is important in that it notifies (audibly) the FXS endpoint user that a call forwarding service
is currently being performed.
Figure 25-3: Call Forward Reminder with Application Server
The device generates a Call Forward Reminder ring burst to the FXS endpoint each time it
receives a SIP NOTIFY message with a “reminder ring” xml body. The NOTIFY request is
sent from the Application Server to the device each time the Application Server forwards an
incoming call. The service is cancelled when an UNSUBSCRIBE request is sent from the
device, or when the Subscription time expires.
The reminder-ring tone can be defined by using the parameter CallForwardRingToneID,
which points to a ring tone defined in the Call Progress Tone file.
The following parameters are used to configure this feature:
 EnableNRTSubscription
 ASSubscribeIPGroupID
 NRTSubscribeRetryTime
 CallForwardRingToneID
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25.5.2 Call Forward Reminder (Off-Hook) Special Dial Tone

The device plays a special dial tone (stutter dial tone - Tone Type #15) to a specific FXS
endpoint when the phone is off-hooked and when a third-party Application server (AS),
e.g., a softswitch is used to forward calls intended for the endpoint, to another destination.
This is useful in that it reminds the FXS user of this service. This feature does not involve
device subscription (SIP SUBSCRIBE) to the AS.
Activation/deactivation of the service is notified by the server. An unsolicited SIP NOTIFY
request is sent from the AS to the device when the Call Forward service is activated or de-
activated. Depending on this NOTIFY request, the device plays either the standard dial
tone or the special dial tone for Call Forward.
For playing the special dial tone, the received SIP NOTIFY message must contain the
following headers:
 From and To: contain the same information, indicating the specific endpoint
 Event: ua-profile
 Content-Type: "application/simservs+xml"
 Message body is the XML body and contains the “dial-tone-pattern” set to "special-
condition-tone" (<ss:dial-tone-pattern>special-condition-tone</ss:dial-tone-pattern>),
which is the special tone indication.
To cancel the special dial tone and playing the regular dial tone, the received SIP NOTIFY
message must contain the following headers:
 From and To: contain the same information, indicating the specific endpoint
 Event: ua-profile
 Content-Type: "application/simservs+xml"
 Message body is the XML body containing the “dial-tone-pattern” set to "standard-
condition-tone" (<ss:dial-tone-pattern>standard-condition-tone</ss:dial-tone-pattern>),
which is the regular dial tone indication.
Therefore, the special dial tone is valid until another SIP NOTIFY is received that instructs
otherwise (as described above).
Note: if the MWI service is active, the MWI dial tone overrides this special Call
Forward dial tone.
25.5.3 Call Forward Reminder Dial Tone (Off-Hook) upon Spanish SIP
Alert-Info
The device plays a special dial tone to FXS phones in off-hook state that are activated with
the call forwarding service. The special dial tone is used as a result of the device receiving
a SIP NOTIFY message from a third-party softswitch providing the call forwarding service
with the following SIP Alert-Info header:
Alert-Info: <http://127.0.0.1/Tono-Espec-Invitacion>;lpi-
aviso=Desvio-Inmediato
This special tone is a stutter dial tone (Tone Type = 15), as defined in the CPT file.
The FXS phone user, connected to the device, activates the call forwarding service by
dialing a special number (e.g., *21*xxxxx) and as a result, the device sends a regular SIP
INVITE message to the softswitch. The softswitch later notifies of the activation of the
forwarding service by sending an unsolicited NOTIFY message with the Alert-Info header,
as mentioned above.

When the call forwarding service is de-activated, for example, by dialing #21# and sending
an INVITE with this number, the softswitch sends another SIP NOTIFY message with the
following Alert-Info header:
Alert-Info: <http://127.0.0.1/ Tono-Normal-Invitacion>; Aviso =
Desvi‫ף‬-Inmediato
From this point on, the device plays a normal dial tone to the FXS phone when it goes off-
hook.
25.6 Call Waiting

The Call Waiting feature enables FXS devices to accept an additional (second) call on
busy endpoints. If an incoming IP call is designated to a busy port, the called party hears a
call waiting tone (several configurable short beeps) and (for Bellcore and ETSI Caller IDs)
can view the Caller ID string of the incoming call. The calling party hears a call waiting
ringback tone. The called party can accept the new call using hook-flash, and can toggle
between the two calls.
 To enable call waiting:

1. Set the parameter EnableCallWaiting to 1.
2. Set the parameter EnableHold to 1.
3. Define the Call Waiting indication and call waiting ringback tones in the Call Progress
Tones file. You can define up to four call waiting indication tones (refer to the
FirstCallWaitingToneID parameter).
4. To configure the call waiting indication tone cadence, modify the following parameters:
NumberOfWaitingIndications, WaitingBeepDuration and
TimeBetweenWaitingIndications.
5. To configure a delay interval before a Call Waiting Indication is played to the currently
busy port, use the parameter TimeBeforeWaitingIndication. This enables the caller to
hang up before disturbing the called party with Call Waiting Indications. Applicable
only to FXS modules.
Both the calling and called sides are supported by FXS interfaces; FXO interfaces support
only the calling side.
To indicate Call Waiting, the device sends a 182 Call Queued response. The device
identifies Call Waiting when a 182 Call Queued response is received.
25.7 Message Waiting Indication

The device supports Message Waiting Indication (MWI) according to IETF RFC 3842,
including SUBSCRIBE to an MWI server.
The FXS device can accept an MWI NOTIFY message that indicates waiting messages or
that the MWI is cleared. Users are informed of these messages by a stutter dial tone. The
stutter and confirmation tones are defined in the CPT file. If the MWI display is configured,
the number of waiting messages is also displayed. If the MWI lamp is configured, the
phone’s lamp (on a phone that is equipped with an MWI lamp) is lit. The device can
subscribe to the MWI server per port (usually used on FXS) or per device (used on FXO).
You can also configure the voltage level mode (low or high) that the FXS port generates to
the connected phone for lighting the phone's lamp used for indicating a message in waiting.
If you require a flashing lamp, you can configure the on-off lamp durations. To configure
this feature, see the following parameters:
 EnableLowVoltageMwiGeneration
 LedMwiOnDurationTime
 LedMwiOffDurationTime
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 NeonMwiOnDurationTime
 NeonMwiOffDurationTime
Note: For more information on IP voice mail configuration, refer to the IP Voice Mail
CPE Configuration Guide.
To configure MWI, use the following parameters:

 EnableMWI
 MWIServerIP, or MWISubscribeIPGroupID and ProxySet
 MWIAnalogLamp
 MWIDisplay
 StutterToneDuration
 EnableMWISubscription
 MWIExpirationTime
 SubscribeRetryTime
 SubscriptionMode
 CallerIDType (determines the standard for detection of MWI signals)
 ETSIVMWITypeOneStandard
 BellcoreVMWITypeOneStandard
 VoiceMailInterface
 EnableVMURI
25.8 Caller ID
This section describes the device's Caller ID support.
25.8.1 Caller ID Detection / Generation on the Tel Side

By default, generation and detection of Caller ID to the Tel side is disabled. To enable
Caller ID, set the parameter EnableCallerID to 1. When the Caller ID service is enabled:
 For FXS: the Caller ID signal is sent to the device's port
 For FXO: the Caller ID signal is detected
The configuration for Caller ID is described below:
 Use the parameter CallerIDType to define the Caller ID standard. Note that the Caller
ID standard that is used on the PBX or phone must match the standard defined in the
device.
 Select the Bellcore caller ID sub standard using the parameter
BellcoreCallerIDTypeOneSubStandard
 Select the ETSI FSK caller ID sub standard using the parameter
ETSICallerIDTypeOneSubStandard
 Enable or disable (per port) the caller ID generation (for FXS) and detection (for FXO)
using the ‘Generate / Detect Caller ID to Tel’ table (EnableCallerID). If a port isn’t
configured, its caller ID generation / detection are determined according to the global
parameter EnableCallerID.
 EnableCallerIDTypeTwo: disables / enables the generation of Caller ID type 2 when
the phone is off-hooked (used for call waiting).

 RingsBeforeCallerID: sets the number of rings before the device starts detection of
caller ID (FXO only). By default, the device detects the caller ID signal between the
first and second rings.
 AnalogCallerIDTimimgMode: determines the time period when a caller ID signal is
generated (FXS only). By default, the caller ID is generated between the first two
rings.
 PolarityReversalType: some Caller ID signals use reversal polarity and/or wink
signals. In these scenarios, it is recommended to set PolarityReversalType to 1 (Hard)
(FXS only).
 The Caller ID interworking can be changed using the parameters
UseSourceNumberAsDisplayName and UseDisplayNameAsSourceNumber.
25.8.2 Debugging a Caller ID Detection on FXO

The procedure below describes debugging caller ID detection in FXO interfaces.
 To debug a Caller ID detection on an FXO interface:

1. Verify that the parameter EnableCallerID is set to 1.
2. Verify that the caller ID standard (and substandard) of the device matches the
standard of the PBX (using the parameters CallerIDType,
BellcoreCallerIDTypeOneSubStandard, and ETSICallerIDTypeOneSubStandard).
3. Define the number of rings before the device starts the detection of caller ID (using the
parameter RingsBeforeCallerID).
4. Verify that the correct FXO coefficient type is selected (using the parameter
CountryCoefficients), as the device is unable to recognize caller ID signals that are
distorted.
5. Connect a phone to the analog line of the PBX (instead of to the device's FXO
interface) and verify that it displays the caller ID.
If the above does not solve the problem, you need to record the caller ID signal (and send
it to AudioCodes), as described below.
 To record the caller ID signal using the debug recording mechanism:

1. Access the FAE page (by appending "FAE" to the device's IP address in the Web
browser's URL, for example, http://10.13.4.13/FAE).
2. Press the Cmd Shell link.
3. Enter the following commands:
dr
ait <IP address of PC to collect the debug traces sent from
the device>
AddChannelIdTrace ALL-WITH-PCM <port number, which starts from
0>
Start
4. Make a call to the FXO.
5. To stop the DR recording, at the CLI prompt, type STOP.
25.8.3 Caller ID on the IP Side

Caller ID is provided by the SIP From header containing the caller's name and "number",
for example:
From: “John” <SIP:[email protected]>;tag=35dfsgasd45dg
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If Caller ID is restricted (received from Tel or configured in the device), the From header is
set to:
From: “anonymous” <[email protected]>; tag=35dfsgasd45dg
The P-Asserted (or P-Preferred) headers are used to present the originating party’s caller
ID even when the caller ID is restricted. These headers are used together with the Privacy
header.
 If Caller ID is restricted:
• The From header is set to “anonymous” <[email protected]>
• The ‘Privacy: id’ header is included
• The P-Asserted-Identity (or P-Preferred-Identity) header shows the caller ID
 If Caller ID is allowed:
• The From header shows the caller ID
• The ‘Privacy: none’ header is included
• The P-Asserted-Identity (or P-Preferred-Identity) header shows the caller ID
The caller ID (and presentation) can also be displayed in the Calling Remote-Party-ID
header.
The ‘Caller Display Information’ table (CallerDisplayInfo) is used for the following:
 FXS interfaces - to define the caller ID (per port) that is sent to IP.
 FXO interfaces - to define the caller ID (per port) that is sent to IP if caller ID isn’t
detected on the Tel side, or when EnableCallerID = 0.
 FXS and FXO interfaces - to determine the presentation of the caller ID (allowed or
restricted).
 To maintain backward compatibility - when the strings ‘Private’ or ‘Anonymous’ are
set in the Caller ID/Name field, the caller ID is restricted and the value in the
Presentation field is ignored.
The value of the ‘Presentation’ field that is defined in the ‘Caller Display Information’ table
can be overridden by configuring the ‘Presentation’ parameter in the ‘Tel to IP Source
Number Manipulation’ table. Therefore, this table can be used to set the presentation for
specific calls according to Source / Destination prefixes.
The caller ID can be restricted/allowed (per port) using keypad features KeyCLIR and
KeyCLIRDeact (FXS only).
AssertedIdMode defines the header that is used (in the generated INVITE request) to
deliver the caller ID (P-Asserted-Identity or P-Preferred-Identity). Use the parameter
UseTelURIForAssertedID to determine the format of the URI in these headers (sip: or tel:).
The parameter EnableRPIheader enables Remote-Party-ID (RPI) headers for calling and
called numbers for Tel-to-IP calls.
25.9 Three-Way Conferencing

The device supports three-way conference calls. These conference calls can also occur
simultaneously. The device supports the following conference modes (configured by the
parameter 3WayConferenceMode):
 Conferencing managed by an external, AudioCodes Conference (media) server:
The Conference-initiating INVITE sent by the device uses the ConferenceID
concatenated with a unique identifier as the Request-URI. This same Request-URI is
set as the Refer-To header value in the REFER messages that are sent to the two
remote parties. For this mode, the 3WayConferenceMode parameter is set to 0
(default.)
 Conferencing managed by an external, third-party Conference (media) server:
The Conference-initiating INVITE sent by the device uses only the ConferenceID as
the Request-URI. The Conference server sets the Contact header of the 200 OK
response to the actual unique identifier (Conference URI) to be used by the

participants. This Conference URI is included (by the device) in the Refer-To header
value in the REFER messages sent by the device to the remote parties. The remote
parties join the conference by sending INVITE messages to the Conference server
using this conference URI. For this mode, the 3WayConferenceMode parameter is set
to 1.
 Local, on-board conferencing: The conference is established on the device without
the need for an external Conference server. This feature includes local mixing and
transcoding of the 3-Way Call legs on the device, and even allowing multi-codec
conference calls. The number of simultaneous, on-board conferences can be limited
using the parameter MaxInBoardConferenceCalls. The device utilizes resources from
idle ports to establish the conference call. You can designate ports that can’t be used
as a resource for conference calls initiated by other ports, using the parameter
3WayConfNoneAllocateablePorts. Ports that are not configured with this parameter
(and that are idle) are used by the device as a resource for establishing these types of
conference calls. The device supports up to two simultaneous, on-board, three-way
conference calls. For this mode, the 3WayConferenceMode parameter is set to 2.
Notes:
• Each three-way conference call requires the resources of two DSP channels.
Consequently, for MP-114, MP-118 and MP-124, each three-way conference call
reduces channel capacity by one; for MP-112, no channel reduction occurs.
• Instead of using the flash-hook button to establish a three-way conference call,
you can dial a user-defined hook-flash code (e.g., "*1"), configured by the
HookFlashCode parameter.
• Three-way conferencing is applicable only to FXS interfaces.
The following example demonstrates three-way conferencing using the device's local, on-
board conferencing feature. In this example, telephone "A" connected to the device
establishes a three-way conference call with two remote IP phones, "B" and "C":
1. A establishes a regular call with B.
2. A places B on hold, by pressing the telephone's flash-hook button and the number "1"
key.
3. A hears a dial tone and then makes a call to C.
4. C answers the call.
5. A establishes a three-way conference call with B and C, by pressing the flash-hook
button and digit 3.
To configure this local, on-board three-way conferencing:
1. Open the Supplementary Services page.
2. Set 'Enable 3-Way Conference' to Enable (Enable3WayConference = 1).
3. Set 'Three Way Conference Mode' to On Board (3WayConferenceMode = 2).
4. Set 'Flash Keys Sequence Style' to Sequence 1 or Sequence 2
(FlashKeysSequenceStyle = 1 or 2).
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Note: For local, on-board three-way conferencing on MP-112, in addition to

configuring the previously mentioned parameters, the following must be configured:
EnableIPMediaChannels = 1
[ IPMediaChannels ]
FORMAT IPMediaChannels_Index = IPMediaChannels_ModuleID,
IPMediaChannels_DSPChannelsReserved;
IPMediaChannels 0 = 1, 2;
[ \IPMediaChannels ]

25.10 Emergency E911 Phone Number Services

This section describes the device's support for emergency phone number services.
25.10.1 Pre-empting Existing Calls for E911 IP-to-Tel Calls

If the device receives an E911 call from the IP network destined to the Tel, and there are
unavailable channels (e.g., all busy), the device terminates one of the calls (arbitrary) and
then sends the E911 call to that channel. The preemption is done only on a channel
pertaining to the same Hunt Group for which the E911 call was initially destined and if the
channel select mode (configured by the ChannelSelectMode parameter) is set to a value
other than “By Dest Number” (0).
The preemption is done only if the incoming IP-to-Tel call is identified as an emergency
call. The device identifies emergency calls by one of the following:
 The destination number of the IP call matches one of the numbers defined by the
EmergencyNumbers parameter. For E911, you must defined this parameter with the
value "911".
 The Priority header of the incoming SIP INVITE message contains the “emergency”
value.
Emergency pre-emption of calls can be enabled for all calls, using the global parameter
CallPriorityMode, or for specific calls using the Tel Profile parameter CallPriorityMode.
Notes:
• For Hunt Groups configured with call preemption, all must be configured to MLPP
[1] or all configured to Emergency [2]. In other words, you cannot set some trunks
to [1] and some to [2].
• The global parameter must be set to the same value as that of the Tel Profile
parameter; otherwise, the Tel Profile parameter is not applied.
• If you configure call preemption using the global parameter and a new Tel Profile
is subsequently added, the TelProfile_CallPriorityMode parameter automatically
acquires the same setting as well.
• This feature is applicable to FXO interfaces.
• For FXO interfaces, the preemption is done only on existing IP-to-Tel calls. In
other words, if all the current FXO channels are busy with calls that were
answered by the FXO device (i.e., Tel-to-IP calls), new incoming emergency IP-to-
Tel calls are rejected.
25.11 Multilevel Precedence and Preemption

The device supports Multilevel Precedence and Preemption (MLPP) service. MLPP is a
call priority scheme, which does the following:
 Assigns a precedence level (priority level) to specific phone calls or messages.
 Allows higher priority calls (precedence call) and messages to preempt lower priority
calls and messages (i.e., terminates existing lower priority calls) that are recognized
within a user-defined domain (MLPP domain ID). The domain specifies the collection
of devices and resources that are associated 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, MLPP service can
preempt the existing call that the called MLPP subscriber is on for a higher-
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precedence call. MLPP service availability does not apply across different domains.
MLPP is typically used in the military where, for example, high-ranking personnel can
preempt active calls during network stress scenarios such as a national emergency or
degraded network situations.
MLPP can be enabled for all calls, using the global parameter, CallPriorityMode, or for
specific calls using the Tel Profile parameter, CallPriorityMode.
Notes:
• For Hunt Groups configured with call preemption, all must be configured to MLPP
[1] or all configured to Emergency [2]. In other words, you cannot set some trunks
to [1] and some to [2].
• The global parameter must be set to the same value as that of the Tel Profile
parameter; otherwise, the Tel Profile parameter is not applied.
• If you configure call preemption using the global parameter and a new Tel Profile
is subsequently added, the TelProfile_CallPriorityMode parameter automatically
acquires the same setting as well.
The Resource Priority value in the Resource-Priority SIP header can be any one of those
listed in the table below. For each MLPP call priority level, the Multiple Differentiated
Services Code Points (DSCP) can be set to a value from 0 to 63.
Table 25-1: MLPP Call Priority Levels (Precedence) and DSCP Configuration Parameters
MLPP Precedence Level Precedence Level in Resource- DSCP Configuration Parameter

Priority SIP Header
0 (lowest) routine MLPPRoutineRTPDSCP

2 priority MLPPPriorityRTPDSCP
4 immediate MLPPImmediateRTPDSCP
6 flash MLPPFlashRTPDSCP
8 flash-override MLPPFlashOverRTPDSCP
9 (highest) flash-override-override MLPPFlashOverOverRTPDSCP
Notes:
• If required, you can exclude the "resource-priority” tag from the SIP Require
header in INVITE messages for Tel-to-IP calls when MLPP priority call handling is
used. This is configured using the RPRequired parameter.
• For a complete list of the MLPP parameters, see 'MLPP and Emergency Call
Parameters' on page 599.
25.11.1 MLPP Preemption Events in SIP Reason Header

The device sends the SIP Reason header (as defined in RFC 4411) to indicate the reason
and type of a preemption event. The device sends a SIP BYE or CANCEL request, or SIP
480, 486, 488 response (as appropriate) with a Reason header whose Reason-params can
includes one of the following preemption cause classes:
 Reason: preemption ;cause=1 ;text=”UA Preemption”
 Reason: preemption ;cause=2 ;text=”Reserved Resources Preempted”

 Reason: preemption ;cause=3 ;text=”Generic Preemption”

 Reason: preemption ;cause=4 ;text=”Non-IP Preemption”
This Reason cause code indicates that the session preemption has occurred in a non-
IP portion of the infrastructure. The device sends this code in the following scenarios:
• The device performs a network preemption of a busy call (when a high priority
call is received), the device sends a SIP BYE or CANCEL request with this
Reason cause code.
• The device performs a preemption of a B-channel for a Tel-to-IP outbound call
request from the softswitch for which it has not received an answer response
(e.g., Connect), and the following sequence of events occurs:
a. The device sends a Q.931 DISCONNECT over the ISDN MLPP PRI to the
partner switch to preempt the remote end instrument.
b. The device sends a 488 (Not Acceptable Here) response with this Reason
cause code.
 Reason: preemption; cause=5; text=”Network Preemption”
This Reason cause code indicates preempted events in the network. Within the
Defense Switched Network (DSN) network, the following SIP request messages and
response codes for specific call scenarios have been identified for signaling this
preemption cause:
• SIP:BYE - If an active call is being preempted by another call
• CANCEL - If an outgoing call is being preempted by another call
• 480 (Temporarily Unavailable), 486 (User Busy), 488 (Not Acceptable Here) -
Due to incoming calls being preempted by another call.
The device receives SIP requests with preemption reason cause=5 in the following
cases:
• The softswitch performs a network preemption of an active call - the following
sequence of events occurs:
a. The softswitch sends the device a SIP BYE request with this Reason cause
code.
b. The device initiates the release procedures for the B-channel associated
with the call request and maps the preemption cause to PRI Cause = #8
‘Preemption’. This value indicates that the call is being preempted. For PRI,
it also indicates that the B-channel is not reserved for reuse.
c. The device sends a SIP 200 OK in response to the received BYE, before the
SIP end instrument can proceed with the higher precedence call.
• The softswitch performs a network preemption of an outbound call request for the
device that has not received a SIP 2xx response - the following sequence of
events occur:
a. The softswitch sends the device a SIP 488 (Not Acceptable Here) response
code with this Reason cause code. The device initiates the release
procedures for the B-channel associated with the call request and maps the
preemption cause to PRI Cause = #8 ‘Preemption’.
b. The device deactivates any user signaling (e.g., ringback tone) and when
the call is terminated, it sends a SIP ACK message to the softswitch.
25.11.2 Precedence Ring Tone

You can assign a ring tone that is defined in the CPT file to be played when a precedence
call is received from the IP side. This is configured by the PrecedenceRingingType
parameter.
Emergency Telecommunications Services (ETS) calls (e.g., E911) can be configured with
a higher priority than any MLPP call (default), using the E911MLPPBehavior parameter.
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25.12 Denial of Collect Calls

You can configure the device to reject (disconnect) incoming Tel-to-IP collect calls and to
signal this denial to the PSTN. This capability is required, for example, in the Brazilian
telecommunication system to deny collect calls. When this feature is enabled upon
rejecting the incoming call, the device sends a sequence of signals to the PSTN. This
consists of an off-hook, an on-hook after one second, and then an off-hook after two
seconds. In other words, this is in effect, a double-answer sequence.
This feature can be enabled for all calls, using the EnableFXODoubleAnswer "global"
parameter, or it can be enabled for specific calls, by enabling this feature in a Tel Profile.
Notes:
• This feature is applicable only to FXO interfaces.
• If automatic dialing is also configured for an FXO port enabled with Denial of
Collect Calls, the FXO line does not answer the incoming call (ringing) until a SIP
200 OK is received from the remote destination. When a 200 OK is received, a
double answer is sent from the FXO line.
• Ensure that the PSTN side is configured to identify this double-answer signal.

25.13 Configuring Voice Mail

The Voice Mail Settings page allows you to configure the voice mail parameters. For a
description of these parameters, see 'Configuration Parameters Reference' on page 475.
Notes:
• The Voice Mail Settings page is available only for FXO interfaces.
• For more information on configuring voice mail, refer to the CPE Configuration
Guide for Voice Mail User's Manual.
 To configure the Voice Mail parameters:

1. Open the Voice Mail Settings page (Configuration tab > VoIP menu > GW and IP to
IP > Advanced Applications > Voice Mail Settings).
Figure 25-4: Voice Mail Settings

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25.14 Out-of-Band Digit Notifications According to KPML

The device supports Out-of-Band digit notifications according to a subset of the KPML
specification (RFC 4730). When an INVITE dialog is initiated, even in early dialog, the
device may receive a SUBSCRIBE message outside of the dialog. An application that
wants to collect digits creates an application/kpml-request+xml document with the digit
patterns of interest to the application and places this document in a SUBSCRIBE request.
This SUBSCRIBE includes the identifiers of the INVITE dialog. The SUBSCRIBE request
consists of DRegex (Digit Regular Expression). Once a subscription is established, the
device sends application/kpml-response+xml documents in NOTIFY requests once a
match is found.
If the SUBSCRIBE request has no KPML body, any KPML document running on that dialog
and addressed by the event id, if present, immediately terminates. However, the
SUBSCRIBE-initiated dialog is still active. If the dialog referenced by the KPML
subscription does not exist, the device returns a 404 Not Found response to the
SUBSCRIBE and terminates the dialog.
It's possible to “re-subscribe”. In such a scenario, the device terminates the existing KPML
request and replaces it with the new request.
Notes:
• Only one KPML subscription per participant/dialog is supported.
• Only one regex per pattern is supported.
• Only single-digit patterns are supported.
• The following tags are not supported: "pre", "flush", "stream", and "enterkey".

User's Manual 26. Analog Gateway
26 Analog Gateway
This section describes configuration of analog settings.
26.1 Configuring Keypad Features

The Keypad Features page enables you to activate and deactivate the following features
directly from the connected telephone's keypad:
 Call Forward
 Caller ID Restriction
 Hotline for automatic dialing
 Call Transfer
 Call Waiting
 Rejection of Anonymous Calls
Notes:
• The Keypad Features page is available only for FXS interfaces.
• The method used by the device to collect dialed numbers is identical to the
method used during a regular call (i.e., max digits, interdigit timeout, digit map,
etc.).
• The activation of each feature remains in effect until it is deactivated (i.e., not
deactivated after a call).
• For a description of the keypad parameters, see 'Telephone Keypad Sequence
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 To configure the keypad features

1. Open the Keypad Features page (Configuration tab > VoIP menu > GW and IP to IP
> Analog Gateway > Keypad Features).
Figure 26-1: Keypad Features Page
2. Configure the keypad features as required.


26.2 Configuring Metering Tones

The FXS interfaces can generate 12/16 KHz metering pulses toward the Tel side (e.g., for
connection to a pay phone or private meter). Tariff pulse rate is determined according to
the device's Charge Codes table. This capability enables users to define different tariffs
according to the source/destination numbers and the time-of-day. The tariff rate includes
the time interval between the generated pulses and the number of pulses generated on
answer.
Notes:
• The Metering Tones page is available only for FXS interfaces.
• Charge Code rules can be assigned to routing rules in the Tel to IP Routing (see
'Configuring Tel to IP Routing' on page 256). When a new call is established, the
Tel to IP Routing is searched for the destination IP address. Once a route is
located, the Charge Code (configured for that route) is used to associate the route
with an entry in the Charge Codes table.
 To configure Metering tones:

1. Open the Metering Tones page (Configuration tab > VoIP menu > GW and IP to IP >
Analog Gateway > Metering Tones).
Figure 26-2: Metering Tones Page
2. Configure the Metering tones parameters as required. For a description of the

parameters appearing on this page, see 'Configuration Parameters Reference' on
page 475.
If you set the 'Generate Metering Tones' parameter to Internal Table, access the Charge
Codes Table page by clicking the Charge Codes Table button. For more information
on configuring the Charge Codes table, see 'Configuring Charge Codes' on page 302.
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26.3 Configuring Charge Codes

The Charge Codes table is used to configure the metering tones (and their time interval)
that the FXS interfaces generate to the Tel side. To associate a charge code to an
outgoing Tel-to-IP call, use the Tel to IP Routing.
You can configure up to 25 different charge codes, where each table row represents a
charge code. Each charge code can include up to four different time periods in a day (24
hours). The device selects the time period by comparing the device 's current time to the
end time of each time period of the selected Charge Code. The device generates the
Number of Pulses on Answer once the call is connected and from that point on, it
generates a pulse each Pulse Interval. If a call starts at a certain time period and crosses
to the next, the information of the next time period is used.
Notes:
• The Charge Codes Table page is available only for FXS interfaces.
• The Charge Codes table can also be configured using the table ini file parameter,
ChargeCode.
 To configure the Charge Codes:

1. Open the Charge Codes Table page (Configuration tab > VoIP menu > GW and IP
to IP submenu > Analog Gateway submenu > Charge Codes). Alternatively, you can
access this page from the Metering Tones page (see 'Configuring Metering Tones' on
page 301).
Figure 26-3: Charge Codes Table Page
2. Configured the charge codes, as required. For a description of the parameters, see
the table below.

Table 26-1: Charge Codes Table Parameter Description
End Time Defines the end of the time period in a 24 hour format, hh. For
[ChargeCode_EndTime<1-4>] example, "04" denotes 4 A.M.
Notes:
 The first time period always starts at midnight (00).
 It is mandatory that the last time period of each rule end at
midnight (00). This prevents undefined time frames in a
day.
Pulse Interval Defines the time interval between pulses (in tenths of a
[ChargeCode_PulseInterval<1-4>] second).
Pulses On Answer Defines the number of pulses sent on answer.
[ChargeCode_PulsesOnAnswer<1-
4>]
26.4 Configuring FXO Settings

The FXO Settings page allows you to configure the device's specific FXO parameters. For
a description of these parameters, see 'Configuration Parameters Reference' on page 475.
Note: The FXO Settings page is available only for FXO interfaces.
 To configure the FXO parameters:

1. Open the FXO Settings page (Configuration tab > VoIP menu > GW and IP to IP >
Analog Gateway > FXO Settings).
Figure 26-4: FXO Settings Page
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26.5 Configuring Authentication

The Authentication page defines a user name and password for authenticating each device
port. Authentication is typically used for FXS interfaces, but can also be used for FXO
interfaces.
Notes:
• For configuring whether authentication is done per port or for the entire device,
use the parameter AuthenticationMode.
• If authentication is configured for the entire device, the configuration in this table is
ignored.
• If the user name or password is not configured in this table, the port's phone
number (configured in the Endpoint Phone Number tableand global password
(configured by the global parameter, Password) are used instead for
authentication of the port.
• After you click Submit, the password is displayed as an asterisk (*).
• The Authentication table can also be configured using the table ini file parameter,
Authentication (see 'Configuration Parameters Reference' on page 475).
 To configure authentication credentials per port:

1. Set the parameter 'Registration Mode' (AuthenticationMode) to Per Endpoint. This
can be configured in any of the following pages:
• Proxy & Registration page (see 'Configuring Proxy and Registration Parameters'
on page 216).
• Trunk Group Settings page (see 'Configuring Hunt Group Settings' on page 237),
where registration method is configured per Trunk Group.
2. Open the Authentication page (Configuration tab > VoIP menu > GW and IP to IP >
Analog Gateway > Authentication).
3. Configure port authentication credentials as required. For a description of the

parameters, see the table below.

Table 26-2: Authentication Table Parameter Description
User Name Defines the user name used for authenticating the port.
[Authentication_UserId]
Password Defines the password used for authenticating the port.
[Authentication_UserPassword]
26.6 Configuring Automatic Dialing

The Automatic Dialing page allows you to define a telephone number that is automatically
dialed when an FXS or FXO port goes off-hook. The dialing can be done immediately upon
off-hook, or after a user-defined interval after off-hook referred to as Hotline dialing.
Note: The Automatic Dialing can also be configured using the table ini file
parameter, TargetOfChannel.
 To configure automatic dialing per port:

1. Open the Automatic Dialing page (Configuration tab > VoIP menu > GW and IP to IP
submenu > Analog Gateway > Automatic Dialing).
Figure 26-5: Automatic Dialing Page
The first table entry in the figure above enables Hotline automatic dialing for an FXS
port, whereby if the port is off-hooked for over 15 seconds, the device automatically
dials 911.
2. Configure automatic dialing per port, as required. See the table below for parameter
descriptions.
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Table 26-3: Automatic Dialing Table Parameter Description
Gateway Port Lists the FXS or FXO port for which you want to configure automatic
dialing.
Destination Phone Number Defines the destination telephone number to automatically dial.
[TargetOfChannel_Destina
tion]
Auto Dial Status Enables automatic dialing.
[TargetOfChannel_Type]  [0] Disable = Automatic dialing for the specific port is disabled.
 [1] Enable = (Default) Automatic dialing is enabled and the phone
number configured in the 'Destination Phone Number' field is
automatically dialed if the following occurs:
 FXS interfaces: The phone is off-hooked
 FXO interfaces: A ring signal (from a PBX/PSTN switch) is
detected on the FXO line. The device initiates a call to the
destination without seizing the FXO line. The line is seized only
after the SIP call is answered.
 [2] Hotline = Automatic dialing is done after an interval configured
by the 'Hotline Dial Tone Duration' parameter:
 FXS interfaces: When the phone is off-hooked and no digit is
dialed within a user-defined time, the configured destination
number is automatically dialed.
 FXO interfaces: If a ring signal is detected, the device seizes
the FXO line, plays a dial tone, and then waits for DTMF digits.
If no digits are detected within a user-defined time, the
configured destination number is automatically dialed by
sending a SIP INVITE message with this number.
Hotline Dial Tone Duration Defines the duration (in seconds) after which the destination phone
[TargetOfChannel_HotLine number is automatically dialed. This is applicable only if the port has
ToneDuration] been configured for Hotline (i.e., 'Auto Dial Status' is set to Hotline).
Note: You can configure this Hotline interval for all ports, using the
global parameter, HotLineToneDuration.

26.7 Configuring Caller Display Information

The Caller Display Information table allows you to define a caller identification string (Caller
ID) for FXS and FXO ports and enable the device to send the Caller ID to the IP when a
call is made. The called party can use this information for caller identification.
The device sends the configured caller ID in the outgoing INVITE message's From header.
For information on Caller ID restriction according to destination/source prefixes, see
'Configuring Source/Destination Number Manipulation' on page 241.
Notes:
• If an FXS port receives 'Private' or 'Anonymous' strings in the SIP From header,
the calling name or number is not sent to the Caller ID display.
• If Caller ID is detected on an FXO line (EnableCallerID = 1), it is used instead of
the Caller ID configured in this table.
• If you set the 'Caller ID/Name' parameter to the strings "Private" or "Anonymous",
Caller ID is restricted and the settings of the 'Presentation' parameter is ignored.
• The Caller Display Information table can also be configured using the table ini file
parameter, CallerDisplayInfo.
 To configure Caller Display:

1. Open the Caller Display Information page (Configuration tab > VoIP menu > GW and
IP to IP > Analog Gateway > Caller Display Information).
Figure 26-6: Caller Display Information Page
2. Configure the table as required. For a description of the parameters, see the table
below.
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Table 26-4: Caller Display Parameter Description
Gateway Port Displays the port.

[CallerDisplayInfo_Port]
Caller ID/Name Defines the Caller ID string.

[CallerDisplayInfo_DisplayString] The valid value is a string of up to 18 characters.
Presentation Enables the sending of the caller ID string.
[CallerDisplayInfo_IsCidRestricted]  [0] Allowed = The caller ID string is sent when a Tel-to-IP
call is made.
 [1] Restricted = The caller ID string is not sent.
Notes:
 This parameter is overridden by the 'Presentation'
parameter in the Source Number Manipulation table (see
'Configuring Source/Destination Number Manipulation' on
page 241).
 If this parameter is set to Restricted, the Caller ID is sent
to the remote side using only the SIP P-Asserted-Identity
and P-Preferred-Identity headers (AssertedIdMode).

26.8 Configuring Call Forward

The Call Forwarding table allows you to configure call forwarding per port for IP-to-Tel
calls. This redirects the call (using SIP 302 response) initially destined to a specific device
Tel port, to a different device port or to an IP destination.
Notes:
• To enable call forwarding, set the 'Enable Call Forward' parameter to Enable. This
is done in the Supplementary Services page (Configuration tab > VoIP menu >
GW and IP to IP > DTMF and Supplementary > Supplementary Services).
• The Call Forward table can also be configured using the table ini file parameter,
FwdInfo.
 To configure Call Forward per port:

1. Open the Call Forward Table page (Configuration tab > VoIP menu > GW and IP to
IP > Analog Gateway > Call Forward).
Figure 26-7: Call Forward Table Page
2. Configure the table as required. For descriptions of the parameters, see the table
below.
Table 26-5: Call Forward Table Parameter Description
Forward Type Defines the condition upon which the call is forwarded.
[FwdInfo_Type]  [0] Deactivate = (Default) Don't forward incoming calls.
 [1] On Busy = Forward incoming calls when the port is busy.
 [2] Unconditional = Always forward incoming calls.
 [3] No Answer = Forward incoming calls that are not answered within
the time specified in the 'Time for No Reply Forward' field.
 [4] On Busy or No Answer = Forward incoming calls when the port is
busy or when calls are not answered within the time specified in the
'Time for No Reply Forward' field.
 [5] Do Not Disturb = Immediately reject incoming calls.
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Forward to Phone Number Defines the telephone number or URI (<number>@<IP address>) to
[FwdInfo_Destination] where the call is forwarded.
Note: If this parameter is configured with only a telephone number and a
Proxy isn't used, this forwarded-to phone number must be specified in
the Tel to IP Routing (see 'Configuring Tel to IP Routing' on page 256).
Time for No Reply If you have set the 'Forward Type' for this port to No Answer, then
Forward configure the number of seconds the device waits before forwarding the
[FwdInfo_NoReplyTime] call to the specified phone number.
26.9 Configuring Caller ID Permissions

The Caller ID Permissions table allows you to enable per port, Caller ID generation for FXS
interfaces and Caller ID detection for FXO interfaces.
Notes:
• If Caller ID permissions is not configured for a port in this table, its Caller ID
generation / detection is determined according to the global parameter, 'Enable
Call ID' in the Supplementary Services page (Configuration tab > VoIP menu >
GW and IP to IP > DTMF and Supplementary > Supplementary Services).
• The Caller ID Permissions table can also be configured using the table ini file
parameter, EnableCallerID.
 To configure Caller ID permissions per port:

1. Open the Caller ID Permissions page (Configuration tab > VoIP menu > GW and IP
to IP > Analog Gateway > Caller ID Permissions).
Figure 26-8: Caller ID Permissions Page
2. Configure the table as required. For a description of the parameter, see the table
below.

Table 26-6: Caller ID Permissions Table Parameter Description
Caller ID Enables Caller ID generation (FXS) or detection (FXO) per port.

[EnableCallerId_IsEnabled]  [0] Disable
 [1] Enable
26.10 Configuring Call Waiting

The Call Waiting table allows you to enable or disable call waiting per FXS port.
Notes:
• This page is applicable only to FXS interfaces.
• You can enable or disable call waiting for all the device's ports using the global
parameter, 'Enable Call Waiting' in the Supplementary Services page
(Configuration tab > VoIP menu > GW and IP to IP > DTMF and
Supplementary > Supplementary Services).
• The CPT file installed on the device must include a 'call waiting Ringback' tone
(caller side) and a 'call waiting' tone (called side, FXS interfaces only).
• The EnableHold parameter must be enabled on both the calling and the called
sides.
• For additional call waiting configuration, see the following parameters:
FirstCallWaitingToneID (in the CPT file), TimeBeforeWaitingIndication,
WaitingBeepDuration, TimeBetweenWaitingIndications, and
NumberOfWaitingIndications.
• The Call Waiting table can also be configured using the table ini file parameter,
CallWaitingPerPort.
 To enable call waiting per port:

1. Open the Call Waiting page (Configuration tab > VoIP menu > GW and IP to IP >
Analog Gateway > Call Waiting).
Figure 26-9: Call Waiting Page
2. Configure the table as required. For a description of the parameter, see the table
below.
Table 26-7: Call Waiting Table Parameter Description
Call Waiting Configuration Enables call waiting for the port.
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[CallWaitingPerPort_IsEnabled]  [0] Disable
 [1] Enable = Enables call waiting for the port. When the device
receives a call on a busy port, it responds with a SIP 182
response (not with a 486 busy). The device plays a call waiting
indication signal. When the device detects a hook-flash from
the FXS port, the device switches to the waiting call. The
device that initiated the waiting call plays a call waiting
ringback tone to the calling party after a 182 response is
received.
26.11 Rejecting Anonymous Calls

You can configure the device to reject anonymous calls received from the IP and destined
for FXS interfaces. This can be configured using the ini file parameter,
RejectAnonymousCallPerPort. If configured, when an FXS interface receives an
anonymous call, the device rejects the call and responds with a SIP 433 (Anonymity
Disallowed) response. For a description of the parameter see 'Caller ID Parameters' on
page 582.
26.12 Configuring FXS Distinctive Ringing and Call

Waiting Tones per Source/Destination Number
You can configure a distinctive ringing tone and call waiting tone per calling (source) and/or
called (destination) number (or prefix) for IP-to-Tel calls. This feature can be configured per
FXS endpoint or for a range of FXS endpoints. Therefore, different tones can be played per
FXS endpoint depending on the source and/or destination number of the received call. You
can also configure multiple entries with different source and/or destination prefixes and
tones for the same FXS port.
Typically, the played ring and/or call waiting tone is indicated in the SIP Alert-info header
field of the received INVITE message. If this header is not present in the received INVITE,
then this feature is used and the tone played is according to the settings in this table.
Notes:
• This page is applicable only to FXS interfaces.
• The Tone Index table can also be configured using the table ini file parameter,
ToneIndex.
 To configure distinctive ringing and call waiting per FXS port:

1. Open the Tone Index Table page (Configuration tab > VoIP menu > GW and IP to IP
> Analog Gateway > Tone Index).


Figure 26-10: Tone Index Table Page
The figure above shows a configuration example for using distinctive ringing and call
waiting tones of Index #9 ('Priority Index' 1) in the CPT file for FXS endpoints 1 to 4
when a call is received from a source number with prefix 2.
3. Configure the table as required. For a description of the parameters, see the table
below.
Table 26-8: Tone index Table Parameter Description
Index Defines the table index entry. Up to 50 entries can be defined.

FXS Port First Defines the first port in the FXS port range.
[ToneIndex_FXSPort_First]
FXS Port Last Defines the last port in the FXS port range.
[ToneIndex_FXSPort_Last]
Source Prefix Defines the prefix of the calling number.
[ToneIndex_SourcePrefix]
Destination Prefix Defines the prefix of the called number.
[ToneIndex_DestinationPrefix]
Priority Index Defines the index of the distinctive ringing and call waiting tones.
[ToneIndex_PriorityIndex] The call waiting tone index equals to the Priority Index plus the
value of the FirstCallWaitingToneID parameter. For example, if you
want to use the call waiting tone in the CPT file at Index #9, you
need to enter "1" as the Priority Index value and set the
FirstCallWaitingToneID parameter to "8". The summation of these
values is 9, i.e., index #9.
The default is 0.
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26.13 FXS/FXO Coefficient Types

The FXS Coefficient and FXO Coefficient types used by the device can be one of the
following:
 US line type of 600 ohm AC impedance and 40 V RMS ringing voltage for REN = 2
 European standard (TBR21)
These Coefficient types are used to increase return loss and trans-hybrid loss performance
for two telephony line type interfaces (US or European). This adaptation is performed by
modifying the telephony interface characteristics. This means, for example, that changing
impedance matching or hybrid balance doesn't require hardware modifications, so that a
single device is able to meet requirements for different markets. The digital design of the
filters and gain stages also ensures high reliability, no drifts (over temperature or time) and
simple variations between different line types.
The FXS Coefficient types provide best termination and transmission quality adaptation for
two FXS line type interfaces. This parameter affects the following AC and DC interface
parameters:
 DC (battery) feed characteristics
 AC impedance matching
 Transmit gain
 Receive gain
 Hybrid balance
 Frequency response in transmit and receive direction
 Hook thresholds
 Ringing generation and detection parameters
 To select the FXO and FXS Coefficient types:

1. Open the Analog Settings page (Configuration tab > VoIP menu > Media > Analog
Settings). This page includes the Coefficient type parameters, as shown below:
Figure 26-11: FXS/FXO Coefficient Parameters in Analog Settings Page
2. From the 'FXS Coefficient Type' drop-down list (FXSCountryCoefficients), select the
required FXS Coefficient type.
3. From the 'FXO Coefficient Type' drop-down list (CountryCoefficients), select the
required FXO Coefficient type.
4. Click Submit.
5. Save your settings to the flash memory ("burn") with a device reset.

26.14 FXO Operating Modes

This section provides a description of the device's FXO operating modes:
 For IP-to-Tel calls (see 'FXO Operations for IP-to-Tel Calls' on page 315)
 For Tel-to-IP calls (see 'FXO Operations for Tel-to-IP Calls' on page 317)
 Call termination on FXO devices (see 'Call Termination on FXO Devices' on page 320)
26.14.1 FXO Operations for IP-to-Tel Calls

The FXO device provides the following operating modes for IP-to-Tel calls:
 One-stage dialing (see 'One-Stage Dialing' on page 315)
• Waiting for dial tone (see 'Two-Stage Dialing' on page 316)
• Time to wait before dialing
• Answer supervision
 Two-stage dialing (see 'Two-Stage Dialing' on page 316)
 Dialing time: DID wink (see 'DID Wink' on page 317)
26.14.1.1 One-Stage Dialing

One-stage dialing is when the FXO device receives an IP-to-Tel call, off-hooks the PBX
line connected to the telephone, and then immediately dials the destination telephone
number. In other words, the IP caller doesn't dial the PSTN number upon hearing a dial
tone.
Figure 26-12: Call Flow for One-Stage Dialing
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One-stage dialing incorporates the following FXO functionality:

 Waiting for Dial Tone: Enables the device to dial the digits to the Tel side only after
detecting a dial tone from the PBX line. The ini file parameter IsWaitForDialTone is
used to configure this operation.
 Time to Wait Before Dialing: Defines the time (in msec) between seizing the FXO
line and starting to dial the digits. The ini file parameter WaitForDialTime is used to
configure this operation.
Note: The ini file parameter IsWaitForDialTone must be disabled for this mode.
 Answer Supervision: The Answer Supervision feature enables the FXO device to
determine when a call is connected, by using one of the following methods:
• Polarity Reversal: the device sends a 200 OK in response to an INVITE only
when it detects a polarity reversal.
• Voice Detection: the device sends a 200 OK in response to an INVITE only
when it detects the start of speech (fax or modem answer tone) from the Tel side.
Note that the IPM detectors must be enabled.
26.14.1.2 Two-Stage Dialing

Two-stage dialing is when the IP caller is required to dial twice. The caller initially dials to
the FXO device and only after receiving a dial tone from the PBX (via the FXO device),
dials the destination telephone number.
Two-stage dialing implements the Dialing Time feature. Dialing Time allows you to define
the time that each digit can be separately dialed. By default, the overall dialing time per
digit is 200 msec. The longer the telephone number, the greater the dialing time.

The relevant parameters for configuring Dialing Time include the following:
 DTMFDigitLength (100 msec): time for generating DTMF tones to the PSTN (PBX)
side
 DTMFInterDigitInterval (100 msec): time between generated DTMF digits to PSTN
(PBX) side
26.14.1.3 DID Wink

The device's FXO ports support Direct Inward Dialing (DID). DID is a service offered by
telephone companies that enables callers to dial directly to an extension on a PBX without
the assistance of an operator or automated call attendant. This service makes use of DID
trunks, which forward only the last three to five digits of a phone number to the PBX. If, for
example, a company has a PBX with extensions 555-1000 to 555-1999, and a caller dials
555-1234, the local central office (CO) would forward, for example, only 234 to the PBX.
The PBX would then ring extension 234.
DID wink enables the originating end to seize the line by going off-hook. It waits for
acknowledgement from the other end before sending digits. This serves as an integrity
check that identifies a malfunctioning trunk and allows the network to send a re-order tone
to the calling party.
The "start dial" signal is a wink from the PBX to the FXO device. The FXO then sends the
last four to five DTMF digits of the called number. The PBX uses these digits to complete
the routing directly to an internal station (telephone or equivalent).
 DID Wink can be used for connection to EIA/TIA-464B DID Loop Start lines
 Both FXO (detection) and FXS (generation) are supported
26.14.2 FXO Operations for Tel-to-IP Calls

The FXO device provides the following FXO operating modes for Tel-to-IP calls:
 Automatic Dialing (see 'Automatic Dialing' on page 317)
 Collecting Digits Mode (see 'Collecting Digits Mode' on page 319)
 FXO Supplementary Services (see 'FXO Supplementary Services' on page 319)
• Hold/Transfer Toward the Tel side
• Hold/Transfer Toward the IP side
• Blind Transfer to the Tel side
26.14.2.1 Automatic Dialing

Automatic dialing is defined using the Web interface's Automatic Dialing (TargetOfChannel
ini file parameter) page, described in see 'Configuring Automatic Dialing' on page 305.
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The SIP call flow diagram below illustrates Automatic Dialing.

Figure 26-13: Call Flow for Automatic Dialing

26.14.2.2 Collecting Digits Mode

When automatic dialing is not defined, the device collects the digits.
The SIP call flow diagram below illustrates the Collecting Digits Mode.
26.14.2.3 FXO Supplementary Services

The FXO supplementary services include the following:
 Hold / Transfer toward the Tel side: The ini file parameter LineTransferMode must
be set to 0 (default). If the FXO receives a hook-flash from the IP side (using out-of-
band or RFC 2833), the device sends the hook-flash to the Tel side by performing one
of the following:
• Performing a hook flash (i.e., on-hook and off-hook)
• Sending a hook-flash code (defined by the ini file parameter HookFlashCode)
The PBX may generate a dial tone that is sent to the IP, and the IP side may dial digits
of a new destination.
 Blind Transfer to the Tel side: A blind transfer is one in which the transferring phone
connects the caller to a destination line before ringback begins. The ini file parameter
LineTransferMode must be set to 1.
The blind transfer call process is as follows:
• FXO receives a REFER request from the IP side
• FXO sends a hook-flash to the PBX, dials the digits (that are received in the
Refer-To header), and then drops the line (on-hook). Note that the time between
flash to dial is according to the WaitForDialTime parameter.
• PBX performs the transfer internally
 Hold / Transfer toward the IP side: The FXO device doesn't initiate hold / transfer as
a response to input from the Tel side. If the FXO receives a REFER request (with or
without replaces), it generates a new INVITE according to the Refer-To header.
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26.14.3 Call Termination on FXO Devices

This section describes the device's call termination capabilities for its FXO interfaces:
 Calls terminated by a PBX (see 'Call Termination by PBX' on page 320)
 Calls terminated before call establishment (see 'Call Termination before Call
Establishment' on page 321)
 Ring detection timeout (see 'Ring Detection Timeout' on page 321)
26.14.3.1 Calls Termination by PBX

The FXO device supports various methods for identifying when a call has been terminated
by the PBX.
The PBX doesn't disconnect calls, but instead signals to the device that the call has been
disconnected using one of the following methods:
 Detection of polarity reversal/current disconnect: The call is immediately
disconnected after polarity reversal or current disconnect is detected on the Tel side
(assuming the PBX/CO generates this signal). This is the recommended method.
Relevant parameters: EnableReversalPolarity, EnableCurrentDisconnect,
CurrentDisconnectDuration, CurrentDisconnectDefaultThreshold, and
TimeToSampleAnalogLineVoltage.
 Detection of Reorder, Busy, Dial, and Special Information Tone (SIT) tones: The
call is immediately disconnected after a Reorder, Busy, Dial, or SIT tone is detected
on the Tel side (assuming the PBX / CO generates this tone). This method requires
the correct tone frequencies and cadence to be defined in the Call Progress Tones
file. If these frequencies are unknown, define them in the CPT file. The tone produced
by the PBX / CO must be recorded and its frequencies analyzed.
Relevant parameters: DisconnectOnBusyTone and DisconnectOnDialTone.
 Detection of silence: The call is disconnected after silence is detected on both call
directions for a specific (configurable) amount of time. The call isn’t disconnected
immediately; therefore, this method should only be used as a backup option.
Relevant parameters: EnableSilenceDisconnect and FarEndDisconnectSilencePeriod.
 Special DTMF code: A digit pattern that when received from the Tel side, indicates to
the device to disconnect the call.
Relevant ini file parameter: TelDisconnectCode.
 Interruption of RTP stream: Relevant parameters: BrokenConnectionEventTimeout
and DisconnectOnBrokenConnection.
Note: This method operates correctly only if silence suppression is not used.
 Protocol-based termination of the call from the IP side
Note: The implemented disconnect method must be supported by the CO or PBX.

26.14.3.2 Call Termination before Call Establishment

The device supports the following call termination methods before a call is established:
 Call termination upon receipt of SIP error response (in Automatic Dialing mode):
By default, when the FXO device operates in Automatic Dialing mode, there is no
method to inform the PBX if a Tel-to-IP call has failed (SIP error response - 4xx, 5xx or
6xx - is received). The reason is that the FXO device does not seize the line until a
SIP 200 OK response is received. Use the FXOAutoDialPlayBusyTone parameter to
allow the device to play a busy / reorder tone to the PSTN line if a SIP error response
is received. The FXO device seizes the line (off-hook) for the duration defined by the
TimeForReorderTone parameter. After playing the tone, the line is released (on-hook).
 Call termination after caller (PBX) on-hooks phone (Ring Detection Timeout
feature): This method operates in one of the following manners:
• Automatic Dialing is enabled: if the remote IP party doesn't answer the call and
the ringing signal (from the PBX) stops for a user-defined time (configured by the
parameter FXOBetweenRingTime), the FXO device releases the IP call.
• No automatic dialing and Caller ID is enabled: the device seizes the line after
detection of the second ring signal (allowing detection of caller ID sent between
the first and the second rings). If the second ring signal is not received within this
timeout, the device doesn't initiate a call to IP.
26.14.3.3 Ring Detection Timeout

The operation of Ring Detection Timeout depends on the following:
 Automatic dialing is disabled and Caller ID is enabled: if the second ring signal is
not received for a user-defined time (using the parameter FXOBetweenRingTime), the
FXO device doesn’t initiate a call to the IP.
 Automatic dialing is enabled: if the remote party doesn't answer the call and the
ringing signal stops for a user-defined time (using the parameter
FXOBetweenRingTime), the FXO device releases the IP call.
Ring Detection Timeout supports full ring cycle of ring on and ring off (from ring start to ring
start).
26.15 Remote PBX Extension between FXO and FXS

Devices
Remote PBX extension offers a company the capability of extending the "power" of its local
PBX by allowing remote phones (remote offices) to connect to the company's PBX over the
IP network (instead of via PSTN). This is as if the remote office is located in the head office
(where the PBX is installed). PBX extensions are connected through FXO ports to the IP
network, instead of being connected to individual telephone stations. At the remote office,
FXS units connect analog phones to the same IP network. To produce full transparency,
each FXO port is mapped to an FXS port (i.e., one-to-one mapping). This allows individual
extensions to be extended to remote locations. To call a remote office worker, a PBX user
or a PSTN caller simply dials the PBX extension that is mapped to the remote FXS port.
This section provides an example on how to implement a remote telephone extension
through the IP network, using 8-port FXO and 8-port FXS interfaces. In this configuration,
the FXO device routes calls received from the PBX to the ‘Remote PBX Extension’
connected to the FXS device. The routing is transparent as if the telephone connected to
the FXS device is directly connected to the PBX.
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The following is required:

 One FXO interfaces with ports connected directly to the PBX lines (shown in the figure
below)
 One FXS interfaces for the 'remote PBX extension'
 Analog phones (POTS)
 PBX (one or more PBX loop start lines)
 LAN network
Figure 26-14: FXO-FXS Remote PBX Extension (Example)
26.15.1 Dialing from Remote Extension (Phone at FXS)

The procedure below describes how to dial from the 'remote PBX extension' (i.e., phone
connected to the FXS interface).
 To make a call from the FXS interface:

1. Off-hook the phone and wait for the dial tone from the PBX. This is as if the phone is
connected directly to the PBX. The FXS and FXO interfaces establish a voice path
connection from the phone to the PBX immediately after the phone is off-hooked.
2. Dial the destination number (e.g., phone number 201). The DTMF digits are sent over
IP directly to the PBX. All the audible tones are generated from the PBX (such as
ringback, busy, or fast busy tones). One-to-one mapping occurs between the FXS
ports and PBX lines.
3. The call disconnects when the phone connected to the FXS goes on-hook.

26.15.2 Dialing from PBX Line or PSTN

The procedure below describes how to dial from a PBX line (i.e., from a telephone directly
connected to the PBX) or from the PSTN to the 'remote PBX extension' (i.e., telephone
connected to the FXS interface).
 To dial from a telephone directly connected to the PBX or from the PSTN:
 Dial the PBX subscriber number (e.g., phone number 101) in the same way as if the
user’s phone was connected directly to the PBX. As soon as the PBX rings the FXO
device, the ring signal is ‘sent’ to the phone connected to the FXS device. Once the
phone connected to the FXS device is off-hooked, the FXO device seizes the PBX line
and the voice path is established between the phone and PBX.
There is one-to-one mapping between PBX lines and FXS device ports. Each PBX
line is routed to the same phone (connected to the FXS device). The call disconnects
when the phone connected to the FXS device is on-hooked.
26.15.3 Message Waiting Indication for Remote Extensions

The device supports the relaying of Message Waiting Indications (MWI) for remote
extensions (and voice mail applications). Instead of subscribing to an MWI server to
receive notifications of pending messages, the FXO device receives subscriptions from the
remote FXS device and notifies the appropriate extension when messages (and the
number of messages) are pending.
The FXO device detects an MWI message from the Tel (PBX) side using any one of the
following methods:
 100 VDC (sent by the PBX to activate the phone's lamp)
 Stutter dial tone from the PBX
 MWI display signal (according to the parameter CallerIDType)
Upon detection of an MWI message, the FXO device sends a SIP NOTIFY message to the
IP side. When receiving this NOTIFY message, the remote FXS device generates an MWI
signal toward its Tel side.
Figure 26-15: MWI for Remote Extensions
26.15.4 Call Waiting for Remote Extensions

When the FXO device detects a Call Waiting indication (FSK data of the Caller Id -
CallerIDType2) from the PBX, it sends a proprietary INFO message, which includes the
caller identification to the FXS device. Once the FXS device receives this INFO message, it
plays a call waiting tone and sends the caller ID to the relevant port for display. The remote
extension connected to the FXS device can toggle between calls using the Hook Flash
button.
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MP-11x & MP-124
Figure 26-16: Call Waiting for Remote Extensions
26.15.5 FXS Gateway Configuration

The procedure below describes how to configure the FXS interface (at the 'remote PBX
extension').
 To configure the FXS interface:

1. In the Endpoint Phone Numbers page (see Configuring Endpoint Phone Numbers on
page 235, assign the phone numbers 100 to 107 to the device's endpoints.
2. In the Automatic Dialing page (see 'Configuring Automatic Dialing' on page 305), enter
the phone numbers of the FXO device in the ‘Destination Phone Number’ fields. When
a phone connected to Port #1 off-hooks, the FXS device automatically dials the
number ‘200’.

3. In the Tel to IP Routing page (see 'Configuring Tel to IP Routing' on page 256), enter
20 for the destination phone prefix, and 10.1.10.2 for the IP address of the FXO
device.
Note: For the transfer to function in remote PBX extensions, Hold must be disabled
at the FXS device (i.e., Enable Hold = 0) and hook-flash must be transferred from the
FXS to the FXO (HookFlashOption = 4).
26.15.6 FXO Gateway Configuration

The procedure below describes how to configure the FXO interface (to which the PBX is
directly connected).
 To configure the FXO interface:

1. In the Endpoint Phone Numbers page (see Configuring Endpoint Phone Numbers on
page 235, assign the phone numbers 200 to 207 to the device’s FXO endpoints.
2. In the Automatic Dialing page, enter the phone numbers of the FXS device in the
‘Destination Phone Number’ fields. When a ringing signal is detected at Port #1, the
FXO device automatically dials the number ‘100’.
3. In the Tel to IP Routing page, enter 10 in the ‘Destination Phone Prefix’ field, and the
IP address of the FXS device (10.1.10.3) in the field ‘IP Address’.
Figure 26-17: FXO Tel-to-IP Routing Configuration
4. In the FXO Settings page (see 'Configuring FXO Parameters' on page 303), set the
parameter ‘Dialing Mode’ to Two Stages (IsTwoStageDial = 1).
Version 6.6 325 MP-11x & MP-124

MP-11x & MP-124

Part VI
Stand-Alone Survivability
Application
User's Manual 27. SAS Overview
27 SAS Overview
The device's Stand-Alone Survivability (SAS) feature ensures telephony communication
continuity (survivability) for enterprises using hosted IP services (such as IP Centrex) or IP-
PBX in cases of failure of these entities. In case of failure of the IP Centrex, IP-PBX
servers (or even WAN connection and access Internet modem), the enterprise typically
loses its internal telephony service at any branch, between its offices, and with the external
environment. Typically, these failures also lead to the inability to make emergency calls
(e.g., 911 in North America). Despite these possible points of failure, the device's SAS
feature ensures that the enterprise's telephony services (e.g., SIP IP phones or soft
phones) are maintained, by routing calls to the PSTN (i.e., providing PSTN fallback).
Notes:
• Throughput this section, the term user agent (UA) refers to the enterprise's LAN
phone user (i.e., SIP telephony entities such as IP phones).
• Throughout this section, the term proxy or proxy server refers to the enterprise's
centralized IP Centrex or IP-PBX.
• Throughout this section, the term SAS refers to the SAS application running on the
device.
27.1 SAS Operating Modes

The device's SAS application can be implemented in one of the following main modes:
 Outbound Proxy: In this mode, SAS receives SIP REGISTER requests from the
enterprise's UAs and forwards these requests to the external proxy (i.e., outbound
proxy). When a connection with the external proxy fails, SAS enters SAS emergency
state and serves as a proxy, by handling internal call routing for the enterprise's UAs -
routing calls between UAs and if setup, routing calls between UAs and the PSTN. For
more information, see 'SAS Outbound Mode' on page 329.
 Redundant Proxy: In this mode, the enterprise's UAs register with the external proxy
and establish calls directly through the external proxy, without traversing SAS (or the
device per se'). Only when connection with the proxy fails, do the UAs register with
SAS, serving now as the UAs redundant proxy. SAS then handles the calls between
UAs, and between the UAs and the PSTN (if setup). This mode is operational only
during SAS in emergency state. This mode can be implemented, for example, for
proxies that accept only SIP messages that are sent directly from the UAs. For more
information, see 'SAS Redundant Mode' on page 331.
Note: It is recommended to implement the SAS outbound mode.
27.1.1 SAS Outbound Mode

This section describes the SAS outbound mode, which includes the following states:
 Normal state (see 'Normal State' on page 330)
 Emergency state (see 'Emergency State' on page 330)
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27.1.1.1 Normal State

In normal state, SAS receives REGISTER requests from the enterprise's UAs and forwards
them to the external proxy (i.e., outbound proxy). Once the proxy replies with a SIP 200
OK, the device records the Contact and address of record (AOR) of the UAs in its internal
SAS registration database. Therefore, in this mode, SAS maintains a database of all the
registered UAs in the network. SAS also continuously maintains a keep-alive mechanism
toward the external proxy, using SIP OPTIONS messages. The figure below illustrates the
operation of SAS outbound mode in normal state:
Figure 27-1: SAS Outbound Mode in Normal State (Example)
27.1.1.2 Emergency State

When a connection with the external proxy fails (detected by the device's keep-alive
messages), the device enters SAS emergency state. The device serves as a proxy for the
UAs, by handling internal call routing of the UAs (within the LAN enterprise).
Note: SAS can also enter Emergency state if no response is received from the proxy
for sent OPTIONS, INVITE, or REGISTER messages. To configure this, set the
SASEnteringEmergencyMode parameter to 1.
When the device receives calls, it searches its SAS registration database to locate the
destination address (according to AOR or Contact). If the destination address is not found,
SAS forwards the call to the default gateway. Typically, the default gateway is defined as
the device itself (on which SAS is running), and if the device has PSTN interfaces, the
enterprise preserves its capability for outgoing calls (from UAs to the PSTN network).
The routing logic of SAS in emergency state is described in detail in 'SAS Routing in
Emergency State' on page 335.

The figure below illustrates the operation of SAS outbound mode in emergency state:
Figure 27-2: SAS Outbound Mode in Emergency State (Example)
When emergency state is active, SAS continuously attempts to communicate with the
external proxy, using keep-alive SIP OPTIONS. Once connection to the proxy returns, the
device exits SAS emergency state and returns to SAS normal state, as explained in 'Exiting
Emergency and Returning to Normal State' on page 332.
27.1.2 SAS Redundant Mode

In SAS redundant mode, the enterprise's UAs register with the external proxy and establish
calls directly through it, without traversing SAS (or the device per se'). Only when
connection with the proxy fails, do the UAs register with SAS, serving now as the UAs
redundant proxy. SAS then handles the calls between UAs, and between the UAs and the
PSTN (if setup).
This mode is operational only during SAS in emergency state.
Note: In this SAS deployment, the UAs (e.g., IP phones) must support configuration
for primary and secondary proxy servers (i.e., proxy redundancy), as well as homing.
Homing allows the UAs to switch back to the primary server from the secondary proxy
once the connection to the primary server returns (UAs check this using keep-alive
messages to the primary server). If homing is not supported by the UAs, you can
configure SAS to ignore messages received from UAs in normal state (the 'SAS
Survivability Mode' parameter must be set to 'Always Emergency' / 2) and thereby,
“force” the UAs to switch back to their primary proxy.
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27.1.2.1 Normal State

In normal state, the UAs register and operate directly with the external proxy.
Figure 27-3: SAS Redundant Mode in Normal State (Example)
27.1.2.2 Emergency State

If the UAs detect that their primary (external) proxy does not respond, they immediately
register to SAS and start routing calls to it.
Figure 27-4: SAS Redundant Mode in Emergency State (Example)
27.1.2.3 Exiting Emergency and Returning to Normal State

Once the connection with the primary proxy is re-established, the following occurs:
 UAs: Switch back to operate with the primary proxy.
 SAS: Ignores REGISTER requests from the UAs, forcing the UAs to switch back to
the primary proxy.
Note: This is applicable only if the 'SAS Survivability Mode' parameter is set to
'Always Emergency' (2).

27.2 SAS Routing

This section provides flowcharts describing the routing logic for SAS in normal and
emergency states.
27.2.1 SAS Routing in Normal State

The flowchart below displays the routing logic for SAS in normal state for INVITE
messages received from UAs:
Figure 27-5: Flowchart of INVITE from UA's in SAS Normal State
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The flowchart below displays the routing logic for SAS in normal state for INVITE
messages received from the external proxy:
Figure 27-6: Flowchart of INVITE from Primary Proxy in SAS Normal State

27.2.2 SAS Routing in Emergency State

The flowchart below shows the routing logic for SAS in emergency state:
Figure 27-7: Flowchart for SAS Emergency State
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User's Manual 28. SAS Configuration
28 SAS Configuration
SAS supports various configuration possibilities, depending on how the device is deployed
in the network and the network architecture requirements. This section provides step-by-
step procedures on configuring the SAS application, using the device's Web interface.
The SAS configuration includes the following:
 General SAS configuration that is common to all SAS deployment types (see 'General
SAS Configuration' on page 337)
 SAS outbound mode (see 'Configuring SAS Outbound Mode' on page 340)
 SAS redundant mode (see 'Configuring SAS Redundant Mode' on page 340)
 Gateway and SAS applications deployed together (see 'Configuring Gateway
Application with SAS' on page 341)
 Optional, advanced SAS features (see 'Advanced SAS Configuration' on page 345)
28.1 General SAS Configuration

This section describes the general configuration required for the SAS application. This
configuration is applicable to all SAS modes.
28.1.1 Enabling the SAS Application

Before you can configure SAS, you need to enable the SAS application on the device.
Once enabled, the SAS menu and related pages appear in the device's Web interface.
 To enable the SAS application:

1. Open the Applications Enabling page (Configuration tab > VoIP menu >
Applications Enabling > Applications Enabling).
2. From the 'SAS Application' drop-down list, select Enable.
3. Click Submit.
4. Save the changes to the flash memory with a device reset.
28.1.2 Configuring Common SAS Parameters

The procedure below describes how to configure SAS settings that are common to all SAS
modes. This includes various SAS parameters as well as configuring the Proxy Set for the
SAS proxy (if required). The SAS Proxy Set ID defines the address of the UAs' external
proxy.
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 To configure common SAS settings:

1. Open the SAS Configuration page (Configuration tab > VoIP menu > SAS >
Configuration).
2. Define the port used for sending and receiving SAS messages. This can be any of the
following port types:
• UDP port - defined in the 'SAS Local SIP UDP Port' field
• TCP port - defined in the 'SAS Local SIP TCP Port' field
• TLS port - defined in the 'SAS Local SIP TLS Port' field
Note: This SAS port must be different than the device's local gateway port (i.e., that
defined for the 'SIP UDP/TCP/TLS Local Port' parameter in the SIP General
Parameters page - Configuration tab > VoIP menu > SIP Definitions > General
Parameters).
3. In the 'SAS Default Gateway IP' field, define the IP address and port (in the format
x.x.x.x:port) of the device (i.e., Gateway application). Note that the port of the device is
defined by the parameter ‘SIP UDP Local Port’ (refer to the note in Step 2 above).
4. In the 'SAS Registration Time' field, define the value for the SIP Expires header, which
is sent in the 200 OK response to an incoming REGISTER message when SAS is in
emergency state.
5. From the 'SAS Binding Mode' drop-down list, select the database binding mode:
• 0-URI: If the incoming AOR in the REGISTER request uses a ‘tel:’ URI or
‘user=phone’, the binding is done according to the Request-URI user part only.
Otherwise, the binding is done according to the entire Request-URI (i.e., user and
host parts - user@host).
• 1-User Part Only: Binding is done according to the user part only.
You must select 1-User Part Only in cases where the UA sends REGISTER
messages as SIP URI, but the INVITE messages sent to this UA include a Tel URI.
For example, when the AOR of an incoming REGISTER is sip:[email protected],
SAS adds the entire SIP URI (e.g., sip:[email protected]) to its database (when the
parameter is set to '0-URI'). However, if a subsequent Request-URI of an INVITE
message for this UA arrives with sip:[email protected] user=phone, SAS searches its
database for "3200", which it does not find. Alternatively, when this parameter is set to
'1-User Part Only', then upon receiving a REGISTER message with
sip:[email protected], SAS adds only the user part (i.e., "3200") to its database.
Therefore, if a Request-URI of an INVITE message for this UA arrives with
sip:[email protected] user=phone, SAS can successfully locate the UA in its database.
Figure 28-1: Configuring Common Settings

6. In the 'SAS Proxy Set' field, enter the Proxy Set used for SAS. The SAS Proxy Set
must be defined only for the following SAS modes:
• Outbound mode: In SAS normal state, SAS forwards REGISTER and INVITE
messages received from the UAs to the proxy servers defined in this Proxy Set.
• Redundant mode and only if UAs don't support homing: SAS sends keep-
alive messages to this proxy and if it detects that the proxy connection has
resumed, it ignores the REGISTER messages received from the UAs, forcing
them to send their messages directly to the proxy.
If you define a SAS Proxy Set ID, you must configure the Proxy Set as described in
Step 8 below.
8. If you defined a SAS Proxy Set ID in Step 6 above, then you must configure the SAS
Proxy Set ID:
a. Open the Proxy Sets Table page (Configuration tab > VoIP menu > Control
Networks > Proxy Set Table).
b. From the 'Proxy Set ID' drop-down list, select the required Proxy Set ID.
Notes:
• The selected Proxy Set ID number must be the same as that specified in the 'SAS
Proxy Set' field in the 'SAS Configuration page (see Step 6).
• Do not use Proxy Set ID 0.
a. In the 'Proxy Address' field, enter the IP address of the external proxy server.
b. From the 'Enable Proxy Keep Alive' drop-down list, select Using Options. This
instructs the device to send SIP OPTIONS messages to the proxy for the keep-
alive mechanism.
Figure 28-2: Defining SAS Proxy Server
c. Click Submit to apply your settings.
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28.2 Configuring SAS Outbound Mode

This section describes how to configure the SAS outbound mode. These settings are in
addition to the ones described in 'Configuring Common SAS Parameters' on page 337.
Note: The VoIP CPEs (such as IP phones or residential gateways) need to be

defined so that their proxy and registrar destination addresses and ports are the same
as that configured for the device's SAS IP address and SAS local SIP port. In some
cases, on the UAs, it is also required to define SAS as their outbound proxy, meaning
that messages sent by the UAs include the host part of the external proxy, but are
sent (on Layer 3/4) to the IP address / UDP port of SAS.
 To configure SAS outbound mode:

Configuration).
2. From the 'SAS Survivability Mode' drop-down list, select Standard.
3. Click Submit.
28.3 Configuring SAS Redundant Mode

This section describes how to configure the SAS redundant mode. These settings are in
addition to the ones described in 'Configuring Common SAS Parameters' on page 337.
Note: The VoIP CPEs (such as IP phones or residential gateways) need to be

defined so that their primary proxy is the external proxy, and their redundant proxy
destination addresses and port is the same as that configured for the device's SAS IP
address and SAS SIP port.
 To configure SAS redundant mode:

Configuration).
2. From the 'SAS Survivability Mode' drop-down list, select one of the following,
depending on whether the UAs support homing (i.e., they always attempt to operate
with the primary proxy, and if using the redundant proxy, they switch back to the
primary proxy whenever it's available):
• UAs support homing: Select Always Emergency. This is because SAS does
not need to communicate with the primary proxy of the UAs; SAS serves only as
the redundant proxy of the UAs. When the UAs detect that their primary proxy is
available, they automatically resume communication with it instead of with SAS.
• UAs do not support homing: Select Ignore REGISTER. SAS uses the keep-
alive mechanism to detect availability of the primary proxy (defined by the SAS
Proxy Set). If the connection with the primary proxy resumes, SAS ignores the
messages received from the UAs, forcing them to send their messages directly to
the primary proxy.
3. Click Submit.

28.4 Configuring Gateway Application with SAS

If you want to run both the Gateway and SAS applications on the device, the configuration
described in this section is required. The configuration steps depend on whether the
Gateway application is operating with SAS in outbound mode or SAS in redundant mode.
Note: The Gateway application must use the same SAS operation mode as the SIP
UAs. For example, if the UAs use the SAS application as a redundant proxy (i.e., SAS
redundancy mode), then the Gateway application must do the same.
28.4.1 Gateway with SAS Outbound Mode

The procedure below describes how to configure the Gateway application with SAS
outbound mode.
 To configure Gateway application with SAS outbound mode:

1. Define the proxy server address for the Gateway application:
a. Open the Proxy & Registration page (Configuration tab > VoIP menu > SIP
b. From the 'Use Default Proxy' drop-down list, select Yes.
Figure 28-3: Enabling Proxy Server for Gateway Application
c. Click Submit.
d. Open the Proxy Sets Table page (Configuration tab > VoIP menu > Control
e. From the 'Proxy Set ID' drop-down list, select 0.
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f. In the first 'Proxy Address' field, enter the IP address and port of the device (in
the format x.x.x.x:port). This is the port as defined in the 'SAS Local
UDP/TCP/TLS Port' field (see 'Configuring Common SAS Parameters' on page
337).
Figure 28-4: Defining Proxy Server for Gateway Application
g. Click Submit.
2. Disable use of user=phone in SIP URL:
a. Open the SIP General Parameters page (Configuration tab > VoIP menu > SIP

b. From the 'Use user=phone in SIP URL' drop-down list, select No. This instructs
the Gateway application not to use user=phone in the SIP URL and therefore,
REGISTER and INVITE messages use SIP URI. (By default, REGISTER
messages are sent with sip uri and INVITE messages with tel uri.)
Figure 28-5: Disabling user=phone in SIP URL
c. Click Submit.
28.4.2 Gateway with SAS Redundant Mode

The procedure below describes how to configure the Gateway application with SAS
redundant mode.
 To configure Gateway application with SAS redundant mode:

1. Define the proxy servers for the Gateway application:
a. Open the Proxy & Registration page (Configuration tab > VoIP menu > SIP
b. From the 'Use Default Proxy' drop-down list, select Yes.
Figure 28-6: Enabling Proxy Server for Gateway Application
c. Click Submit.
d. Open the Proxy Sets Table page (Configuration tab > VoIP menu > Control
e. From the 'Proxy Set ID' drop-down list, select 0.
f. In the first 'Proxy Address' field, enter the IP address of the external proxy server.
g. In the second 'Proxy Address' field, enter the IP address and port of the device (in
the format x.x.x.x:port). This is the same port as defined in the 'SAS Local
UDP/TCP/TLS Port' field (see 'Configuring Common SAS Parameters' on page
337).
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h. From the 'Proxy Redundancy Mode' drop-down list, select Homing.

Figure 28-7: Defining Proxy Servers for Gateway Application
i. Click Submit.
2. Disable the use of user=phone in the SIP URL:
a. Open the SIP General Parameters page (Configuration tab > VoIP menu > SIP
b. From the 'Use user=phone in SIP URL' drop-down list, select No. This instructs
the Gateway application not to use user=phone in SIP URL and therefore,
REGISTER and INVITE messages use SIP URI. (By default, REGISTER
messages are sent with sip uri and INVITE messages with tel uri.)
c. Click Submit.

28.5 Advanced SAS Configuration

This section describes the configuration of advanced SAS features that can optionally be
implemented in your SAS deployment.
28.5.1 Manipulating URI user part of Incoming REGISTER and/or INVITE

There are scenarios in which the UAs register to the proxy server with their full phone
number (for example, "976653434"), but can receive two types of INVITE messages (calls):
 INVITEs whose destination is the UAs' full number (when the call arrives from outside
the enterprise)
 INVITES whose destination is the last four digits of the UAs' phone number ("3434" in
our example) when it is an internal call within the enterprise
Therefore, it is important that the device registers the UAs in the SAS registered database
with their extension numbers (for example, "3434") in addition to their full numbers. To do
this, you can define a manipulation rule to manipulate the SIP Request-URI user part of the
AOR (in the To header) in incoming REGISTER requests. Once manipulated, it is saved in
this manipulated format in the SAS registered users database in addition to the original
(un-manipulated) AOR. Alternatively, you can register the user with the original contact
details of the REGISTER message, but then configure a manipulation rule to manipulate
the INVITE messages so that the device can locate the correct user in its registration
database.
For example: Assume the following incoming REGISTER message is received and that
you want to register in the SAS database the UA's full number as well as the last four digits
from the right of the SIP URI user part:
Via: SIP/2.0/UDP 10.33.4.226:5050;branch=z9hG4bKac10827
Max-Forwards: 70
CSeq: 1 REGISTER
Contact: <sip: [email protected]:5050>;expires=180
Allow:
REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUB
SCRIBE,UPDATE
Expires: 180
User-Agent: Audiocodes-Sip-Gateway-/v.
Content-Length: 0
After manipulation, SAS registers the user in its database as follows:
 AOR: [email protected]
 Associated AOR: [email protected] (after manipulation, in which only the four digits
from the right of the URI user part are retained)
 Contact: [email protected]
 To manipulate incoming Request-URI user part of REGISTER/INVITE message:

1. Open the SAS Registration Manipulation table (Configuration tab > VoIP menu >
SAS > Registration Manipulation Table).
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3. Configure the rule as required. For a description of the parameters, see the table
below.
Table 28-1: SAS Registration Manipulation Table Parameter Description

[SASRegistrationManipulation_Index] Note: You can configure up to three SAS
Registration Manipulation rules.
Remove From Right Defines the number of digits (e.g., "4") to remove
[SASRegistrationManipulation_RemoveFromRight] from the right side of the user part; all other
digits in the user part are kept.
Leave From Right Defines the number of digits (e.g., "4") to leave
[SASRegistrationManipulation_LeaveFromRight] from the right side of the user part; all other
digits in the user part are removed.
Rule Apply To Defines the type of SIP message you want to
[SASRegistrationManipulation_RuleApplyTo] manipulate.
 [0] Register and Invite (default)
 [1] Register Only
 [2] Invite Only
Note: The device first does manipulation according to the 'Remove From Right'
parameter and only then according to the 'Leave From Right' parameter.
28.5.2 Manipulating Destination Number of Incoming INVITE

You can define a manipulation rule to manipulate the destination number in the Request-
URI of incoming INVITE messages when SAS is in emergency state. This is required, for
example, if the call is destined to a registered user but the destination number in the
received INVITE is not the number assigned to the registered user in the SAS registration
database. To overcome this and successfully route the call, you can define manipulation
rules to change the INVITE's destination number so that it matches that of the registered
user in the database. This is done using the IP to IP Inbound Manipulation table.
For example, in SAS emergency state, assume an incoming INVITE has a destination
number "7001234" which is destined to a user registered in the SAS database as
"552155551234". In this scenario, the received destination number needs to be
manipulated to the number "552155551234". The outgoing INVITE sent by the device then
also contains this number in the Request-URI user part.

In normal state, the numbers are not manipulated. In this state, SAS searches the number
552155551234 in its database and if found, it sends the INVITE containing this number to
the UA.
 To manipulate the destination number in SAS emergency state:

Configuration).
2. From the 'SAS Inbound Manipulation Mode' (SASInboundManipulationMode) drop-
down list, select Emergency Only.
3. Click Submit; the SAS Inbound Manipulation Mode Table button appears on
the page.
4. Click this button to open the IP to IP Inbound Manipulation page.
5. Add your SAS manipulation rule as required. See the table below for descriptions of
the parameters.
6. Click Submit to save your changes.
Notes:
• The following fields in the IP to IP Inbound Manipulation table are not applicable to
SAS and must be left at their default values:
- 'Additional Manipulation' - default is 0
- 'Manipulation Purpose' - default is Normal
- 'Source IP Group' - default is -1
• The IP to IP Inbound Manipulation table can also be configured using the table ini
file parameter, IPInboundManipulation.
Table 28-2: SAS IP to IP Inbound Manipulation Parameters

Additional Manipulation Determines whether additional SIP URI user part manipulation is
[IPInboundManipulation_IsAd done for the table entry rule listed directly above it.
ditionalManipulation]  [0] No = (Default) Regular manipulation rule (not done in
addition to the rule above it).
 [1] Yes = If the above row entry rule matched the call, consider
this row entry as a match as well and perform the manipulation
specified by this rule.
Note: Additional manipulation can only be done on a different SIP
URI, source or destination, to the rule configured in the row above
as configured by the 'Manipulated URI' parameter (see below).
Manipulation Purpose Defines the purpose of the manipulation:
[IPInboundManipulation_Mani  [0] Normal = (Default) Inbound manipulations affect the routing
pulationPurpose] input and source and/or destination number.
 [1] Routing input only = Inbound manipulations affect the routing
input only, retaining the original source and destination number.
Source IP Group ID Defines the IP Group from where the incoming INVITE is received.
[IPInboundManipulation_SrcI For any IP Group, enter the value "-1".
pGroup]
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Source Username Prefix Defines the prefix of the source SIP URI user name (usually in the
[IPInboundManipulation_SrcU From header).
sernamePrefix] For any prefix, enter the asterisk "*" symbol (default).
Note: The prefix can be a single digit or a range of digits. For
available notations, see 'Dialing Plan Notation for Routing and
Manipulation' on page 473.
Source Host Defines the source SIP URI host name - full name (usually in the
[IPInboundManipulation_SrcH From header). For any host name, enter the asterisk "*" symbol
ost] (default).
Destination Username Prefix Defines the prefix of the destination SIP URI user name (usually in
[IPInboundManipulation_Dest the Request-URI).
UsernamePrefix] For any prefix, enter the asterisk "*" symbol (default).
Note: The prefix can be a single digit or a range of digits. For
available notations, see 'Dialing Plan Notation for Routing and
Manipulation' on page 473.
Destination Host Defines the destination SIP URI host name - full name (usually in
[IPInboundManipulation_Dest the Request URI).
Host] For any host name, enter the asterisk "*" symbol (default).
Request Type Defines the SIP request type to which the manipulation rule is
[IPInboundManipulation_Req applied.
uestType]  [0] All = (Default) All SIP messages.
 [1] INVITE = All SIP messages except REGISTER and
SUBSCRIBE.
 [2] REGISTER = Only REGISTER messages.
 [3] SUBSCRIBE = Only SUBSCRIBE messages.
 [4] INVITE and REGISTER = All SIP messages except
SUBSCRIBE.
 [5] INVITE and SUBSCRIBE = All SIP messages except
REGISTER.
Manipulated URI Determines whether the source or destination SIP URI user part is
[IPInboundManipulation_Mani manipulated.
pulatedURI]  [0] Source = (Default) Manipulation is done on the source SIP
URI user part.
 [1] Destination = Manipulation is done on the destination SIP
URI user part.
Operation Rule (Action)
Remove From Left Defines the number of digits to remove from the left of the user
[IPInboundManipulation_Rem name prefix. For example, if you enter 3 and the user name is
oveFromLeft] "john", the new user name is "n".
Remove From Right Defines the number of digits to remove from the right of the user
[IPInboundManipulation_Rem name prefix. For example, if you enter 3 and the user name is
oveFromRight] "john", the new user name is "j".
Note: If both 'Remove From Right' and 'Leave From Right'
parameters are configured, the 'Remove From Right' setting is
applied first.

Leave From Right Defines the number of characters that you want retained from the
[IPInboundManipulation_Leav right of the user name.
eFromRight] Note: If both 'Remove From Right' and 'Leave From Right'
parameters are configured, the 'Remove From Right' setting is
applied first.
Prefix to Add Defines the number or string that you want added to the front of the
[IPInboundManipulation_Prefi user name. For example, if you enter 'user' and the user name is
x2Add] "john", the new user name is "userjohn".
Suffix to Add Defines the number or string that you want added to the end of the
[IPInboundManipulation_Suffi user name. For example, if you enter '01' and the user name is
x2Add] "john", the new user name is "john01".
28.5.3 SAS Routing Based on IP-to-IP Routing Table

SAS routing that is based on SAS Routing table rules is applicable for the following SAS
states:
 Normal, if the 'SAS Survivability Mode' parameter is set to Use Routing Table only in
Normal mode.
 Emergency,, if the 'SAS Survivability Mode' parameter is not set to Use Routing
Table only in Normal mode.
The SAS routing rule destination can be an IP Group, IP address, Request-URI, or ENUM
query.
The IP-to-IP Routing Table page allows you to configure up to 120 SAS routing rules (for
Normal and Emergency modes). The device routes the SAS call (received SIP INVITE
message) once a rule in this table is matched. If the characteristics of an incoming call do
not match the first rule, the call characteristics is then compared to the settings of the
second rule, and so on until a matching rule is located. If no rule is matched, the call is
rejected.
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When SAS receives a SIP INVITE request from a proxy server, the following routing logic
is performed:
a. Sends the request according to rules configured in the IP-to-IP Routing table.
b. If no matching routing rule exists, the device sends the request according to its SAS
registration database.
c. If no routing rule is located in the database, the device sends the request according to
the Request-URI header.
Note: The IP-to-IP Routing table can also be configured using the table ini file
parameter, IP2IPRouting (see 'Configuration Parameters Reference' on page 475).
 To configure the IP-to-IP Routing table for SAS:

1. Open the IP-to-IP Routing Table (Configuration tab > VoIP menu > SAS > IP-to-IP
Routing Table).
2. Click Add; the Add Record dialog box appears:
Figure 28-8: Add Record Dialog Box of SAS IP2IP Routing Page
3. Configure the rule according to the table below.

Note: The following parameters are not applicable to SAS and must be ignored:
• 'Source IP Group ID'
• 'Destination IP Group ID'
• 'Destination SRD ID'
• 'Alternative Route Options'

Table 28-3: SAS IP-to-IP Routing Table Parameters
Matching Characteristics
Source Username Prefix Defines the prefix of the user part of the incoming SIP dialog's
[IP2IPRouting_SrcUsernamePr source URI (usually the From URI). You can use special
efix] notations for denoting the prefix. For example, to denote any
prefix, use the asterisk (*) symbol; to denote calls without a user
part in the URI, use the $ sign. For available notations, see
'Dialing Plan Notation for Routing and Manipulation' on page 473.
The default is * (i.e., any prefix).
Source Host Defines the host part of the incoming SIP dialog's source URI
[IP2IPRouting_SrcHost] (usually the From URI). If this rule is not required, leave the field
empty. To denote any host name, use the asterisk (*) symbol
(default).
Destination Username Prefix Defines the prefix of the incoming SIP dialog's destination URI
[IP2IPRouting_DestUsernameP (usually the Request URI) user part. You can use special
refix] notations for denoting the prefix. For example, to denote any
prefix, use the asterisk (*) symbol; to denote calls without a user
part in the URI, use the $ sign. For available notations, see
'Dialing Plan Notation for Routing and Manipulation' on page 473.
The default is * (i.e., any prefix).
Destination Host Defines the host part of the incoming SIP dialog’s destination URI
[IP2IPRouting_DestHost] (usually the Request-URI). If this rule is not required, leave the
field empty. The asterisk (*) symbol (default) can be used to
denote any destination host.
Message Condition Selects a Message Condition rule. To configure Message
[IP2IPRouting_MessageConditi Condition rules, see Configuring Condition Rules.
on]
ReRoute IP Group ID Defines the IP Group that initiated (sent) the SIP redirect
[IP2IPRouting_ReRouteIPGrou response (e.g., 3xx) or REFER message. This field is typically
pID] used for re-routing requests (e.g., INVITEs) when interworking is
required for SIP 3xx redirect responses or REFER messages (for
more information, see Interworking SIP 3xx Redirect Responses
and Interworking SIP REFER Messages, respectively). This
parameter functions together with the 'Call Trigger' field (see
below).
The default is -1 (i.e., not configured).
Call Trigger Defines the reason (i.e, trigger) for re-routing the SIP request:
[IP2IPRouting_Trigger]  [0] Any = (Default) This routing rule is used for all scenarios
(re-routes and non-re-routes).
 [1] 3xx = Re-routes the request if it was triggered as a result of
a SIP 3xx response.
 [2] REFER = Re-routes the INVITE if it was triggered as a
result of a REFER request.
 [3] 3xx or REFER = Applies to options [1] and [2].
 [4] Initial only = This routing rule is used for regular requests
that the device forwards to the destination. This rule is not
used for re-routing of requests triggered by the receipt of
REFER or 3xx.
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Operation Routing Rule

Destination Type Determines the destination type to which the outgoing SIP dialog
[IP2IPRouting_DestType] is sent.
 [0] IP Group = (Default) The SIP dialog is sent to the IP
Group’s Proxy Set (SERVER-type IP Group) or registered
contact from the database (if USER-type IP Group).
 [1] Dest Address = The SIP dialog is sent to the address
configured in the following fields: 'Destination SRD ID',
'Destination Address', 'Destination Port', and 'Destination
Transport Type'.
 [2] Request URI = The SIP dialog is sent to the address
indicated in the incoming Request-URI. If the fields
'Destination Port' and 'Destination Transport Type' are
configured, the incoming Request-URI parameters are
overridden and these fields take precedence.
 [3] ENUM = An ENUM query is sent to include the destination
address. If the fields 'Destination Port' and 'Destination
Transport Type' are configured, the incoming Request-URI
parameters are overridden and these fields take precedence.
 [4] Hunt Group = Used for call center survivability. For more
information, see Call Survivability for Call Centers.
 [5] Dial Plan = The IP destination is determined by a Dial Plan
index of the loaded Dial Plan file. The syntax of the Dial Plan
index in the Dial Plan file is as follows: <destination / called
prefix number>,0,<IP destination>
Note that the second parameter "0" is ignored. An example of
a configured Dial Plan (# 6) in the Dial Plan file is shown
below:
[ PLAN6 ]
200,0,10.33.8.52 ; called prefix 200 is
routed to destination 10.33.8.52
201,0,10.33.8.52
300,0,itsp.com ; called prefix 300 is
routed to destination itsp.com
Once the Dial Plan is defined, you need to assign it (0 to 7) to
the routing rule as the destination in the 'Destination Address'
parameter, where "0" denotes [PLAN1], "1" denotes [PLAN2],
and so on.
Destination IP Group ID Defines the IP Group ID to where you want to route the call. The
[IP2IPRouting_DestIPGroupID] SIP dialog messages are sent to the IP address defined for the
Proxy Set associated with this IP Group. If you select an IP
Group, it is unnecessary to configure a destination IP address (in
the 'Destination Address' field). However, if both parameters are
configured, then the IP Group takes precedence.
If the destination IP Group is of USER type, the device searches
for a match between the Request-URI (of the received SIP dialog)
to an AOR registration record in the device's database. The SIP
dialog is then sent to the IP address of the registered contact.
The default is -1.
Notes:
 This parameter is only relevant if the parameter 'Destination
Type' is set to 'IP Group'. However, regardless of the settings
of the parameter 'Destination Type', the IP Group is still used -

only for determining the IP Profile or outgoing SRD. If neither
IP Group nor SRD are defined in this table, the destination
SRD is determined according to the source SRD associated
with the Source IP Group (configured in the IP Group table,
see 'Configuring IP Groups' on page 205). If this table does
not define an IP Group but only an SRD, then the first IP
Group associated with this SRD (in the IP Group table) is
used.
 If the selected destination IP Group ID is type SERVER, the
request is routed according to the IP Group addresses.
 If the selected destination IP Group ID is type USER, the
request is routed according to the IP Group specific database
(i.e., only to registered users of the selected database).
 If the selected destination IP Group ID is ANY USER ([-2]), the
request is routed according to the general database (i.e., any
matching registered user).
Destination Address Defines the destination IP address (or domain name, e.g.,
[IP2IPRouting_DestAddress] domain.com) to where the call is sent.
If ENUM-based routing is used (i.e., the 'Destination Type'
parameter is set to ENUM) this parameter defines the IP address
or domain name (FQDN) of the ENUM service, for example,
e164.arpa, e164.customer.net, or NRENum.net. The device
sends the ENUM query containing the destination phone number
to an external DNS server, configured in the Multiple Interface
table. The ENUM reply includes a SIP URI (user@host) which is
used as the destination Request-URI in this routing table.
Notes:
 This parameter is applicable only if the parameter 'Destination
Type' is set to 'Dest Address' [1] or ENUM [3].
 When using domain names, enter a DNS server IP address or
alternatively, define these names in the 'Internal DNS Table'
(see 'Configuring the Internal SRV Table' on page 143).
Destination Port Defines the destination port to where the call is sent.
[IP2IPRouting_DestPort]
Destination Transport Type Defines the transport layer type for sending the call:
[IP2IPRouting_DestTransportT  [-1] Not Configured (default)
ype]  [0] UDP
 [1] TCP
 [2] TLS
Note: When this parameter is set to -1, the transport type is
determined by the parameter SIPTransportType.
Cost Group Assigns a Cost Group to the routing rule for determining the cost
[IP2IPRouting_CostGroup] of the call. To configure Cost Groups, see 'Configuring Cost
By default, no Cost Group is assigned to the rule.
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28.5.4 Blocking Calls from Unregistered SAS Users

To prevent malicious calls, for example, service theft, it is recommended to configure the
feature for blocking SIP INVITE messages received from SAS users that are not registered
in the SAS database. This applies to SAS in normal and emergency states.
 To block calls from unregistered SAS users:

Configuration).
2. From the 'SAS Block Unregistered Users' drop-down list, select Block.
28.5.5 Configuring SAS Emergency Calls

You can configure SAS to route emergency calls (such as 911 in North America) directly to
the PSTN through its FXO interface. Thus, even during a communication failure with the
external proxy, enterprise UAs can still make emergency calls.
You can define up to four emergency numbers, where each number can include up to four
digits. When SAS receives a SIP INVITE (from a UA) that includes one of the user-defined
emergency numbers in the SIP user part, it forwards the INVITE directly to the default
gateway (see 'SAS Routing in Emergency State' on page 335). The default gateway is
defined in the 'SAS Default Gateway IP' field, and this is the device itself. The device then
sends the call directly to the PSTN.
This feature is applicable to SAS in normal and emergency states.
 To configure SAS emergency numbers:

Configuration).
2. In the ‘SAS Default Gateway IP' field, define the IP address and port (in the format
x.x.x.x:port) of the device (Gateway application).
Note: The port of the device is defined in the 'SIP UDP/TCP/TLS Local Port' field in
the SIP General Parameters page (Configuration tab > VoIP menu > SIP
Definitions > General Parameters).
3. In the 'SAS Emergency Numbers' field, enter an emergency number in each field box.
Figure 28-9: Configuring SAS Emergency Numbers

28.5.6 Adding SIP Record-Route Header to SIP INVITE

You can configure SAS to add the SIP Record-Route header to SIP requests (e.g. INVITE)
received from enterprise UAs. SAS then sends the request with this header to the proxy.
The Record-Route header includes the IP address of the SAS application. This ensures
that future requests in the SIP dialog session from the proxy to the UAs are routed through
the SAS application. If not configured, future request within the dialog from the proxy are
sent directly to the UAs (and do not traverse SAS). When this feature is enabled, the SIP
Record-Route header includes the URI "lr" parameter, indicating loose routing, as shown in
the following example:
Record-Route: <sip:server10.biloxi.com;lr>
Note: This feature is applicable only to the SAS Outbound mode.
 To enable the Record-Route header:

Configuration).
2. From the ‘Enable Record-Route' drop-down list, select Enable.
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28.5.7 Re-using TCP Connections

You can enable the SAS application to re-use the same TCP connection for sessions
(multiple SIP requests / responses) with the same SIP UA. The benefits of this feature
include less CPU and memory usage because fewer TCP connections are open and
reduced network congestion. For example, assume User A sends a REGISTER message
to SAS with transport=TCP, and User B sends an INVITE message to A using SAS. In this
scenario, the SAS application forwards the INVITE request using the same TCP
connection that User A initially opened with the REGISTER message.
 To re-use TCP connection sessions in SAS

Configuration).
2. From the ‘SAS Connection Reuse' drop-down list, select Enable.
28.5.8 Replacing Contact Header for SIP Messages

You can configure SAS to change the SIP Contact header so that it points to the SAS host.
This ensures that in the message, the top-most SIP Via header and the Contact header
point to the same host.
Notes:
• This feature is applicable only to the SAS Outbound mode.
• The device may become overloaded if this feature is enabled, as all incoming SIP
dialog requests traverse the SAS application.
Currently, this feature can be configured only by the ini file parameter,
SASEnableContactReplace:
 [0] (Default): Disable - when relaying requests, SAS adds a new Via header (with the
IP address of the SAS application) as the top-most Via header and retains the original
Contact header. Thus, the top-most Via header and the Contact header point to
different hosts.
 [1]: Enable - SAS changes the Contact header so that it points to the SAS host and
therefore, the top-most Via header and the Contact header point to the same host.

28.6 Viewing Registered SAS Users

You can view all the users that are registered in the SAS registration database. This is
displayed in the 'SAS/SBC Registered Users page, as described in 'Viewing Registered
Users' on page 418.
Note: You can increase the maximum number of registered SAS users, by
implementing the SAS Cascading feature, as described in 'SAS Cascading' on page
359.
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User's Manual 29. SAS Cascading
29 SAS Cascading
The SAS Cascading feature allows you to increase the number of SAS users above the
maximum supported by the SAS gateway. This is achieved by deploying multiple SAS
gateways in the network. For example, if the SAS gateway supports up to 600 users, but
your enterprise has 1,500 users, you can deploy three SAS gateways to accommodate all
users: the first SAS gateway can service 600 registered users, the second SAS gateway
the next 600 registered users, and the third SAS gateway the rest (i.e., 300 registered
users).
In SAS Cascading, the SAS gateway first attempts to locate the called user in its SAS
registration database. Only if the user is not located, does the SAS gateway send it on to
the next SAS gateway according to the SAS Cascading configuration.
There are two methods for configuring SAS Cascading. This depends on whether the users
can be identified according to their phone extension numbers:
 SAS Routing Table: If users can be identified with unique phone extension numbers,
then the SAS Routing table is used to configure SAS Cascading. This SAS Cascading
method routes calls directly to the SAS Gateway (defined by IP address) to which the
called SAS user is registered.
The following is an example of a SAS Cascading deployment of users with unique
phone extension numbers:
• users registered to the first SAS gateway start with extension number “40”
• users registered to the second SAS gateway start with extension number “20”
• users registered to the third SAS gateway start with extension number “30”
The SAS Routing table rules for SAS Cascading are created using the destination
(called) extension number prefix (e.g., “30”) and the destination IP address of the SAS
gateway to which the called user is registered. Such SAS routing rules must be
configured at each SAS gateway to allow routing between the SAS users. The routing
logic for SAS Cascading is similar to SAS routing in Emergency state (see the
flowchart in 'SAS Routing in Emergency State' on page 335). For a description on the
SAS Routing table, see 'SAS Routing Based on IP-to-IP Routing Table' on page 349.
The figure below illustrates an example of a SAS Cascading call flow configured using
the SAS Routing table. In this example, a call is routed from SAS Gateway (A) user to
a user on SAS Gateway (B).
Figure 29-1: SAS Cascading Using SAS Routing Table - Example
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 SAS Redundancy mode: If users cannot be distinguished (i.e., associated to a

specific SAS gateway), then the SAS Redundancy feature is used to configure SAS
Cascading. This mode routes the call in a loop fashion, from one SAS gateway to the
next, until the user is located. Each SAS gateway serves as the redundant SAS
gateway (“redundant SAS proxy server”) for the previous SAS gateway (in a one-way
direction). For example, if a user calls a user that is not registered on the same SAS
gateway, the call is routed to the second SAS gateway, and if not located, it is sent to
the third SAS gateway. If the called user is not located on the third (or last) SAS
gateway, it is then routed back to the initial SAS gateway, which then routes the call to
the default gateway (i.e., to the PSTN).
Each SAS gateway adds its IP address to the SIP via header in the INVITE message
before sending it to the next (“redundant”) SAS gateway. If the SAS gateway receives
an INVITE and its IP address appears in the SIP via header, it sends it to the default
gateway (and not to the next SAS gateway), as defined by the SASDefaultGatewayIP
parameter. Therefore, this mode of operation prevents looping between SAS
gateways when a user is not located on any of the SAS gateways.
The figure below illustrates an example of a SAS Cascading call flow when configured
using the SAS Redundancy feature. In this example, a call is initiated from a SAS
Gateway (A) user to a user that is not located on any SAS gateway. The call is
subsequently routed to the PSTN.
Figure 29-2: SAS Cascading Using SAS Redundancy Mode - Example

Part VII
Maintenance
User's Manual 30. Basic Maintenance
30 Basic Maintenance
The Maintenance Actions page allows you to perform the following:
 Reset the device - see 'Resetting the Device' on page 363
 Lock and unlock the device - see 'Locking and Unlocking the Device' on page 365
 Save configuration to the device's flash memory - see 'Saving Configuration' on page
366
 To access the Maintenance Actions page, do one of the following:

 On the toolbar, click the Device Actions button, and then from the drop-down menu,
choose Reset.
 On the Navigation bar, click the Maintenance tab, and then in the Navigation tree,
select the Maintenance menu and choose Maintenance Actions.
Figure 30-1: Maintenance Actions Page
30.1 Resetting the Device

The Maintenance Actions page allows you to remotely reset the device. In addition, before
resetting the device, you can choose the following options:
 Save the device's current configuration to the device's flash memory (non-volatile).
 Perform a graceful shutdown, whereby device reset starts only after a user-defined
time (i.e., timeout) or after no more active traffic exists (the earliest thereof).
Notes:
• Throughout the Web interface, parameters displayed with a lightning symbol
are not applied on-the-fly and require that you reset the device for them to take
effect.
• When you modify parameters that require a device reset, once you click the
Submit button in the relevant page, the toolbar displays "Reset" (see 'Toolbar
Description' on page 42) to indicate that a device reset is required.
• After you reset the device, the Web GUI is displayed in Basic view (see 'Displaying
Navigation Tree in Basic and Full View' on page 43).
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 To reset the device:

1. Open the Maintenance Actions page (see 'Basic Maintenance' on page 363).
2. Under the 'Reset Configuration' group, from the 'Burn To FLASH' drop-down list,
select one of the following options:
• Yes: The device's current configuration is saved (burned) to the flash memory
prior to reset (default).
• No: Resets the device without saving the current configuration to flash (discards
all unsaved modifications).
3. Under the 'Reset Configuration' group, from the 'Graceful Option' drop-down list,
select one of the following options:
• Yes: Reset starts only after the user-defined time in the 'Shutdown Timeout' field
(see Step 4) expires or after no more active traffic exists (the earliest thereof). In
addition, no new traffic is accepted.
• No: Reset starts regardless of traffic, and any existing traffic is terminated at
once.
4. In the 'Shutdown Timeout' field (relevant only if the 'Graceful Option' in the previous
step is set to Yes), enter the time after which the device resets. Note that if no traffic
exists and the time has not yet expired, the device resets.
5. Click the Reset button; a confirmation message box appears, requesting you to
confirm.
Figure 30-2: Reset Confirmation Message Box
6. Click OK to confirm device reset; if the parameter 'Graceful Option' is set to Yes (in
Step 3), the reset is delayed and a screen displaying the number of remaining calls
and time is displayed. When the device begins to reset, a message appears notifying
you of this.
30.2 Remotely Resetting Device using SIP NOTIFY

The device can be remotely reset upon the receipt of a SIP NOTIFY that includes an Event
header set to 'check-sync;reboot=true', as shown in the example below:
NOTIFY sip:<user>@<dsthost> SIP/2.0
To: sip:<user>@<dsthost>
From: sip:sipsak@<srchost>
CSeq: 10 NOTIFY
Call-ID: 1234@<srchost>
Event: check-sync;reboot=true

User's Manual 30. Basic Maintenance
 To enable remote reset upon receipt of SIP NOTIFY:

1. Open the Advanced Parameters page (Configuration tab > VoIP menu > SIP
Definitions > Advanced Parameters).
2. Under the Misc Parameters group, set the 'SIP Remote Rest' parameter to Enable.
3. Click Submit.
Note: This SIP Event header value is proprietary to AudioCodes.
30.3 Locking and Unlocking the Device

The Lock and Unlock option allows you to lock the device so that it doesn't accept any new
calls and maintains only the current calls. This is useful when, for example, you are
uploading new software files to the device and you don't want any traffic to interfere with
the process.
 To lock the device:

2. Under the 'LOCK / UNLOCK' group, from the 'Graceful Option' drop-down list, select
one of the following options:
• Yes: The device is 'locked' only after the user-defined time in the 'Lock Timeout'
field (see Step 3) expires or no more active traffic exists (the earliest thereof). In
addition, no new traffic is accepted.
• No: The device is 'locked' regardless of traffic. Any existing traffic is terminated
immediately.
Note: These options are only available if the current status of the device is in the
Unlock state.
3. In the 'Lock Timeout' field (relevant only if the parameter 'Graceful Option' in the
previous step is set to Yes), enter the time (in seconds) after which the device locks.
Note that if no traffic exists and the time has not yet expired, the device locks.
4. Click the LOCK button; a confirmation message box appears requesting you to
confirm device Lock.
Figure 30-3: Device Lock Confirmation Message Box
5. Click OK to confirm device Lock; if 'Graceful Option' is set to Yes, the lock is delayed
and a screen displaying the number of remaining calls and time is displayed.
Otherwise, the lock process begins immediately. The Current Admin State' field
displays the current state - "LOCKED" or "UNLOCKED".
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 To unlock the device:

1. Open the Maintenance Actions page (see 'Maintenance Actions' on page 363).
2. Under the 'LOCK / UNLOCK' group, click the UNLOCK button. Unlock starts
immediately and the device accepts new incoming calls.
Note: The Home page's General Information pane displays whether the device is
locked or unlocked (see 'Viewing the Home Page' on page 63).
30.4 Saving Configuration

The Maintenance Actions page allows you to save (burn) the current parameter
configuration (including loaded auxiliary files) to the device's non-volatile memory (i.e.,
flash). The parameter modifications that you make throughout the Web interface's pages
are temporarily saved (to the volatile memory - RAM) when you click the Submit button on
these pages. Parameter settings that are saved only to the device's RAM revert to their
previous settings after a hardware/software reset (or power failure). Therefore, to ensure
that your configuration changes are retained, you must save them to the device's flash
memory using the burn option described below.
 To save the changes to the non-volatile flash memory :

2. Under the 'Save Configuration' group, click the BURN button; a confirmation message
appears when the configuration successfully saves.
Notes:
• Saving configuration to the non-volatile memory may disrupt current traffic on the
device. To avoid this, disable all new traffic before saving, by performing a
graceful lock (see 'Locking and Unlocking the Device' on page 365).
• Throughout the Web interface, parameters displayed with the lightning symbol
are not applied on-the-fly and require that you reset the device for them to take
effect (see 'Resetting the Device' on page 363).
• The Home page's General Information pane displays whether the device is
currently "burning" the configuration (see 'Viewing the Home Page' on page 63).

User's Manual 31. Resetting an Analog Channel
31 Resetting an Analog Channel

You can inactivate (reset) an FXO or FXS analog channel. This is sometimes useful, for
example, when the device (FXO) is connected to a PBX and the communication between
the two can't be disconnected (e.g., when using reverse polarity). This is done in the Web
interface's Home page.
 To reset an analog channel:

1. Open the Home page.
2. Click the required FXS or FXO port icon; a shortcut menu appears.
3. From the shortcut menu, choose Reset Channel; the channel is changed to inactive
and the port icon is displayed in gray.
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User's Manual 32. Software Upgrade
32 Software Upgrade
The Software Update menu allows you do the following:
 Load Auxiliary Files (see 'Loading Auxiliary Files' on page 369)
 Load Software License Key (see 'Software License Key' on page 381)
 Upgrade device using Software Upgrade Wizard (see 'Software Upgrade Wizard' on
page 385)
 Load / save Configuration File (see 'Backing Up and Loading Configuration File' on
page 388)
32.1 Loading Auxiliary Files

Various Auxiliary files can be installed on the device. These Auxiliary files provide the
device with additional configuration settings. The table below lists the different types of
Auxiliary files:
Table 32-1: Auxiliary Files
File Description
INI Configures the device. The Web interface enables practically full device
provisioning. However, some features may only be configured by ini file or you
may wish to configure your device using the ini file. For more information on
using the ini file to configure the device, see 'INI File-Based Management' on
page 95.
Call Progress Region-specific, telephone exchange-dependent file that contains the Call
Tones Progress Tones (CPT) levels and frequencies for the device. The default CPT
file is U.S.A. For more information, see 'Call Progress Tones File' on page 371.
Prerecorded The Prerecorded Tones (PRT) file enhances the device's capabilities of playing
Tones a wide range of telephone exchange tones that cannot be defined in the CPT
file. For more information, see Prerecorded Tones File on page 375.
Note: PRT is not supported by MP-124 Rev. E.
Dial Plan Provides dialing plans, for example, to know when to stop collecting dialed digits
and start forwarding them or for obtaining the destination IP address for
outbound IP routing. For more information, see 'Dial Plan File' on page 376.
User Info The User Information file maps PBX extensions to IP numbers. This file can be
used to represent PBX extensions as IP phones in the global 'IP world'. For
more information, see 'User Information File' on page 379.
The Auxiliary files can be loaded to the device using one of the following methods:
 Web interface.
 TFTP: This is done by specifying the name of the Auxiliary file in an ini file (see
Auxiliary and Configuration Files Parameters) and then loading the ini file to the
device. The Auxiliary files listed in the ini file are then automatically loaded through
TFTP during device startup. If the ini file does not contain a specific auxiliary file type,
the device uses the last auxiliary file of that type that was stored on its non-volatile
memory.
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Notes:
• You can schedule automatic loading of updated auxiliary files using HTTP/HTTPS,
FTP, or NFS. For more information on automatic updates, ee 'Automatic Update'
on page 389.
• When loading an ini file using this Web page, parameters that are excluded from
the loaded ini file retain their current settings (incremental).
• Saving an auxiliary file to flash memory may disrupt traffic on the device. To avoid
this, disable all traffic on the device by performing a graceful lock as described in
'Locking and Unlocking the Device' on page 365.
• For deleting auxiliary files, see 'Viewing Device Information' on page 411.
The procedure below describes how to load Auxiliary files using the Web interface.
 To load auxiliary files to the device using the Web interface:

1. Open the Load Auxiliary Files page (Maintenance tab > Software Update menu >
Load Auxiliary Files).
Note: The appearance of certain file load fields depends on the installed Software
License Key.
2. Click the Browse button corresponding to the file type that you want to load, navigate
to the folder in which the file is located, and then click Open; the name and path of the
file appear in the field next to the Browse button.
3. Click the Load File button corresponding to the file you want to load.
4. Repeat steps 2 through 3 for each file you want to load.
5. Save the loaded auxiliary files to flash memory, see 'Saving Configuration' on page
366 and reset the device (if you have loaded a Call Progress Tones file), see
'Resetting the Device' on page 363.
You can also load auxiliary files using an ini file that is loaded to the device with BootP.
Each auxiliary file has a specific ini file parameter that specifies the name of the auxiliary
file that you want to load to the device with the ini file. For a description of these ini file
parameters, see Auxiliary and Configuration Files Parameters.

 To load auxiliary files using an ini file:

1. In the ini file, define the auxiliary files to be loaded to the device. You can also define
in the ini file whether the loaded files must be stored in the non-volatile memory so
that the TFTP process is not required every time the device boots up.
2. Save the auxiliary files and the ini file in the same directory on your local PC.
3. Invoke a BootP/TFTP session; the ini and associated auxiliary files are loaded to the
device.
32.1.1 Call Progress Tones File

The Call Progress Tones (CPT) and Distinctive Ringing auxiliary file is comprised of two
sections:
 The first section contains the definitions of the Call Progress Tones (levels and
frequencies) that are detected / generated by the device.
 The second section contains the characteristics of the Distinctive Ringing signals that
are generated by the device (see Distinctive Ringing on page 373).
You can use one of the supplied auxiliary files (.dat file format) or create your own file. To
create your own file, it's recommended to modify the supplied usa_tone.ini file (in any
standard text editor) to suit your specific requirements and then convert the modified ini file
into binary format, using AudioCodes DConvert utility. For a description on converting a
CPT ini file into a binary dat file, refer to the DConvert Utility User's Guide.
Note: Only the dat file format can be loaded to the device.
You can create up to 32 different Call Progress Tones, each with frequency and format
attributes. The frequency attribute can be single or dual-frequency (in the range of 300 to
1980 Hz) or an Amplitude Modulated (AM). Up to 64 different frequencies are supported.
Only eight AM tones, in the range of 1 to 128 kHz, can be configured (the detection range
is limited to 1 to 50 kHz). Note that when a tone is composed of a single frequency, the
second frequency field must be set to zero.
The format attribute can be one of the following:
 Continuous: A steady non-interrupted sound (e.g., a dial tone). Only the 'First Signal
On time' should be specified. All other on and off periods must be set to zero. In this
case, the parameter specifies the detection period. For example, if it equals 300, the
tone is detected after 3 seconds (300 x 10 msec). The minimum detection time is 100
msec.
 Cadence: A repeating sequence of on and off sounds. Up to four different sets of
on/off periods can be specified.
 Burst: A single sound followed by silence. Only the 'First Signal On time' and 'First
Signal Off time' should be specified. All other on and off periods must be set to zero.
The burst tone is detected after the off time is completed.
You can specify several tones of the same type. These additional tones are used only for
tone detection. Generation of a specific tone conforms to the first definition of the specific
tone. For example, you can define an additional dial tone by appending the second dial
tone's definition lines to the first tone definition in the ini file. The device reports dial tone
detection if either of the two tones is detected.
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The Call Progress Tones section of the ini file comprises the following segments:
 [NUMBER OF CALL PROGRESS TONES]: Contains the following key:
'Number of Call Progress Tones' defining the number of Call Progress Tones that are
defined in the file.
 [CALL PROGRESS TONE #X]: containing the Xth tone definition, starting from 0 and
not exceeding the number of Call Progress Tones less 1 defined in the first section
(e.g., if 10 tones, then it is 0 to 9), using the following keys:
• Tone Type: Call Progress Tone types:
♦ [1] Dial Tone
♦ [2] Ringback Tone
♦ [3] Busy Tone
♦ [4] Congestion Tone
♦ [6] Warning Tone
♦ [7] Reorder Tone
♦ [8] Confirmation Tone
♦ [9] Call Waiting Tone - heard by the called party
♦ [15] Stutter Dial Tone
♦ [16] Off Hook Warning Tone
♦ [17] Call Waiting Ringback Tone - heard by the calling party
♦ [18] Comfort Tone
♦ [23] Hold Tone
♦ [46] Beep Tone
• Tone Modulation Type: Amplitude Modulated (1) or regular (0)
• Tone Form: The tone's format can be one of the following:
♦ Continuous (1)
♦ Cadence (2)
♦ Burst (3)
• Low Freq [Hz]: Frequency (in Hz) of the lower tone component in case of dual
frequency tone, or the frequency of the tone in case of single tone. This is not
relevant to AM tones.
• High Freq [Hz: Frequency (in Hz) of the higher tone component in case of dual
frequency tone, or zero (0) in case of single tone (not relevant to AM tones).
• Low Freq Level [-dBm]: Generation level 0 dBm to -31 dBm in dBm (not
relevant to AM tones).
• High Freq Level: Generation level of 0 to -31 dBm. The value should be set to
32 in the case of a single tone (not relevant to AM tones).
• First Signal On Time [10 msec]: 'Signal On' period (in 10 msec units) for the
first cadence on-off cycle. For continuous tones, this parameter defines the
detection period. For burst tones, it defines the tone's duration.
• First Signal Off Time [10 msec]: 'Signal Off' period (in 10 msec units) for the
first cadence on-off cycle (for cadence tones). For burst tones, this parameter
defines the off time required after the burst tone ends and the tone detection is
reported. For continuous tones, this parameter is ignored.
• Second Signal On Time [10 msec]: 'Signal On' period (in 10 msec units) for the
second cadence on-off cycle. Can be omitted if there isn't a second cadence.
• Second Signal Off Time [10 msec]: 'Signal Off' period (in 10 msec units) for the
second cadence on-off cycle. Can be omitted if there isn't a second cadence.
• Third Signal On Time [10 msec]: 'Signal On' period (in 10 msec units) for the
third cadence on-off cycle. Can be omitted if there isn't a third cadence.

• Third Signal Off Time [10 msec]: 'Signal Off' period (in 10 msec units) for the
third cadence on-off cycle. Can be omitted if there isn't a third cadence.
• Fourth Signal On Time [10 msec]: 'Signal On' period (in 10 msec units) for the
fourth cadence on-off cycle. Can be omitted if there isn't a fourth cadence.
• Fourth Signal Off Time [10 msec]: 'Signal Off' period (in 10 msec units) for the
fourth cadence on-off cycle. Can be omitted if there isn't a fourth cadence.
• Carrier Freq [Hz]: Frequency of the carrier signal for AM tones.
• Modulation Freq [Hz]: Frequency of the modulated signal for AM tones (valid
range from 1 to 128 Hz).
• Signal Level [-dBm]: Level of the tone for AM tones.
• AM Factor [steps of 0.02]: Amplitude modulation factor (valid range from 1 to
50). Recommended values from 10 to 25.
Notes:
• When the same frequency is used for a continuous tone and a cadence tone, the
'Signal On Time' parameter of the continuous tone must have a value that is
greater than the 'Signal On Time' parameter of the cadence tone. Otherwise, the
continuous tone is detected instead of the cadence tone.
• The tones frequency must differ by at least 40 Hz between defined tones.
For example, to configure the dial tone to 440 Hz only, enter the following text:
[NUMBER OF CALL PROGRESS TONES]
Number of Call Progress Tones=1
#Dial Tone
[CALL PROGRESS TONE #0]
Tone Type=1
Tone Form =1 (continuous)
Low Freq [Hz]=440
High Freq [Hz]=0
Low Freq Level [-dBm]=10 (-10 dBm)
High Freq Level [-dBm]=32 (use 32 only if a single tone is
required)
First Signal On Time [10msec]=300; the dial tone is detected after
3 sec
First Signal Off Time [10msec]=0
Second Signal On Time [10msec]=0
Second Signal Off Time [10msec]=0
32.1.1.1 Distinctive Ringing

Distinctive Ringing is applicable only to FXS interfaces. Using the Distinctive Ringing
section of the Call Progress Tones auxiliary file, you can create up to 16 Distinctive Ringing
patterns. Each ringing pattern configures the ringing tone frequency and up to four ringing
cadences. The same ringing frequency is used for all the ringing pattern cadences. The
ringing frequency can be configured in the range of 10 to 200 Hz with a 5 Hz resolution.
Each of the ringing pattern cadences is specified by the following parameters:
 Burst Ring On Time: Configures the cadence to be a burst cadence in the entire
ringing pattern. The burst relates to On time and the Off time of the same cadence. It
must appear between 'First/Second/Third/Fourth' string and the 'Ring On/Off Time'
This cadence rings once during the ringing pattern. Otherwise, the cadence is
interpreted as cyclic: it repeats for every ringing cycle.
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 Ring On Time: Specifies the duration of the ringing signal.

 Ring Off Time: Specifies the silence period of the cadence.
The Distinctive Ringing section of the ini file format contains the following strings:
 [NUMBER OF DISTINCTIVE RINGING PATTERNS]: Contains the following key:
• 'Number of Distinctive Ringing Patterns' defining the number of Distinctive
Ringing signals that are defined in the file.
 [Ringing Pattern #X]: Contains the Xth ringing pattern definition (starting from 0 and
not exceeding the number of Distinctive Ringing patterns defined in the first section
minus 1) using the following keys:
• Ring Type: Must be equal to the Ringing Pattern number.
• Freq [Hz]: Frequency in hertz of the ringing tone.
• First (Burst) Ring On Time [10 msec]: 'Ring On' period (in 10 msec units) for
the first cadence on-off cycle.
• First (Burst) Ring Off Time [10 msec]: 'Ring Off' period (in 10 msec units) for
the first cadence on-off cycle.
• Second (Burst) Ring On Time [10 msec]: 'Ring On' period (in 10 msec units)
for the second cadence on-off cycle.
• Second (Burst) Ring Off Time [10 msec]: 'Ring Off' period (in 10 msec units)
for the second cadence on-off cycle.
• Third (Burst) Ring On Time [10 msec]: 'Ring On' period (in 10 msec units) for
the third cadence on-off cycle.
• Third (Burst) Ring Off Time [10 msec]: 'Ring Off' period (in 10 msec units) for
the third cadence on-off cycle.
• Fourth (Burst) Ring On Time [10 msec]: 'Ring Off' period (in 10 msec units) for
the fourth cadence on-off cycle.
• Fourth (Burst) Ring Off Time [10 msec]: 'Ring Off' period (in 10 msec units) for
the fourth cadence on-off cycle.
Note: In SIP, the Distinctive Ringing pattern is selected according to the Alert-Info
header in the INVITE message. For example:
Alert-Info:<Bellcore-dr2>, or Alert-Info:<http://…/Bellcore-dr2>
'dr2' defines ringing pattern #2. If the Alert-Info header is missing, the default ringing
tone (0) is played.
An example of a ringing burst definition is shown below:

#Three ringing bursts followed by repeated ringing of 1 sec on and
3 sec off.
[NUMBER OF DISTINCTIVE RINGING PATTERNS]
Number of Ringing Patterns=1
[Ringing Pattern #0]
Ring Type=0
Freq [Hz]=25
First Burst Ring On Time [10msec]=30
First Burst Ring Off Time [10msec]=30
Second Burst Ring On Time [10msec]=30
Second Burst Ring Off Time [10msec]=30
Third Burst Ring On Time [10msec]=30
Third Burst Ring Off Time [10msec]=30
Fourth Ring On Time [10msec]=100
Fourth Ring Off Time [10msec]=300

An example of various ringing signals definition is shown below:

[NUMBER OF DISTINCTIVE RINGING PATTERNS]
Number of Ringing Patterns=3
#Regular North American Ringing Pattern
Ring Type=0
Freq [Hz]=20
First Ring On Time [10msec]=200
First Ring Off Time [10msec]=400
#GR-506-CORE Ringing Pattern 1
Ring Type=1
Freq [Hz]=20
#GR-506-CORE Ringing Pattern 2
Ring Type=2
Freq [Hz]=20
Second Ring On Time [10msec]=80
Second Ring Off Time [10msec]=400
32.1.2 Prerecorded Tones File

The CPT file mechanism has several limitations such as a limited number of predefined
tones and a limited number of frequency integrations in one tone. To overcome these
limitations and provide tone generation capability that is more flexible, the Prerecorded
Tones (PRT) file can be used. If a specific prerecorded tone exists in the PRT file, it takes
precedence over the same tone that exists in the CPT file and is played instead of it.
Notes
• The PRT are used only for generation of tones. Detection of tones is performed
according to the CPT file.
• PRT is not supported by MP-124 Rev. E.
The PRT is a .dat file containing a set of prerecorded tones that can be played by the
device. Up to 40 tones (totaling approximately 10 minutes) can be stored in a single PRT
file on the device's flash memory. The prerecorded tones are prepared offline using
standard recording utilities (such as CoolEditTM) and combined into a single file, using
AudioCodes DConvert utility (refer to DConvert Utility User's Guide for more information).
The raw data files must be recorded with the following characteristics:
 Coders: G.711 A-law or G.711 µ-law
 Rate: 8 kHz
 Resolution: 8-bit
 Channels: mono
Once created, the PRT file can then be loaded to the device using AudioCodes' AcBootP
utility or the Web interface (see 'Loading Auxiliary Files' on page 369).
The prerecorded tones are played repeatedly. This allows you to record only part of the
tone and then play the tone for the full duration. For example, if a tone has a cadence of 2
seconds on and 4 seconds off, the recorded file should contain only these 6 seconds. The
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PRT module repeatedly plays this cadence for the configured duration. Similarly, a
continuous tone can be played by repeating only part of it.
32.1.3 Dial Plan File

The Dial Plan file can be used for various digit mapping features, as described in this
section.
32.1.3.1 Creating a Dial Plan File

Creating a Dial Plan file is similar between all Dial Plan features. The main difference is the
syntax used in the Dial Plan file and the method for selecting the Dial Plan index to use for
the specific feature.
The Dial Plan file is a text-based file that can contain up to eight Dial Plans (Dial Plan
indices) and up to 8,000 rules (lines). The general syntax rules for the Dial Plan file are as
follows (syntax specific to the feature is described in the respective section):
 Each Dial Plan index must begin with a Dial Plan name enclosed in square brackets
"[...]" on a new line.
 Each line under the Dial Plan index defines a rule.
 Empty lines are ignored.
 Lines beginning with a semicolon ";" are ignored. The semicolon can be used for
comments.
 To create a Dial Plan file:

1. Create a new file using a text-based editor (such as Notepad) and configure your Dial
Plans, as required.
2. Save the file with the ini file extension name (e.g., mydialplanfile.ini).
3. Convert the ini file to a dat binary file, using AudioCodes DConvert utility. For more
information, refer to DConvert Utility User's Guide.
4. Install the converted file on the device, as described in 'Loading Auxiliary Files' on
page 369.
5. Select the Dial Plan index that you want to use. This depends on the feature and is
described in the respective section.
32.1.3.2 Dialing Plans for Digit Collection

The device enables you to configure multiple dialing plans in an external Dial Plan file,
which can be installed on the device. If a Dial Plan file is implemented, the device first
attempts to locate a matching digit pattern in a specified Dial Plan index listed in the file
and if not found, attempts to locate a matching digit pattern in the Digit Map. The Digit Map
is configured by the 'Digit Mapping Rules' parameter, located in the DTMF & Dialing page
(Configuration tab > VoIP menu > GW and IP to IP > DTMF and Supplementary >
DTMF & Dialing).
The Dial Plan is used for the following:
 FXS, and FXO collecting digit mode (Tel-to-IP calls): The file allows the device to
know when digit collection ends, after which it starts sending all the collected (or
dialed) digits in the outgoing INVITE message. This also provides enhanced digit
mapping.
The Dial Plan file can contain up to eight Dial Plans (Dial Plan indices), with a total of up to
8,000 dialing rules (lines) of distinct prefixes (e.g. area codes, international telephone
number patterns) for the PSTN to which the device is connected.
The Dial Plan file is created in a textual ini file with the following syntax:
<called number prefix>,<total digits to wait before sending>

 Each new Dial Plan index begins with a Dial Plan name enclosed in square brackets
"[...]" on a new line.
 Each line under the Dial Plan index defines a dialing prefix and the number of digits
expected to follow that prefix. The prefix is separated by a comma "," from the number
of additional digits.
 The prefix can include numerical ranges in the format [x-y], as well as multiple
numerical ranges [n-m][x-y] (no comma between them).
 The prefix can include the asterisk "*" and number "#" signs.
 The number of additional digits can include a numerical range in the format x-y.
 Empty lines are ignored.
 Lines beginning with a semicolon ";" are ignored. The semicolon can be used for
comments.
Below shows an example of a Dial Plan file (in ini-file format), containing two dial plans:
; Example of dial-plan configuration.
; This file contains two dial plans:
[ PLAN1 ]
; Destination cellular area codes 052, 054, and 050 with 8 digits.
052,8
054,8
050,8
; Defines International prefixes 00, 012, 014.
; The number following these prefixes may
; be 7 to 14 digits in length.
00,7-14
012,7-14
014,7-14
; Defines emergency number 911. No additional digits are expected.
911,0
[ PLAN2 ]
; Defines area codes 02, 03, 04.
; In these area codes, phone numbers have 7 digits.
0[2-4],7
; Operator services starting with a star: *41, *42, *43.
; No additional digits are expected.
*4[1-3],0
The procedure below provides a summary on how to create a Dial Plan file and select the
required Dial Plan index.
 To create a Dial Plan file:

1. Create a new file using a text-based editor (such as Notepad) and configure your Dial
Plans, as required.
2. Save the file with the ini file extension name (e.g., mydialplans.ini).
3. Convert the ini file to a dat binary file, using AudioCodes DConvert utility. For more
information, refer to DConvert Utility User's Guide.
4. Install the converted file on the device, as described in 'Loading Auxiliary Files' on
page 369.
5. The required Dial Plan is selected using the 'Dial Plan Index' parameter. This
parameter can be set to 0 through 7, where 0 denotes PLAN1, 1 denotes PLAN2, and
so on.
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Notes:
• The Dial Plan file must not contain overlapping prefixes. Attempting to process an
overlapping configuration by the DConvert utility results in an error message
specifying the problematic line.
• The Dial Plan index can be selected globally for all calls (as described in the
previous procedure), or per specific calls using Tel Profiles.
• It may be useful to configure both Dial Plan file and Digit Maps. For example, the
Digit Map can be used for complex digit patterns (which are not supported by the
Dial Plan) and the Dial Plan can be used for long lists of relatively simple digit
patterns. In addition, as timeout between digits is not supported by the Dial Plan,
the Digit Map can be used to configure digit patterns that are shorter than those
defined in the Dial Plan or left at default (MaxDigits parameter). For example, the
“xx.T” digit map instructs the device to use the Dial Plan and if no matching digit
pattern is found, it waits for two more digits and then after a timeout
(TimeBetweenDigits parameter), it sends the collected digits. Therefore, this
ensures that calls are not rejected as a result of their digit pattern not been
completed in the Dial Plan.
• By default, if no matching digit pattern is found in both the Dial Plan and Digit Map,
the device rejects the call. However, if you set the DisableStrictDialPlan parameter
to 1, the device attempts to complete the call using the MaxDigits and
TimeBetweenDigits parameters. In such a setup, it collects the number of digits
configured by the MaxDigits parameters. If more digits are received, it ignores the
settings of this parameter and collects the digits until the inter-digit timeout
configured by the TimeBetweenDigits parameter is exceeded.
32.1.3.3 Obtaining IP Destination from Dial Plan File

You can use a Dial Plan index listed in a loaded Dial Plan file for determining the IP
destination of Tel-to-IP /IP-to-IP calls. This enables the mapping of called numbers to IP
addresses (in dotted-decimal notation) or FQDNs (up to 15 characters).
 To configure routing to an IP destination based on Dial Plan:

1. Create the Dial Plan file. The syntax of the Dial Plan index for this feature is as
follows:
<destination / called prefix number>,0,<IP destination>
Note that the second parameter "0" is not used and ignored.
An example of a configured Dial Plan (# 6) in the Dial Plan file is shown below:
[ PLAN6 ]
200,0,10.33.8.52 ; called prefix 200 is routed to
10.33.8.52
201,0,10.33.8.52
300,0,itsp.com ; called prefix 300 is routed to itsp.com
2. Convert the file to a loadable file and then load it to the device.
3. Assign the Dial Plan index to the required routing rule:
a. Open the Outbound IP Routing table.
b. In the 'Destination Address' field, enter the required Dial Plan index using the
following syntax:
DialPlan<index>
Where "DialPlan0" denotes [PLAN1] in the Dial Plan file, "DialPlan1" denotes
[PLAN2], and so on.

Note: The "DialPlan" string is case-sensitive.
32.1.4 User Information File

This section describes the various uses of the User Info file.
You can load the User Info file using any of the following methods:
 Web interface (see 'Loading Auxiliary Files' on page 369)
 ini file - using the UserInfoFileName parameter, e.g., UserInfoFileName =
'UserInformationFile.txt' (see 'Auxiliary and Configuration File Name Parameters' on
page 644)
 Automatic update mechanism - using the UserInfoFileURL parameter, e.g.,
UserInfoFileUrl = 'http://192.168.0.250/Audiocodes/ UserInformationFile.txt' (see
'Automatic Update Mechanism' on page 389)
32.1.4.1 User Information File for PBX Extensions and "Global" Numbers
The User Info file contains a User Info table that can be used for the following Gateway-
related:
 Mapping (Manipulating) PBX Extension Numbers with Global Phone Numbers:
maps PBX extension number, connected to the device, with any "global" phone
number (alphanumerical) for the IP side. In this context, the "global" phone number
serves as a routing identifier for calls in the "IP world" and the PBX extension uses this
mapping to emulate the behavior of an IP phone. This feature is especially useful in
scenarios where unique or non-consecutive number translation per PBX is needed.
This number manipulation feature supports the following call directions:
• IP-to-Tel Calls: Maps the called "global" number (in the Request-URI user part) to
the PBX extension number. For example, if the device receives an IP call
destined for "global" number 638002, it changes this called number to the PBX
extension number 402, and then sends the call to the PBX extension on the Tel
side.
Note: If you have configured regular IP-to-Tel manipulation rules (see 'Configuring
Source/Destination Number Manipulation' on page 241), the device applies these
rules before applying the mapping rules of the User Info table.
• Tel-to-IP Calls: Maps the calling (source) PBX extension to the "global" number.
For example, if the device receives a Tel call from PBX extension 402, it changes
this calling number to 638002, and then sends call to the IP side with this calling
number. In addition to the "global" phone number, the display name (caller ID)
configured for the PBX user in the User Info table is used in the SIP From header.
Note: If you have configured regular Tel-to-IP manipulation rules (see 'Configuring
Source/Destination Number Manipulation' on page 241), the device applies these
rules before applying the mapping rules of the User Info table.
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• IP-to-IP Calls: Maps SIP From (calling number) and To (called number) of IP PBX
extension numbers with "global" numbers. For example, if the device receives a
call from IP PBX extension number 402 (calling / SIP From) that is destined to IP
PBX extension number 403 (called / SIP To), the device changes both these
numbers into their "global" numbers 638002 and 638003, respectively.
 Registering Users: The device can register each PBX user configured in the User
Info table. For each user, the device sends a SIP REGISTER to an external IP-based
Registrar server, using the "global" number in the From/To headers. If authentication
is necessary for registration, the device sends the user's username and password,
configured in the User Info table, in the SIP MD5 Authorization header.
Notes:
• To enable the User Info table, see 'Enabling the User Info Table' on page 381.
• To modify the Use Info table, you need to load a new User Info table containing
your modifications.
• To enable user registration, set the following parameters on the Proxy &
Registration page (Configuration tab > VoIP menu > SIP Definitions > Proxy &
Registration) as shown:
√ 'Enable Registration' parameter set to Enable (IsRegisterNeeded is set to
1).
√ 'Registration Mode' parameter set to Per Endpoint (AuthenticationMode is
set to 0).
• For FXS ports, when the device needs to send a new SIP request with the
Authorization header (e.g., after receiving a SIP 401 response), it uses the
username and password configured in the Authentication table (see 'Configuring
Authentication per Port' on page 304). To use the username and password
configured in the User Info file, set the 'Password' parameter to any value other
than its default value.
The User Info file is a text-based file that you can create using any text-based program
such as Notepad. To add mapping rules to this file, use the following syntax:
[ GW ]
FORMAT
PBXExtensionNum,GlobalPhoneNum,DisplayName,UserName,Password
Where:
 PBXExtensionNum is the PBX extension number (up to 10 characters)
 GlobalPhoneNum is the "global" phone number (up to 20 characters) for the IP side
 DisplayName is the Caller ID (string of up to 30 characters) of the PBX extension
 UserName is the username (string of up to 40 characters) for registering the user
when authentication is necessary
 Password is the password (string of up to 20 characters) for registering the user when
authentication is necessary
Each line in the file represents a mapping rule of a single PBX extension user.
You can add up to 25 mapping rules. The maximum size of the User Info file is 10,800
bytes.
Note:
• Make sure that there are no spaces between the values.
• Make sure that the last line in the User Info file ends with a carriage return (i.e., by
pressing the <Enter> key).

An example of a configured User Info file is shown below:

[ GW ]
FORMAT
PBXExtensionNum,GlobalPhoneNum,DisplayName,UserName,Password
401,638001,Mike,miked,1234
402,638002,Lee,leep,4321
403,638003,Sue,suer,8790
404,638004,John,johnd,7694
405,638005,Pam,pame,3928
406,638006,Steve,steveg,1119
407,638007,Fred,frede,8142
408,638008,Maggie,maggiea,9807
32.1.4.2 Enabling the User Info Table

The procedure below describes how to load a User Info file to the device and enable the
use of the User Info table:
 To enable the User Info table:

Definitions > Advanced Parameters).
2. Set the 'Enable User-Information Usage' parameter to Enable.
32.2 Software License Key

The device is shipped with a pre-installed Software License Key, which determines the
device's supported features, capabilities, and available resources. You can upgrade or
change your device's supported features by purchasing and installing a new Software
License Key to match your requirements.
Note: The availability of certain Web pages depends on the installed Software
License Key.
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32.2.1 Obtaining the Software License Key File

Before you can install a new Software License Key, you need to obtain a Software License
Key file for your device with the required features from your AudioCodes representative.
The Software License Key is an encrypted key in string format that is associated with the
device's serial number ("S/N") and supplied in a text-based file.
If you need a Software License Key for more than one device, the Software License Key
file can include multiple Software License Keys (see figure below). In such cases, each
Software License Key in the file is associated with a unique serial number identifying the
specific device. When loading such a Software License Key file, the device installs only the
Software License Key that is associated with its serial number.
Figure 4: Software License Key File with Multiple S/N Lines
 To obtain a Software License Key:

1. Make a note of the MAC address and/or serial number of the device:
a. Open the Device Information page (Status & Diagnostics tab > System Status
menu > Device Information).
b. The MAC address is displayed in the “MAC Address” field and the serial number
in the “Serial Number” field.
2. If you need a Software License Key for more than one device, repeat Step 1 for each
device.
3. Request the required Software License Key from your AudioCodes representative and
provide them with the MAC address and/or serial number of the device(s).
4. When you receive the new Software License Key file, check the file as follows:
a. Open the file with any text-based program such as Notepad.
b. Verify that the first line displays "[LicenseKeys]".
c. Verify that the file contains one or more lines in the following format:
"S/N<serial number> = <Software License Key string>".
For example: "S/N370604 = jCx6r5tovCIKaBBbhPtT53Yj..."
d. Verify that the "S/N" value reflects the serial number of your device. If you have
multiple Software License Keys, ensure that each "S/N" value corresponds to a
device.
Warning: Do not modify the contents of the Software License Key file.
5. Install the Software License Key on the device as described in 'Installing the Software
License Key' on page 383.

32.2.2 Installing the Software License Key

Once you have received your Software License Key file from your AudioCodes
representative, you can install it on the device using one of the following management
tools:
 Web interface - see 'Installing Software License Key using Web Interface' on page 383
 AudioCodes AcBootP utility - see Installing Software License Key using AcBootP on
page 384
 AudioCodes EMS - refer to the EMS User’s Manual or EMS Product Description
Note: When you install a new Software License Key, it is loaded to the device's non-
volatile flash memory and overwrites the previously installed Software License Key.
32.2.2.1 Installing Software License Key using Web Interface

The procedure below describes how to install the Software License Key using the Web
interface.
 To install the Software License Key using the Web interface:

1. Open the Software Upgrade Key Status page (Maintenance tab > Software Update
menu > Software Upgrade Key).
2. As a precaution, backup the Software License Key currently installed on the device. If
the new Software License Key does not comply with your requirements, you can re-
load this backup to restore the device's original capabilities.
a. In the 'Current Key' field, select the entire text string and copy it to any standard
text file (e.g., Notepad).
b. Save the text file with any file name and file extension (e.g., key.txt) to a folder on
your computer.
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3. Depending on whether you are loading a Software License Key file with a single
Software License Key (i.e., one "S/N") or with multiple Software License Keys (i.e.,
more than one "S/N"), do one of the following:
• Loading a File with a Single Software License Key:
a. Open the Software License Key file using a text-based program such as
Notepad.
b. Copy-and-paste the string from the file to the 'Add a Software Upgrade Key'
field.
c. Click the Add Key button.
• Loading a File with Multiple Software License Keys:
a. In the 'Load Upgrade Key file ...' field, click the Browse button and navigate
to the folder in which the Software License Key file is located on your
computer.
b. Click Load File; the new key is installed on the device.
If the Software License Key is valid, it is burned to the device's flash memory and
displayed in the 'Current Key' field.
4. Verify that the Software License Key was successfully installed, by doing one of the
following:
• In the Software Upgrade Key Status page, check that the listed features and
capabilities activated by the installed Software License Key match those that
were ordered.
• Access the Syslog server and ensure that the following message appears in the
Syslog server:
"S/N___ Key Was Updated. The Board Needs to be Reloaded with ini file\n"
5. Reset the device; the new capabilities and resources enabled by the Software License
Key are active.
Note: If the Syslog server indicates that the Software License Key was
unsuccessfully loaded (i.e., the "SN_" line is blank), do the following preliminary
troubleshooting procedures:
1. Open the Software License Key file and check that the "S/N" line appears. If it
does not appear, contact AudioCodes.
2. Verify that you have loaded the correct file. Open the file and ensure that the first
line displays "[LicenseKeys]".
3. Verify that the content of the file has not been altered.
32.2.2.2 Installing Software License Key using BootP/TFTP

The procedure below describes how to install a Software License Key using AudioCodes
AcBootP utility.
Notes:
• When loading the Software License Key file, a cmp file must also be loaded during
this BootP process.
• For more information on using the AcBootP utility, refer to the document AcBootP
Utility User's Guide.

 To install a Software License Key using the AcBootP utility:

1. Change the file extension name of the Software License Key file from .txt to .ini.
2. Place the Software License Key file in the same folder in which the device's cmp file is
located.
3. Start the AcBootP utility.
4. Click the Client Configuration tab, and then from the 'INI File' drop-down list, select
the Software License Key file.
5. From the 'BootP File' drop-down list, select the device's cmp file.
6. Configure the initial BootP/TFTP parameters as required, and then click Apply.
7. Reset the device; the cmp and Software License Key files are loaded to the device.
32.3 Software Upgrade Wizard

The Software Upgrade Wizard allows you to upgrade the device's firmware. The firmware
file has the .cmp file extension name. The wizard also enables you to load an ini file and/or
auxiliary files (typically loaded using the Load Auxiliary File page described in 'Loading
Auxiliary Files' on page 369). However, it is mandatory when using the wizard to first load a
.cmp file to the device. You can then choose to also load an ini file and/or auxiliary files, but
this cannot be done without first loading a .cmp file. For the ini and each auxiliary file type,
you can choose to load a new file or not load a file but use the existing file (i.e., maintain
existing configuration) running on the device.
Warning: The Software Upgrade Wizard requires the device to be reset at the end of
the process, which may disrupt traffic. To avoid this, disable all traffic on the device
before initiating the wizard by performing a graceful lock (see 'Basic Maintenance' on
page 363).
Notes:
• You can get the latest software files from AudioCodes Web site at
http://www.audiocodes.com/downloads.
• Before upgrading the device, it is recommended that you save a copy of the
device's configuration settings (i.e., ini file) to your computer. If an upgrade failure
occurs, you can then restore your configuration settings by uploading the backup
file to the device. For saving and restoring configuration, see 'Backing Up and
Loading Configuration File' on page 388.
• If you wish to also load an ini or auxiliary file, it is mandatory to first load a .cmp
file.
• When you activate the wizard, the rest of the Web interface is unavailable. After
the files are successfully loaded, access to the full Web interface is restored.
• If you upgraded your .cmp and the "SW version mismatch" message appears in
the Syslog or Web interface, then your Software License Key does not support the
new .cmp file version. If this occurs, contact AudioCodes support for assistance.
• If you use the wizard to load an ini file, parameters excluded from the ini file are
assigned default values (according to the .cmp file running on the device) thereby,
overriding values previously defined for these parameters.
• You can schedule automatic loading of these files using HTTP/HTTPS, FTP, or
NFS (see 'Automatic Update' on page 389).
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 To load files using the Software Upgrade Wizard:

1. Stop all traffic on the device using the Graceful Lock feature (refer to the warning
bulletin above).
2. Open the Software Upgrade wizard, by performing one of the following:
• Select the Maintenance tab, click the Software Update menu, and then click
Software Upgrade Wizard.
• On the toolbar, click Device Actions, and then choose Software Upgrade
Wizard.
Figure 32-5: Start Software Upgrade Wizard Screen
3. Click the Start Software Upgrade button; the wizard starts, requesting you to
browses to a .cmp file for uploading.
Note: At this stage, you can quit the Software Update Wizard, by clicking Cancel
, without requiring a device reset. However, once you start uploading a cmp file,
the process must be completed with a device reset. If you choose to quit the process
in any of the subsequent pages, the device resets.
4. Click the Browse button, navigate to the .cmp file, and then click Load File; a
progress bar appears displaying the status of the loading process. When the .cmp file
is successfully loaded to the device, a message appears notifying you of this.
5. If you want to load only a .cmp file, then click the Reset button to reset the
device with the newly loaded .cmp file, utilizing the existing configuration (ini) and
auxiliary files. To load additional files, skip to the next Step.
Note: Device reset may take a few minutes depending on cmp file version (this may
even take up to 10 minutes).
6. Click the Next button; the wizard page for loading an ini file appears. You can
now perform one of the following:
• Load a new ini file: Click Browse, navigate to the ini file, and then click Send
File; the ini file is loaded to the device and you're notified as to a successful
loading.
• Retain the existing configuration (ini file): Do not select an ini file, and ensure that
the 'Use existing configuration' check box is selected (default).

• Return the device's configuration settings to factory defaults: Do not select an ini
file, and clear the 'Use existing configuration' check box.
7. Click the Next button to progress to the relevant wizard pages for loading the
desired auxiliary files. To return to the previous wizard page, click the Back
button. As you navigate between wizard pages, the relevant file type corresponding to
the Wizard page is highlighted in the left pane.
8. When you have completed loading all the desired files, click the Next button until
the last wizard page appears ("FINISH" is highlighted in the left pane).
9. Click the Reset button to complete the upgrade process; the device 'burns' the
newly loaded files to flash memory and then resets the device.
Note: Device reset may take a few minutes (depending on .cmp file version, this
may even take up to 30 minutes).
After the device resets, the End of Process wizard page appears displaying the new
.cmp and auxiliary files loaded to the device.
Figure 32-6: Software Upgrade Process Completed Successfully
10. Click End Process to close the wizard; the Web Login dialog box appears.
11. Enter your login user name and password, and then click OK; a message box appears
informing you of the new .cmp file.
12. Click OK; the Web interface becomes active, reflecting the upgraded device.
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32.4 Backing Up and Loading Configuration File

You can save a copy/backup of the device's current configuration settings as an ini file to a
folder on your computer, using the Configuration File page. The saved ini file includes only
parameters that were modified and parameters with other than default values. The
Configuration File page also allows you to load an ini file to the device. If the device has
"lost" its configuration, you can restore the device's configuration by loading the previously
saved ini file or by simply loading a newly created ini file.
Notes:
• When loading an ini file using this Web page, parameters not included in the ini file
are reset to default settings.
•
 To save the ini file:

1. Open the Configuration File page by doing one of the following:
• From the Navigation tree, click the Maintenance tab, click the Software Update
menu, and then click Configuration File.
• On the toolbar, click Device Actions, and then from the drop-down menu,
choose Load Configuration File or Save Configuration File.
2. To save the ini file to a folder on your computer, do the following:

a. Click the Save INI File button; the File Download dialog box appears.
b. Click the Save button, navigate to the folder where you want to save the ini file,
and then click Save.
3. To load the ini file to the device, do the following:
a. Click the Browse button, navigate to the folder where the ini file is located, select
the file, and then click Open; the name and path of the file appear in the field
beside the Browse button.
b. Click the Load INI File button, and then at the prompt, click OK; the device
uploads the ini file and then resets (from the cmp version stored on the flash
memory). Once complete, the Web Login screen appears, requesting you to
enter your user name and password.

User's Manual 33. Automatic Update
33 Automatic Update
This chapter describes the device's automatic provisioning mechanisms.
33.1 Automatic Configuration Methods

The device supports the following automatic provisioning methods:
 DHCP (Option 66, Option 67, Option 160)
 HTTP/S
 TFTP
 FTP
 NFS
 SNMP (EMS)
 BootP / TFTP
33.1.1 BootP Request and DHCP Discovery upon Device Initialization

After the device powers up or is physically reset, it broadcasts a BootP request to the
network. If it receives a reply from a BootP server, it changes its network parameters (IP
address, subnet mask and default gateway address) according to the values provided by
the BootP server. If there is no reply from a BootP server and if DHCP is enabled, the
device initiates a standard DHCP procedure to configure its network parameters.
After changing the network parameters, the device attempts to load the firmware file (cmp)
and various configuration files from the TFTP server’s IP address received from the
BootP/DHCP server. If a TFTP server’s IP address is not obtained, the device attempts to
load the cmp file and/or configuration files from a preconfigured TFTP server. Thus, the
device can obtain its network parameters from BootP or DHCP servers, and its software
and configuration files from a different TFTP server (preconfigured in the ini configuration
file).
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If BootP/DHCP servers are not found or when the device is reset using the Web interface
or SNMP, it retains its network parameters and attempts to load the cmp file and/or
configuration files from a preconfigured TFTP server. If a preconfigured TFTP server does
not exist, the device operates using the existing software and configuration files on its flash
memory.
Figure 33-1: BootP Request and DHCP Discovery upon Startup

Note: By default, the duration between BootP/DHCP requests sent by the device is
one second (configured by the BootPDelay ini file parameter). By default, the number
of requests is three (configured by the BootPRetries ini file parameter).
33.1.2 VLAN ID Discovery using LLDP

You can enable the device, through the EnableLLDP ini file parameter, to use the Link
Layer Discovery Protocol (LLDP) discovery protocol to obtain over the Layer-2 data link
layer, a VLAN ID for its OAMP interface (per IEEE 802.1, IEEE 802.3 and TR-41). The
device supports LLDP attributes referred to as TLVs (type, length, and value descriptions),
which are sent and received in LLDP messages. The device is defined as a Media
Endpoint Device (LLDP-MED, Class II), which is an extension of LLDP. The device
supports all TLV's per specification:
 IEEE 802.1 TLV's: Chassis ID; Port ID; Time To Live; End Of LLDPDU.
 Basic management TLV's: Port Description; System Name; System Description;
System Capabilities.
 IEEE 802.3 Organizationally Specific TLV's: MAC/PHY Configuration/Status (indicates
the auto-negotiation capability and the duplex/speed status of IEEE 802.3
MAC/PHYs); Maximum Frame Size (indicates the maximum supported IEEE 802.3
frame size).
 LLDP-MED TLVs: LLDP-MED Capabilities; Network Policy.
Upon startup (reset or powered up), the device sends an LLDP broadcast message
containing its identity to request a VLAN ID from a server in the network. If it receives an
LLDP reply with a VLAN ID from a server within 30 seconds, the device overwrites the
current VLAN ID of its OAMP interface (in the Interface table) with this new VLAN ID. If no
LLDP reply is received within 30 seconds, the device continues with its normal startup
process.
33.1.3 Local Configuration Server with BootP/TFTP

Local configuration server with BootP/TFTP provides an easy and efficient method for
automatic provisioning, where configuration occurs at a staging warehouse, as follows:
1. Install AudioCodes AcBootP/TFTP utility program on a computer located in a staging
warehouse.
2. Prepare a standard configuration ini file and place it in the TFTP directory.
3. Enter the MAC address of each device in the AcBootP utility.
4. For each device added in the BootP utility, select the cmp and ini file in the 'BootP File'
field.
5. Connect each device to the network and then power up the device.
6. The BootP reply contains the cmp and ini file names entered in the 'BootP File' field.
Each device retrieves these files using BootP and stores them in its flash memory. If
Auxiliary files are required (e.g., call progress tones), they may also be specified in the
ini file and downloaded from the same TFTP server.
7. When the devices' LEDs turn green indicating that the files were successfully loaded,
disconnect the devices and ship to the customer.
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Notes:
• Typically, IP addressing at the customer site is done by DHCP.
• For more information on the AcBootP utility, refer to the AcBootP Utility User's
Guide.
33.1.4 DHCP-based Provisioning

This method is similar to the setup described in Local Configuration Server with
BootP/TFTP on page 391, except that DHCP is used instead of BootP. A third-party DHCP
server can be configured to automatically provide each device, acting as a DHCP client,
with a temporary IP address so that individual MAC addresses are not required. The DHCP
server can provide additional networking parameters such as subnet mask, default
gateway, primary and secondary DNS server, and two SIP server addresses. These
network parameters have a time limit, after which the device must 'renew' its lease from the
DHCP server.
The device can use a host name in the DHCP request. The host name is set to acl_nnnnn,
where nnnnn denotes the device's serial number. The serial number is the last six digits of
the MAC address converted to decimal representation. In networks that support this feature
and if the DHCP server registers this host name to a DNS server, you can access the
device (through a Web browser) using the URL, http://acl_<serial number> (instead of
using the device's IP address). For example, if the device's MAC address is 00908f010280,
the DNS name is acl_66176.
Notes:
• Throughout the DHCP procedure, make sure that the BootP/TFTP program
(AcBootP utility) is deactivated; otherwise the device receives a response from the
BootP server instead of the DHCP server. Typically, after the device powers up, it
attempts to communicate with a BootP server. If a BootP server does not respond
and DHCP is enabled, the device attempts to obtain its networking parameters
from the DHCP server.
• When using DHCP to acquire an IP address, the Interface table, VLANs and other
advanced configuration options are disabled.
• For more information on DHCP, see BootP Request and DHCP Discovery upon
Device Initialization on page 389.
• For additional DHCP parameters, see ''DHCP Parameters'' on page 483.
 To enable the device as a DHCP client:

Figure 33-2: Enabling DHCP - Application Settings Page
2. From the 'Enable DHCP" drop-down list, select Enable.

3. Click Submit.
4. To activate the DHCP process, reset the device.
The following shows an example of a configuration file for a Linux DHCP server
(dhcpd.conf). The devices are allocated temporary IP addresses in the range 10.31.4.53 to

10.31.4.75. TFTP is assumed to be on the same computer as the DHCP server

(alternatively, the "next-server" directive may be used).
ddns-update-style ad-hoc;
default-lease-time 60;
max-lease-time 60;
class "gateways" {
match if(substring(hardware, 1, 3) = 00:90:8f);
}
subnet 10.31.0.0 netmask 255.255.0.0 {
pool {
allow members of "audiocodes";
range 10.31.4.53 10.31.4.75;
filename "SIP_F6.60A.217.003.cmp –fb;device.ini";
option routers 10.31.0.1;
option subnet-mask 255.255.0.0;
}
}
Notes:
• If, during operation, the device's IP address is changed as a result of a DHCP
renewal, the device automatically resets.
• If the DHCP server denies the use of the device's current IP address and specifies
a different IP address (according to RFC 1541), the device must change its
networking parameters. If this occurs while calls are in progress, they are not
automatically rerouted to the new network address. Therefore, administrators are
advised to configure DHCP servers to allow renewal of IP addresses.
• If the device's network cable is disconnected and then reconnected, a DHCP
renewal is performed (to verify that the device is still connected to the same
network). The device also includes its product name in the DHCP Option 60
Vendor Class Identifier. The DHCP server can use this product name to assign an
IP address accordingly.
• After power-up, the device performs two distinct DHCP sequences. Only in the
second sequence is DHCP Option 60 included. If the device is software reset
(e.g., from the Web interface or SNMP), only a single DHCP sequence containing
Option 60 is sent.
33.1.4.1 Provisioning from HTTP Server using DHCP Option 67

Most DHCP servers support the configuration of individual DHCP option values for different
devices on the network. The DHCP configuration should be modified so that the device
receives a URL to the configuration file in Option 67, along with IP addressing and DNS
server information. The DHCP response is processed by the device upon startup and the
device automatically downloads the configuration file from the HTTP server specified in the
DHCP response. This method is NAT-safe.
Below is an example of a Linux DHCP configuration file (dhcpd.conf) showing the required
format of Option 67:
ddns-update-style ad-hoc;
default-lease-time 3600;
max-lease-time 3600;
class "audiocodes" {
match if(substring(hardware, 1, 3) = 00:90:8f);
}
subnet 10.31.0.0 netmask 255.255.0.0 {
pool {
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allow members of "audiocodes";

range 10.31.4.53 10.31.4.75;
option routers 10.31.0.1;
option subnet-mask 255.255.0.0;
option domain-name-servers 10.1.0.11;
option bootfile-name
"INI=http://www.corp.com/master.ini";
option dhcp-parameter-request-list 1,3,6,51,67;
}
}
33.1.4.2 Provisioning from TFTP Server using DHCP Option 66

This method is suitable when the network in which the device is deployed contains a
provisioning TFTP server for all network equipment, without being able to distinguish
between AudioCodes and non-AudioCodes devices.
Upon startup, the device searches for Option 66 in the DHCP response from the DHCP
server. If Option 66 contains a valid IP address, the device attempts to download, through
TFTP, a file that has a filename containing the device's MAC address (e.g.,
00908f0130aa.ini). This method requires a provisioning server at the customer premises.
This method loads the configuration file to the device as a one-time action. The download
is only repeated if the device is manually restored to factory defaults (by pressing the
hardware reset button while the Ethernet cable is not connected) and DHCP is enabled
(see note below).
Notes:
• For TFTP configuration using DHCP Option 66, enable DHCP on your device:
DHCPEnable = 1 and DHCPRequestTFTPParams = 1.
• Access to the core network using TFTP is not NAT-safe.
• The TFTP data block size (packets) when downloading a file from a TFTP server
for the Automatic Update mechanism can be configured using the
AUPDTftpBlockSize parameter.
33.1.4.3 Provisioning the Device using DHCP Option 160

You can provision the device using DHCP Option 160. DHCP Option 160 provides the
device with the URL address of the provisioning server from where it can download its
software (.cmp) and configuration (.ini) files. The URL can also include the names of the
required files to download and their folder location on the server.
If you enable DHCP client functionality with DHCP Option 160, upon a device reset or
power up, the device (as a DHCP client) sends a DHCP request to the DHCP server to
obtain networking information (e.g., the device's IP address) and the URL address of the
provisioning server.
 The following syntax is supported for defining the URL and configuration/firmware
filenames in DHCP Option 160 (on the DHCP server):
 <protocol>://<server IP address or hostname>
 <protocol>://<server IP address or hostname>/<software filename>
 <protocol>://<server IP address or hostname>/;<configuration filename>
 <protocol>://<server IP address or hostname>/<software filename>;<configuration
filename>
The protocol can be HTTP, HTTPS, FTP, or TFTP. As shown above, the URL can include
both the software and configuration filenames. In this case, they must be separated by a
semicolon (;) and without spaces.

If the URL does not specify a configuration filename or the file does not exist on the
provisioning server, the device requests from the server a "default" configuration file whose
name includes the device's product name and MAC address (<Product><MAC>.ini, for
example, "MP114FXS00908f5b1035.ini"). If this "default" file also does not exist on the
server, the device attempts to retrieve another "default" configuration file whose name
includes only the device's product name (<Product>.ini, for example, "MP114FXS.ini"). The
device makes up to three attempts to download the configuration file if a failure occurs (i.e.,
file not exist or any other failure reason). This applies to each of the configuration files, as
mentioned previously.
If the URL specifies a software file, the device makes only one attempt to download the file
(even if a failure occurs). If the URL does not specify a software file, the device does not
make any attempt to download a software file.
Once the device downloads the file(s), it undergoes a reset to apply the configuration
and/or software. In addition, once the file(s) has been downloaded, the device ignores all
future DHCP Option 160 messages. Only if the device is restored to factory defaults will it
process Option 160 again (and download any required files).
 To enable provisioning using DHCP Option 160:

1. Make sure that the DHCP server is configured with the appropriate information
(including the URL address of the provisioning server for Option 160).
2. Make sure that the required configuration and/or software files are located on the
provisioning server.
3. Enable DHCP client functionality, as described in DHCP-based Provisioning on page
394.
4. Enable the device to include DHCP Option 160 in the DHCP Parameter Request List
field of the DHCP request packet that is sent to the DHCP server. Do this by loading
an ini file to the device with the following parameter setting:
DhcpOption160Support = 1
5. Reset the device with a save-to-flash for your settings to take effect.
33.1.5 HTTP-based Provisioning

An HTTP or HTTPS server can be located in the network in which the device is deployed,
storing configuration and software files for the device to download. This does not require
additional servers and is NAT-safe.
For example, assume the core network HTTPS server is https://www.corp.com. A master
configuration ini file can be stored on the server, e.g.,
https://www.corp.com/gateways/master.ini. This file could point to additional ini files,
Auxiliary files (e.g., call progress tones), and software files (cmp), all on the same HTTP
server or different HTTP servers in the network.
The main advantage of this method is that the device can be configured to periodically
check the HTTP server for file updates. HTTP(S) is not sensitive to NAT devices, enabling
configuration whenever needed without on-site intervention. For additional security, the
URL may contain a different port, and username and password.
The only configuration required is to preconfigure the device(s) with the URL of the initial
(master) ini file. This can be done using one of the following methods:
 DHCP as described in ''DHCP-based Provisioning'' on page 392or via TFTP at a
staging warehouse. The URL is configured using the IniFileURL parameter.
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 Private labeling (preconfigured during the manufacturing process).

 Using DHCP Option 67 (see Provisioning from HTTP Server using DHCP Option 67
on page 393).
 Manually on-site, using the RS-232 port or Web interface.
When the device is deployed at the customer site, local DHCP server provides the devices
with IP addressing and DNS server information. From the URL provided in the DHCP
response, the device can then contact the HTTP server at the core network and
automatically download its configuration. The URL can be a simple file name or contain the
device's MAC or IP address, e.g.:
 http://corp.com/config-<MAC>.ini - which becomes, for example,
http://corp.com/config-00908f030012.ini
 http://corp.com/<IP>/config.ini - which becomes, for example,
http://corp.com/192.168.0.7/config.ini
For more information on HTTP-based provisioning, see ''HTTP/S-Based Provisioning using
the Automatic Update Feature'' on page 397.
33.1.5.1 Loading Files Securely by Disabling TFTP

The TFTP protocol is not considered secure and some network operators block it using a
firewall. It is possible to disable TFTP completely, using the ini file parameter
EnableSecureStartup (set to 1). Secure protocols such as HTTPS may be used to fetch the
device configuration instead.
 To download the ini file to the device using HTTPS instead of TFTP:
1. Prepare the device's configuration file on an HTTPS server and obtain a URL to the
file (e.g., https://192.168.100.53/gateways.ini).
2. Enable DHCP, if necessary.
3. Enable SSH and connect to it.
4. In the CLI, use the ini file parameters IniFileURL (for defining the URL of the
configuration file) and EnableSecureStartup (for disabling TFTP), and then restart the
device with the new configuration:
/conf/scp IniFileURL https://192.168.100.53/gateways.ini
/conf/scp EnableSecureStartup 1
/conf/sar bootp
Note: Once Secure Startup has been enabled, it can only be disabled by setting
EnableSecureStartup to 0 using the CLI. Loading a new ini file using BootP/TFTP is
not possible until EnableSecureStartup is disabled.

33.1.6 FTP- or NFS-based Provisioning

Some networks block access to HTTP(S). The Automatic Update feature provides limited
support for FTP/FTPS connectivity. Periodic polling for updates is not possible since these
protocols do not support conditional fetching, i.e., updating files only if it is changed on the
server.
The only difference between this method and those described in ''HTTP-based
Provisioning'' on page 395 and Provisioning from HTTP Server using DHCP Option 67 on
page 393 is that the protocol in the URL is "ftp" (instead of "http").
Notes:
• Unlike FTP, NFS is not NAT-safe.
• NFS v2/v3 is also supported.
33.1.7 Provisioning using AudioCodes EMS

AudioCodes EMS server functions as a core-network provisioning server. The device's
SNMP Manager should be configured with the IP address of the EMS server, using one of
the methods detailed in the previous sections. As soon as a registered device contacts the
EMS server through SNMP, the EMS server handles all required configuration
automatically, upgrading software as needed. This alternative method doesn't require
additional servers at the customer premises, and is NAT-safe.
33.2 HTTP/S-Based Provisioning using the Automatic

Update Feature
The Automatic Update feature can be used for automatic provisioning of the device through
HTTP/S. Automatic provisioning is useful for large-scale deployment of devices. In some
cases, the devices are shipped to the end customer directly from the manufacturer. In other
cases, they may pass through a staging warehouse. Configuration may occur at the
staging warehouse or at the end-customer premises.
The device may be preconfigured during the manufacturing process (commonly known as
private labeling). Typically, a two-stage configuration process is implemented whereby
initial configuration includes only basic configuration, while the final configuration is done
only when the device is deployed in the live network.
Warning: If you use the IniFileURL parameter for the Automatic Update feature, do
not use the Web interface to configure the device. If you do configure the device
through the Web interface and save (burn) the new settings to the device's flash
memory, the IniFileURL parameter is automatically set to 0 and Automatic Updates is
consequently disabled. To enable Automatic Updates again, you need to re-load the
ini file (using the Web interface or BootP) with the correct IniFileURL settings. As a
safeguard to an unintended burn-to-flash when resetting the device, if the device is
configured for Automatic Updates, the 'Burn To FLASH' field under the Reset
Configuration group in the Web interface's Maintenance Actions page is automatically
set to No by default.
Version 6.6 397 MP-11x & MP-124

MP-11x & MP-124
Notes:
• For a description of all the Automatic Update parameters, see ''Automatic Update
Parameters'' on page 646.
• For additional security, use HTTPS or FTPS. The device supports HTTPS (RFC
2818) and FTPS using the AUTH TLS method <draft-murray-auth-ftp-ssl-16>.
33.2.1 Files Provisioned by Automatic Update

You can use the Automatic Update feature to update the device with any of the following
files:
 Software file (cmp)
 Auxiliary files (e.g., Call Progress Tones, SSL Certificates, SSL Private Key)
 Configuration file (ini file)
33.2.2 File Location for Automatic Update

The files for updating the device can be stored on any standard Web (HTTP/S), FTP, or
NFS server. The files can be loaded periodically to the device using HTTP, HTTPS, FTP,
or NFS. This mechanism can be used even when the device is installed behind NAT and
firewalls.
The Automatic Update feature is done per file and configured by specifying the file name
and URL address of the provisioning server where the file is located. For a description of
the parameters used to configure URLs per file, see ''Automatic Update Parameters'' on
page 646. Below are examples for configuring the file names and their URLs for Automatic
Update:
 ini File:
IniFileURL = 'http://www.corp.com/configuration.ini'
CptFileURL = 'http://www.corp.com/call_progress.dat'
AutoCmpFileUrl = 'http://www.corp.com/SIP_F7.00A.008.cmp
Note: For configuration files (ini), the file name in the URL can automatically contain
the device's MAC address for enabling the device to download a file unique to the
device. For more information, see ''MAC Address Automatically Inserted in
Configuration File Name'' on page 402.
33.2.3 Triggers for Automatic Update

The Automatic Update feature can be triggered by the following:
 Upon device startup (reset or power up).
 Upon startup, but before the device is operational, if the Secure Startup feature is
enabled (see Loading Files Securely by Disabling TFTP on page 396).
 Periodically:
• Specified time of day (e.g., 18:00), configured by the ini file parameter
AutoUpdatePredefinedTime.
• Interval between Automatic Updates (e.g., every 60 minutes), configured by the
ini file parameter AutoUpdateFrequency.
 Centralized provisioning server request:
• Upon receipt of an SNMP request from the provisioning server.

• Upon receipt of a special SIP NOTIFY message from the provisioning server. The
NOTIFY message includes an Event header with the AudioCodes proprietary
value, "check-sync;reboot=false", as shown in the example below:
NOTIFY sip:<user>@<dsthost> SIP/2.0
To: sip:<user>@<dsthost>
From: sip:sipsak@<srchost>
CSeq: 10 NOTIFY
Call-ID: 1234@<srchost>
Event: check-sync;reboot=false
To enable this feature through the Web interface:
a. Open the Advanced Parameters page (Configuration tab > VoIP menu >
SIP Definitions > Advanced Parameters).
b. Under the Misc Parameters group, set the 'SIP Remote Reset' parameter to
Enable.
c. Click Submit.
33.2.4 Access Authentication with HTTP Server

You can configure the device to authenticate itself with the HTTP/S server. The device
authenticates itself by providing the HTTP/S server with its authentication username and
password. You can configure one of the following HTTP authentication schemes:
 Basic Access Authentication: The device provides its username and password to
the HTTP server. The username and password is configured in the URL that you
define for downloading the file:
• ini file:
AutoCmpFileUrl = 'https://<username>:<password>@<IP address
or domain name>/<file name>'
 Digest Access Authentication: The authentication username and password is
negotiated between the device and HTTP/S server, using digest MD5 cryptographic
hashing. This method is safer than basic access authentication. The digest
authentication username and password are configured using the
AUPDDigestUsername and AUPDDigestPassword parameters, respectively.
33.2.5 Querying Provisioning Server for Updated Files

Each time the Automatic Update feature is triggered, for each file and its configured URL
the device does the following:
1. If you have configured the device to authenticate itself to the HTTP/S server for secure
access, the device sends the access authentication username and password to the
HTTP/S server (for more information, see ''Access Authentication with HTTP Server''
on page 399). If authentication succeeds, Step 2 occurs.
2. The device establishes an HTTP/S connection with the URL host (provisioning
server). If the connection is HTTPS, the device verifies the certificate of the
provisioning server, and presents its own certificate if requested by the server.
3. The device queries the provisioning server for the requested file by sending an HTTP
Get request. This request contains the HTTP User-Agent Header, which identifies the
device to the provisioning server. By default, the header includes the device's model
name, MAC address, and currently installed software and configuration versions.
Based on its own dynamic applications for logic decision making, the provisioning
server uses this information to check if it has relevant files available for the device and
determines which files must be downloaded (working in conjunction with the HTTP If-
Modified-Since header, described further on in this section).
Version 6.6 399 MP-11x & MP-124

MP-11x & MP-124
You can configure the information sent in the User-Agent header, using the
AupdHttpUserAgent parameter. The information can include any user-defined string or
the following supported string variable tags (case-sensitive):
• <NAME>: product name, according to the installed Software License Key
• <MAC>: device's MAC address
• <VER>: software version currently installed on the device, e.g., "7.00.200.001"
• <CONF>: configuration version, as configured by the ini file parameter,
INIFileVersion
The device automatically populates these tag variables with actual values in the sent
header. By default, the device sends the following in the User-Agent header:
User-Agent: Mozilla/4.0 (compatible; AudioCodes;
<NAME>;<VER>;<MAC>;<CONF>)
For example, if you set AupdHttpUserAgent = MyWorld-<NAME>;<VER>(<MAC>), the
device sends the following User-Agent header:
User-Agent: MyWorld-Mediant;7.00.200.001(00908F1DD0D3)
Note: If you configure the AupdHttpUserAgent parameter with the <CONF> variable
tag, you must reset the device with a burn-to-flash for your settings to take effect.
4. If the provisioning server has relevant files available for the device, the following
occurs, depending on file type and configuration:
• File Download upon each Automatic Update process: This is applicable to
software (.cmp), ini files. In the sent HTTP Get request, the device uses the
HTTP If-Modified-Since header to determine whether to download these files.
The header contains the date and time (timestamp) of when the device last
downloaded the file from the specific URL. This date and time is regardless of
whether the file was installed or not on the device. An example of an If-Modified-
Since header is shown below:
If-Modified-Since: Mon, 1 January 2014 19:43:31 GMT
If the file on the provisioning server was unchanged (modified) since the date and
time specified in the header, the server replies with an HTTP 304 response and
the file is not downloaded. If the file was modified, the provisioning server sends
an HTTP 200 OK response with the file in the body of the HTTP response. The
device downloads the file and compares the version of the file with the currently
installed version on its flash memory. If the downloaded file is of a later version,
the device installs it after the device resets (which is only done after the device
completes all file downloads); otherwise, the device does not reset and does not
install the file.
To enable the automatic software (.cmp) file download method based on this
timestamp method, use the ini file parameter, AutoCmpFileUrl. The device uses
the same configured URL to download the .cmp file for each subsequent
Automatic Update process.
You can also enable the device to run a CRC on the downloaded configuration
file (ini) to determine whether the file has changed in comparison to the
previously downloaded file. Depending on the CRC result, the device can install
or discard the downloaded file. For more information, see ''Cyclic Redundancy
Check on Downloaded Configuration Files'' on page 402.

Notes:
• When this method is used, there is typically no need for the provisioning server to
check the device’s current firmware version using the HTTP-User-Agent header.
• The Automatic Update feature assumes that the Web server conforms to the
HTTP standard. If the Web server ignores the If-Modified-Since header or doesn’t
provide the current date and time during the HTTP 200 OK response, the device
may reset itself repeatedly. To overcome this problem, modify the update
frequency, using the ini file parameter AutoUpdateFrequency.
• One-time File Download: This is applicable to software (.cmp) and Auxiliary

(e.g., call progress tone / CPT) files. The device downloads these files only once,
regardless of how many times the device may repeat the Automatic Update
process. Once they are downloaded, the device discards their configured URLs.
To update these files again, you need to configure their URL addresses and
filenames again. Below is an example of how to configure URLs for some of
these files:
Auxiliary Files:
♦ ini:
CptFileURL =
'https://www.company.com/call_progress.dat'
Software (.cmp) File:
♦ ini:
CmpFileUrl =
'https://www.company.com/device/v.6.80A.227.005.cmp'
Notes:
• For one-time file download, the HTTP Get request sent by the device does not
include the If-Modified-Since header. Instead, the HTTP-User-Agent header can
be used in the HTTP Get request to determine whether firmware update is
required.
• When downloading SSL certificates (Auxiliary file), it is recommended to use
HTTPS with mutual authentication for secure transfer of the SSL Private Key.
5. If the device receives an HTTP 301/302/303 redirect response from the provisioning
server, it establishes a connection with the new server at the redirect URL and re-
sends the HTTP Get request.
33.2.6 File Download Sequence

Whenever the Automatic Update feature is triggered (see ''Triggers for Automatic Update''
on page 398), the device attempts to download each file from the configured URLs, in the
following order:
1. ini file
2. Periodic software file (.cmp) download
3. One-time software file (.cmp) download
4. Auxiliary file(s)
The following files automatically instruct the device to reset:
 Periodic software file (.cmp)
 One-time software file (.cmp)
Version 6.6 401 MP-11x & MP-124

MP-11x & MP-124
When multiple files requiring a reset are downloaded, the device resets only after it has
downloaded and installed all the files. However, you can explicitly instruct the device to
immediately reset for the following files:
 ini file: Use the ResetNow in file parameter
Warning: If you use the ResetNow parameter in an ini file for periodic automatic
provisioning with non-HTTP (e.g., TFTP) and without CRC, the device resets after
every file download. Therefore, use the parameter with caution and only if necessary
for your deployment requirements.
Notes:
• For ini file downloads, by default, parameters not included in the file are set to
defaults. To retain the current settings of these parameters, set the
SetDefaultOnINIFileProcess parameter to 0.
• If you have configured one-time software file (.cmp) download (configured by the
ini file parameter CmpFileURL), the device will only apply the file if one-time
software updates are enabled. This is disabled by default to prevent unintentional
software upgrades. To enable one-time software upgrades, set the ini file
parameter AutoUpdateCmpFile to 1.
• If you need to update the device's software and configuration, it is recommended
to first update the software. This is because the current ("old") software (before the
upgrade) may not be compatible with the new configuration. However, if both files
are available for download on the provisioning server(s), the device first
downloads and applies the new configuration, and only then does it download and
install the new software. Therefore, this is a very important issue to take into
consideration.
33.2.7 Cyclic Redundancy Check on Downloaded Configuration Files

You can enable the device to perform cyclic redundancy checks (CRC) on downloaded
configuration files (ini) during the Automatic Update process. The CRC checks whether the
content (raw data) of the downloaded file is different to the content of the previously
downloaded file from the previous Automatic Update process. The device compares the
CRC check value (code) result with the check value of the previously downloaded file. If
the check values are identical, it indicates that the file has no new configuration settings,
and the device discards the file. If the check values are different, it indicates that the
downloaded file is different (i.e., includes updates), and the device installs the downloaded
file and applies the new configuration settings.
CRC is useful, for example, when the service provider replaces a file, on the provisioning
server, with another file whose contents are the same. When the device sends an HTTP
Get request during the Automatic Update process, the provisioning server sends the new
file to the device. This occurs as the timestamp between the previously downloaded file
and this new file is different (determined by the HTTP If-Modified-Since header in the Get
request). Therefore, the CRC feature can be used to prevent the device from installing
such files.
For enabling CRC, use the ini file parameter AUPDCheckIfIniChanged. By default, CRC is
disabled. For more information on the parameter, see ''Automatic Update Parameters'' on
page 646.

33.2.8 MAC Address Automatically Inserted in Configuration File Name

You can configure the file name of the configuration file (ini) in the URL to automatically
include the MAC address of the device. As described in ''File Location for Automatic
Update'' on page 398, the file name is included in the configured URL of the provisioning
server where the file is located.
Including the MAC address in the file name is useful if you want the device to download a
file that is unique to the device. This feature is typically implemented in mass provisioning
of devices where each device downloads a specific configuration file. In such a setup, the
provisioning server stores configuration files per device, where each file includes the MAC
address of a specific device in its file name.
To support this feature, you need to include the case-sensitive string, "<MAC>" anywhere
in the configured file name of the URL, for example:
IniFileURL = 'https://www.company.com/config_<MAC>.ini'
The device automatically replaces the string with its hardware MAC address, resulting in a
file name request that contains the device's MAC address, for example,
config_00908F033512.ini. Therefore, you can configure all the devices with the same URL
and file name.
33.2.9 Automatic Update Configuration Examples

This section provides a few examples on configuring the Automatic Update feature.
33.2.9.1 Automatic Update for Single Device

This simple example describes how to configure the Automatic Update feature for updating
a single device. In this example, the device queries the provisioning server for software,
configuration and Auxiliary files every 24 hours.
 To set up Automatic Provisioning for single device (example):

1. Set up an HTTP Web server (e.g., http://www.company.com) and place all the
required configuration files on this server.
2. Configure the device with the IP address of the DNS server for resolving the domain
name (e.g., http://www.company.com) that is used in the URL of the provisioning
server. You configure this in the Interface table:
• ini File:
[ InterfaceTable ]
FORMAT InterfaceTable_Index =
InterfaceTable_ApplicationTypes,
InterfaceTable_InterfaceMode, InterfaceTable_IPAddress,
InterfaceTable_PrefixLength, InterfaceTable_Gateway,
InterfaceTable_VlanID, InterfaceTable_InterfaceName,
InterfaceTable_PrimaryDNSServerIPAddress,
InterfaceTable_SecondaryDNSServerIPAddress,
InterfaceTable_UnderlyingDevice;
InterfaceTable 0 = 6, 10, 10.15.7.95, 16, 10.15.0.1, 1,
"Voice", 80.179.52.100, 0.0.0.0, "vlan 1";
[ \InterfaceTable ]
3. Configure the device with the following Automatic Update settings:
a. Automatic Update is done every 24 hours (1440 minutes):
♦ ini File:
AutoUpdateFrequency = 1440
Version 6.6 403 MP-11x & MP-124

MP-11x & MP-124
b. Automatic Update of software file (.cmp):

♦ ini File:
AutoCmpFileUrl = 'https://www.company.com/sw.cmp'
c. Automatic Update of Call Progress Tone file:
♦ ini File:
CptFileURL =
'https://www.company.com/call_progress.dat'
d. Automatic Update of ini configuration file:
♦ ini File:
IniFileURL = 'https://www.company.com/config.ini'
e. Enable Cyclical Redundancy Check (CRC) on downloaded ini file:
♦ ini File:
AUPDCheckIfIniChanged = 1
4. Power down and then power up the device.
33.2.9.2 Automatic Update from NFS, FTP and HTTP Servers

This example describes how to configure the Automatic Update feature where files are
stored and downloaded from different file server types. The example scenario includes the
following:
 NFS server (Version 2) at 10.13.2.10 for storing the CPT file.
 FTPS server at ftpserver.corp.com for storing the Voice Prompts (VP) file. The login
credentials to the server are username "root" and password "wheel".
 HTTP server at www.company.com for storing the configuration file (ini).
 DNS server at 80.179.52.100 for resolving the domain names of the provisioning
servers (FTPS and HTTP).
 To set up Automatic Provisioning for files stored on different server types

(example):
1. CPT file:
a. Set up an NFS server and copy the CPT file to the directory /usr/shared/public on
the NFS server.
b. Configure the device with the NFS server:
[ NFSServers ]
FORMAT NFSServers_Index = NFSServers_HostOrIP,
NFSServers_RootPath, NFSServers_NfsVersion,
NFSServers_AuthType, NFSServers_UID, NFSServers_GID,
NFSServers_VlanType;
NFSServers 0 = "10.31.2.10", "/usr/share/public/", 2, 1, 0,
1, 1;
[ \NFSServers ]
c. Configure the device with the URL path of the CPT file:
♦ ini File:
CptFileURL =
'file://10.31.2.10/usr/share/public/usa_tones.dat'
2. VP file:
a. Set up an FTPS server and copy the VP file to the server.
b. Configure the device with the URL path of the VP file:
♦ ini File:
VPFileUrl =
'ftps://root:[email protected]/vp.dat'

3. Software (.cmp) and ini files:

a. Set up an HTTP Web server and copy the .cmp and configuration files to the
server.
b. Configure the device with the URL paths of the .cmp and ini files:
♦ ini File:
AutoCmpFileUrl =
'http://www.company.com/device/sw.cmp'
IniFileURL = 'http://www.company.com/device/inifile.ini'
4. Configure the device with the IP address of the DNS server for resolving the domain
names of the FTPS and HTTP servers:
[ InterfaceTable ]
FORMAT InterfaceTable_Index = InterfaceTable_ApplicationTypes,
InterfaceTable 0 = 6, 10, 10.15.7.95, 16, 10.15.0.1, 1,
"Voice", 80.179.52.100, 0.0.0.0, "vlan 1";
[ \InterfaceTable ]
5. Configure the device to perform the Automatic Update process daily at 03:00 (3 a.m):
• ini File:
AutoUpdateFrequency = '03:00'
33.2.9.3 Automatic Update for Mass Deployment

This example describes how to configure the Automatic Update feature for updating
multiple devices (i.e., mass deployment) using an HTTP provisioning server. In this
example, all the devices are configured to download the same "master" configuration file.
This file serves as the configuration template and instructs the devices which files to
download and how often to perform the Automatic Update process. In addition, the master
file also instructs each device to download an ini configuration file whose file name
contains the MAC address of the device.
The example scenario is as follows:
 All devices download a "master" configuration file that contains the following:
• Common configuration shared by all device's.
• Specific configuration that instructs each device to download a specific
configuration file based on the device's MAC address, using the special string
"<MAC>" in the URL, as described in ''MAC Address Automatically Inserted in
Configuration File Name'' on page 402.
 Device queries the provisioning server daily at 24:00 (midnight) for software,
configuration and Auxiliary files.
 HTTP-based provisioning server at www.company.com for storing the files.
 DNS server at 80.179.52.100 for resolving the domain name of the provisioning
server.
Version 6.6 405 MP-11x & MP-124

MP-11x & MP-124
 To set up automatic provisioning for mass provisioning (example):

1. Create a "master" configuration file template named "master_configuration.ini" with the
following settings:
• Common configuration for all devices:
♦ ini file:
AutoUpdatePredefinedTime = '24:00'
CptFileURL = 'https://www.company.com/call_progress.dat'
AutoCmpFileUrl = 'https://www.company.com/sw.cmp'
• Configuration per device based on MAC address:
♦ ini file:
IniFileURL = 'http://www.company.com/config_<MAC>.ini'
(automatic-update)# voice-configuration
http://www.company.com/config_<MAC>.ini
2. Copy the master configuration file that you created in Step 1 as well as the CPT and
.cmp files to the HTTP-based provisioning server.
3. Configure each device with the following:
a. URL of the master configuration file:
♦ ini File:
IniFileURL =
'http://www.company.com/master_configuration.ini'
b. Configure the device with the IP address of the DNS server for resolving the
domain name (e.g., http://www.company.com) that is used in the URL for the
provisioning server. This is done in the Interface table:
♦ ini File:
[ InterfaceTable ]
InterfaceTable 0 = 6, 10, 10.15.7.95, 16, 10.15.0.1, 1,
"Voice", 80.179.52.100, 0.0.0.0, "vlan 1";
[ \InterfaceTable ]
4. Power down and then power up the device.

User's Manual 34. Restoring Factory Defaults
34 Restoring Factory Defaults

You can restore the device's configuration to factory defaults using one of the following
methods:
 CLI (see 'Restoring Defaults using CLI' on page 407)
 Hardware reset pinhole button (see Restoring Defaults using Hardware Reset Button
on page 407)
 Loading an empty ini file (see 'Restoring Defaults using an ini File' on page 408)
34.1 Restoring Defaults using CLI

The device can be restored to factory defaults using CLI, as described in the procedure
below.
 To restore factory defaults using CLI:

1. Access the CLI:
a. Connect the RS-232 serial port of the device to the communication port on your
computer. For serial cabling, refer to the Hardware Installation Manual.
b. Establish serial communication with the device using a serial communication
program (such as HyperTerminalTM) with the following communication port
settings:
♦ Baud Rate: 9,600 bps for MP-11x; 115,200 bps for MP-124
♦ Data Bits: 8
♦ Parity: None
♦ Stop Bits: 1
♦ Flow Control: None
2. At the CLI prompt, type the following command to access the configuration mode, and
then press Enter:
# conf
3. At the prompt, type the following command to reset the device to default settings, and
then press Enter:
# RestoreFactorySettings
34.2 Restoring Defaults using Hardware Reset Button

The device's hardware reset pinhole button can be used to reset the device to default
settings.
 To restore default settings using the hardware reset pinhole button:

 To restore MP-124 to factory default settings:
a. Disconnect the Ethernet cable from the device.
b. With a paper clip or any other similar pointed object, press and hold down the
reset pinhole button for at least 12 seconds, but no more than 25 seconds.
 To restore MP-11x to factory default settings:
a. Disconnect the Ethernet cable from the device.
b. With a paper clip or any other similar pointed object, press and hold down the
reset pinhole button for about six seconds; the Fail LED turns red and the device
restores to factory default settings.
c. When the Fail LED turns off, reconnect the Ethernet cable to the device.
Version 6.6 407 MP-11x & MP-124

MP-11x & MP-124
34.3 Restoring Defaults using an ini File

You can restore the device to factory default settings by loading an empty ini file to the
device. This is done using the Web interface's Configuration File page (see 'Backing Up
and Loading Configuration File' on page 388). If the ini file does include content (e.g.,
parameters), ensure that they are on lines beginning with comment signs (i.e., semicolons
";") so that the device ignores them.
Note: The only settings that are not restored to default are the management (OAMP)
IP address and the Web interface's login user name and password.

Part VIII
Status, Performance Monitoring
and Reporting
User's Manual 35. System Status
35 System Status
This section describes how to view various system statuses.
35.1 Viewing Device Information

The Device Information page displays various hardware and software information of the
device. This page also lists any Auxiliary files that have been installed on the device and
allows you to remove them.
 To access the Device Information page:

 Open the Device Information page (Status & Diagnostics tab > System Status
menu > Device Information).
 To delete a loaded file:

 Click the Delete button corresponding to the file that you want to delete. Deleting a file
takes effect only after device reset (see 'Resetting the Device' on page 363).
35.2 Viewing Ethernet Port Information

The Ethernet Port Information page displays read-only information on the Ethernet port
connections.
Note: The Ethernet Port Information page can also be accessed from the Home page
(see 'Viewing the Home Page' on page 63).
 To view Ethernet port information:

 Open the Ethernet Port Information page (Status & Diagnostics tab > System
Version 6.6 411 MP-11x & MP-124

MP-11x & MP-124
Status menu > Ethernet Port Information).
Table 35-1: Ethernet Port Information Parameters
Port Duplex Mode Displays whether the port is in half or duplex mode.
Port Speed Displays the speed (in Mbps) of the Ethernet port.

User's Manual 36. Carrier-Grade Alarms
36 Carrier-Grade Alarms
This section describes how to view the following types of alarms:
 Active alarms - see 'Viewing Active Alarms' on page 413
 Alarm history - see 'Viewing Alarm History' on page 413
36.1 Viewing Active Alarms

The Active Alarms page displays a list of currently active alarms. You can also access this
page from the Home page (see 'Viewing the Home Page' on page 63).
Note:
• The alarms in the table are deleted upon a device reset.
• To configure the maximum number of active alarms that can be displayed in the
table, see the ini file parameter, ActiveAlarmTableMaxSize.
• For more information on SNMP alarms, refer to the SNMP Reference Guide
document.
 To view the list of active alarms:

 Open the Active Alarms page (Status & Diagnostics tab > System Status menu >
Carrier-Grade Alarms > Active Alarms).
For each alarm, the following information is provided:

 Severity: severity level of the alarm:
• Critical (red)
• Major (orange)
• Minor (yellow)
 Source: unit from which the alarm was raised
 Description: brief explanation of the alarm
 Date: date and time that the alarm was generated
You can view the next 20 alarms (if exist), by clicking the Go to page button.
36.2 Viewing Alarm History

The Alarms History page displays a list of alarms that have been raised and traps that have
been cleared.
 To view the list of history alarms:

 Open the Alarms History page (Status & Diagnostics tab > System Status menu >
Carrier-Grade Alarms > Alarms History).
Version 6.6 413 MP-11x & MP-124

MP-11x & MP-124
For each alarm, the following information is provided:

 Severity: severity level of the alarm:
• Critical (red)
• Major (range)
• Minor (yellow)
• Cleared (green)
 Source: unit from which the alarm was raised
 Description: brief explanation of the alarm
 Date: date and time that the alarm was generated
You can view the next 20 alarms (if exist), by clicking the Go to page button.
 To delete all the alarms in the table:

1. Click the Delete History Table button; a confirmation message box appears.
2. Click OK to confirm.

User's Manual 37. VoIP Status
37 VoIP Status
This section describes how to view VoIP status and statistics.
37.1 Viewing Analog Port Information

The Home page allows you to view detailed information on selected FXS and FXO analog
ports such as RTP/RTCP and voice settings.
 To view information on an analog port:

1. Open the Home page.
2. On the graphical display of the device, click the required analog port; a shortcut menu
appears.
3. From the shortcut menu, choose Port Settings; the Basic Channel Information page
appears with the Basic tab selected (displayed in green):
Figure 37-1: Basic Channel Information Page
4. To view additional channel information, click the required tab - SIP, RTP/RTCP, and
Voice Settings.
37.2 Viewing Active IP Interfaces

The IP Interface Status page displays the device's active IP interfaces that are listed in the
Multiple Interface Table page (see 'Configuring IP Network Interfaces' on page 124).
 To view the active IP network interfaces:

 Open the IP Interface Status page (Status & Diagnostics tab > VoIP Status menu >
IP Interface Status).
Version 6.6 415 MP-11x & MP-124

MP-11x & MP-124
37.3 Viewing Performance Statistics

The Basic Statistics page provides read-only, device performance statistics. This page is
refreshed every 60 seconds. The duration that the currently displayed statistics has been
collected is displayed above the statistics table.
 To view performance statistics:

 Open the Basic Statistics page (Status & Diagnostics tab > VoIP Status menu >
Performance Statistics).
Figure 37-2: Basic Statistics Page
The duration that the displayed statistics were collected is displayed in seconds above the
table. To reset the performance statistics to zero, click the Reset Statistics button.
37.4 Viewing Call Counters

The IP to Tel Calls Count page and Tel to IP Calls Count page provide you with statistical
information on incoming (IP-to-Tel) and outgoing (Tel-to-IP) calls. The statistical
information is updated according to the release reason that is received after a call is
terminated (during the same time as the end-of-call Call Detail Record or CDR message is
sent). The release reason can be viewed in the 'Termination Reason' field in the CDR
message.
You can reset the statistical data displayed on the page (i.e., refresh the display), by
clicking the Reset Counters button located below the table.
 To view IP-to-Tel and Tel-to-IP call counters:

 Open the Call Counters page that you want to view (Status & Diagnostics tab > VoIP
Status menu > IP to Tel Calls Count or Tel to IP Calls Count); the figure below
shows the IP to Tel Calls Count page.
Figure 37-3: Calls Count Page

The fields in this page are described in the following table:

Table 37-1: Call Counters Description
Counter Description
Number of Attempted Indicates the number of attempted calls. It is composed of established

Calls and failed calls. The number of established calls is represented by the
'Number of Established Calls' counter. The number of failed calls is
represented by the failed-call counters. Only one of the established /
failed call counters is incremented every time.
Number of Established Indicates the number of established calls. It is incremented as a result of
Calls one of the following release reasons if the duration of the call is greater
than zero:
 GWAPP_REASON_NOT_RELEVANT (0)
 GWAPP_NORMAL_CALL_CLEAR (16)
 GWAPP_NORMAL_UNSPECIFIED (31)
And the internal reasons:
 RELEASE_BECAUSE_UNKNOWN_REASON
 RELEASE_BECAUSE_REMOTE_CANCEL_CALL
 RELEASE_BECAUSE_MANUAL_DISC
 RELEASE_BECAUSE_SILENCE_DISC
 RELEASE_BECAUSE_DISCONNECT_CODE
Note: When the duration of the call is zero, the release reason
GWAPP_NORMAL_CALL_CLEAR increments the 'Number of Failed
Calls due to No Answer' counter. The rest of the release reasons
increment the 'Number of Failed Calls due to Other Failures' counter.
Percentage of The percentage of established calls from attempted calls.
Successful Calls (ASR)
Number of Calls Indicates the number of calls that failed as a result of a busy line. It is
Terminated due to a incremented as a result of the following release reason:
Busy Line GWAPP_USER_BUSY (17)
Number of Calls Indicates the number of calls that weren't answered. It's incremented as
Terminated due to No a result of one of the following release reasons:
Answer  GWAPP_NO_USER_RESPONDING (18)
 GWAPP_NO_ANSWER_FROM_USER_ALERTED (19)
 GWAPP_NORMAL_CALL_CLEAR (16) (when the call duration is
zero)
Number of Calls Indicates the number of calls that were terminated due to a call forward.
Terminated due to The counter is incremented as a result of the following release reason:
Forward RELEASE_BECAUSE_FORWARD
Number of Failed Calls Indicates the number of calls whose destinations weren't found. It is
due to No Route incremented as a result of one of the following release reasons:
 GWAPP_UNASSIGNED_NUMBER (1)
 GWAPP_NO_ROUTE_TO_DESTINATION (3)
Number of Failed Calls Indicates the number of calls that failed due to mismatched device
due to No Matched capabilities. It is incremented as a result of an internal identification of
Capabilities capability mismatch. This mismatch is reflected to CDR via the value of
the parameter DefaultReleaseReason (default is
GWAPP_NO_ROUTE_TO_DESTINATION (3)) or by the
GWAPP_SERVICE_NOT_IMPLEMENTED_UNSPECIFIED (79) reason.
Version 6.6 417 MP-11x & MP-124

MP-11x & MP-124
Counter Description
Number of Failed Calls Indicates the number of calls that failed due to unavailable resources or
due to No Resources a device lock. The counter is incremented as a result of one of the
following release reasons:
 GWAPP_RESOURCE_UNAVAILABLE_UNSPECIFIED
 RELEASE_BECAUSE_GW_LOCKED
Number of Failed Calls This counter is incremented as a result of calls that failed due to reasons
due to Other Failures not covered by the other counters.
Average Call Duration The average call duration (ACD) in seconds of established calls. The
(ACD) [sec] ACD value is refreshed every 15 minutes and therefore, this value
reflects the average duration of all established calls made within a 15
minute period.
Attempted Fax Calls Indicates the number of attempted fax calls.
Counter
Successful Fax Calls Indicates the number of successful fax calls.
Counter
37.5 Viewing Registered Users

The SAS/SBC Registered Users page displays a list of registered SAS users recorded in
the device's database.
 To view registered SAS users:

 Open the Registration Status page (Status & Diagnostics tab > VoIP Status menu >
Registered Users).
Figure 37-4: SAS/SBC Registered Users Page
Table 37-2: SAS Registered Users Parameters
Column Name Description
Address of An address-of-record (AOR) is a SIP or SIPS URI that points to a domain with
Record a location service that can map the URI to another URI (Contact) where the
user might be available.
Contact SIP URI that can be used to contact that specific instance of the User Agent for
subsequent requests.

37.6 Viewing Registration Status

The Registration Status page displays whether the device as a whole, its endpoints (FXS /
FXO), and SIP Accounts are registered to a SIP Registrar/Proxy server.
 To view the registration status:

 Open the Registration Status page (Status & Diagnostics tab > VoIP Status menu >
Registration Status).
Figure 37-5: Registration Status
 Registered Per Gateway:

• "YES" = Registration is per device
• "NO"= Registration is not per device
 Ports Registration Status:
• "REGISTERED" = channel is registered
• "NOT REGISTERED" = channel not registered
 Accounts Registration Status: registration status based on the Accounts table
(configured in 'Configuring Account Table' on page 213):
• Group Type: type of served group - Hunt Group or IP Group
• Group Name: name of the served group, if applicable
• Status: indicates whether or not the group is registered ("Registered" or
"Unregistered")
Note: The registration mode (i.e., per device, endpoint, account. or no registration) is
configured in the Hunt Group Settings table (see 'Configuring Hunt Group Settings' on
page 237) or using the TrunkGroupSettings ini file parameter.
Version 6.6 419 MP-11x & MP-124

MP-11x & MP-124
37.7 Viewing Call Routing Status

The Call Routing Status page provides you with information on the current routing method
used by the device. This information includes the IP address and FQDN (if used) of the
Proxy server with which the device currently operates.
 To view call routing status:

 Open the Call Routing Status page (Status & Diagnostics tab > VoIP Status menu >
Call Routing Status).
Figure 37-6: Call Routing Status Page
Table 37-3: Call Routing Status Parameters
Call-Routing Method  Proxy/GK = Proxy server is used to route calls.

 Routing Table = The Tel to IP Routing table is used to route calls.
IP Address  Not Used = Proxy server isn't defined.
 IP address and FQDN (if exists) of the Proxy server with which the
device currently operates.
State  N/A = Proxy server isn't defined.
 OK = Communication with the Proxy server is in order.
 Fail = No response from any of the defined Proxies.

37.8 Viewing IP Connectivity

The IP Connectivity page displays on-line, read-only network diagnostic connectivity
information on all destination IP addresses configured in the Tel to IP Routing page (see
'Configuring Tel to IP Routing' on page 256).
Note: The information in columns 'Quality Status' and 'Quality Info' (per IP address)
is reset if two minutes elapse without a call to that destination.
 To view IP connectivity information:

1. In the Routing General Parameters page, set the 'Enable Alt Routing Tel to IP'
parameter (AltRoutingTel2IPMode) to Enable or Status Only (see 'Configuring
General Routing Parameters' on page 255).
2. Open the IP Connectivity page (Status & Diagnostics tab > VoIP Status menu > IP
Connectivity).
Figure 37-7: IP Connectivity Page
Table 37-4: IP Connectivity Parameters
IP Address The IP address can be one of the following:

 IP address defined as the destination IP address in the Tel to IP Routing.
 IP address resolved from the host name defined as the destination IP
address in the Tel to IP Routing.
Host Name Host name (or IP address) as defined in the Tel to IP Routing.
Connectivity The method according to which the destination IP address is queried
Method periodically (ICMP ping or SIP OPTIONS request).
Version 6.6 421 MP-11x & MP-124

MP-11x & MP-124
Connectivity The status of the IP address' connectivity according to the method in the
Status 'Connectivity Method' field.
 OK = Remote side responds to periodic connectivity queries.
 Lost = Remote side didn't respond for a short period.
 Fail = Remote side doesn't respond.
 Init = Connectivity queries not started (e.g., IP address not resolved).
 Disable = The connectivity option is disabled, i.e., parameter 'Alt Routing Tel
to IP Mode' (AltRoutingTel2IPMode ini) is set to 'None' or 'QoS'.
Quality Status Determines the QoS (according to packet loss and delay) of the IP address.
 Unknown = Recent quality information isn't available.
 OK
 Poor
Notes:
 This parameter is applicable only if the parameter 'Alt Routing Tel to IP
Mode' is set to 'QoS' or 'Both' (AltRoutingTel2IPMode = 2 or 3).
 This parameter is reset if no QoS information is received for 2 minutes.
Quality Info. Displays QoS information: delay and packet loss, calculated according to
previous calls.
Notes:
 This parameter is applicable only if the parameter 'Alt Routing Tel to IP
Mode' is set to 'QoS' or 'Both' (AltRoutingTel2IPMode = 2 or 3).
 This parameter is reset if no QoS information is received for 2 minutes.
DNS Status DNS status can be one of the following:
 DNS Disable
 DNS Resolved
 DNS Unresolved

User's Manual 38. Reporting Information to External Party
38 Reporting Information to External Party

This section describes features for reporting various information to an external party.
38.1 RTP Control Protocol Extended Reports (RTCP XR)

RTP Control Protocol Extended Reports (RTCP XR) is a VoIP management control that
defines a set of metrics containing information for assessing VoIP call quality and for
diagnosing problems. RTCP XR (RFC 3611) extends the RTCP reports defined in RFC
3550 by providing additional VoIP metrics. RTCP XR information publishing is implemented
in the device according to <draft-johnston-sipping-rtcp-summary-07>. This draft defines
how a SIP User Agent (UA) publishes the detailed information to a defined collector. RTCP
XR measures VoIP call quality such as packet loss, delay, signal / noise / echo levels,
estimated R-factor, and mean opinion score (MOS). RTCP XR measures these parameters
using metrics as listed in the table below.
Note: RTCP XR is a customer ordered feature and thus, must be included in the
Software License Key installed on the device.
RTCP XR messages containing key call-quality-related metrics are exchanged periodically

(user-defined) between the device and the SIP UA. This allows an analyzer to monitor
these metrics midstream, or a device to retrieve them using SNMP.
You can configure the device to send RTCP XR to an Event State Compositor (ESC)
server or for Gateway calls, to a specific IP Group (using the PublicationIPGroupID ini file
parameter). If you configure it to send RTCP XR to an IP Group, the RTCP XR is sent to
the address configured for the Proxy Set associated with the IP Group.
The device sends RTCP XR in SIP PUBLISH messages. The PUBLISH message contains
the following RTCP XR related header values:
 From and To: Telephone extension number of the user.
 Request-URI: IP address and port of the SEM server when sent to the ESC server.
When sent to an IP Group, the Request-URI value contains the name of the IP Group
as configured by the 'IP Group Name' parameter (IPGroup_Name).
 Event: "vq-rtcpxr"
 Content-Type: "application/vq-rtcpxr"
You can configure the stage of the call at which you want the device to send RTCP XR:
 End of the call.
 Periodically, according to a user-defined interval between consecutive reports.
 (Gateway Application Only) End of a media segment. A media segment is a change in
media, for example, when the coder is changed or when the caller toggles between
two called parties (using call hold/retrieve). The RTCP XR sent at the end of a media
segment contains information only of that segment. For call hold, the device sends
RTCP XR each time the call is placed on hold and each time it is retrieved. In addition,
the Start timestamp in the RTCP XR indicates the start of the media segment; the End
timestamp indicates the time of the last sent periodic RTCP XR (typically, up to 5
seconds before reported segment ends).
Table 38-1: RTCP XR Published VoIP Metrics
Group Metric Name
General Start Timestamp
Version 6.6 423 MP-11x & MP-124

MP-11x & MP-124
Group Metric Name
Stop Timestamp
Call-ID
Local Address (IP, Port & SSRC)
Remote Address (IP, Port & SSRC)
Session Description Payload Type
Payload Description
Sample Rate
Frame Duration
Frame Octets
Frames per Packets
Packet Loss Concealment
Silence Suppression State
Jitter Buffer Jitter Buffer Adaptive
Jitter Buffer Rate
Jitter Buffer Nominal
Jitter Buffer Max
Jitter Buffer Abs Max
Packet Loss Network Packet Loss Rate
Jitter Buffer Discard Rate
Burst Gap Loss Burst Loss Density
Burst Duration
Gap Loss Density
Gap Duration
Minimum Gap Threshold
Delay Round Trip Delay
End System Delay
One Way Delay
Interarrival Jitter
Min Absolute Jitter
Signal
Signal Level
Noise Level
Residual Echo Return Noise
Quality Estimates Listening Quality R
RLQ Est. Algorithm
Conversational Quality R

Group Metric Name
RCQ Est. Algorithm

External R In
Ext. R In Est. Algorithm
External R Out
Ext. R Out Est. Algorithm
MOS-LQ
MOS-LQ Est. Algorithm
MOS-CQ
MOS-CQ Est. Algorithm
QoE Est. Algorithm
Below shows an example of a SIP PUBLISH message sent with RTCP XR and QoE
information:
PUBLISH sip:172.17.116.201 SIP/2.0
Via: SIP/2.0/UDP 172.17.116.201:5060;branch=z9hG4bKac2055925925
Max-Forwards: 70
From: <sip:172.17.116.201>;tag=1c2055916574
To: <sip:172.17.116.201>
CSeq: 1 PUBLISH
Contact: <sip:172.17.116.201:5060>
Allow:
REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUB
SCRIBE,UPDATE
Event: vq-rtcpxr
Expires: 3600
User-Agent: device/<swver>
Content-Type: application/vq-rtcpxr
Content-Length: 1066
VQSessionReport
[email protected]
LocalID: <sip:[email protected]>
RemoteID: <sip:[email protected];user=phone>
OrigID: <sip:[email protected]>
LocalAddr: IP=172.17.116.201 Port=6000 SSRC=0x54c62a13
RemoteAddr: IP=172.17.116.202 Port=6000 SSRC=0x243220dd
LocalGroup:
RemoteGroup:
LocalMAC: 00:90:8f:57:d9:71
LocalMetrics:
Timestamps: START=2015-12-16T20:09:45Z STOP=2015-12-16T20:09:52Z
SessionDesc: PT=8 PD=PCMA SR=8000 FD=20 PLC=3 SSUP=Off
JitterBuffer: JBA=3 JBR=0 JBN=7 JBM=10 JBX=300
PacketLoss: NLR=0.00 JDR=0.00
BurstGapLoss: BLD=0.00 BD=0 GLD=0.00 GD=6325 GMIN=16
Version 6.6 425 MP-11x & MP-124

MP-11x & MP-124
Delay: RTD=0 ESD=11

Signal: SL=-34 NL=-67 RERL=17
QualityEst: RLQ=93 MOSLQ=4.1
MOSCQ=4.10
RemoteMetrics:
Timestamps: START=2015-12-16T20:09:45Z STOP=2015-12-16T20:09:52Z
JitterBuffer: JBA=3 JBR=0 JBN=0 JBM=0 JBX=300
PacketLoss: NLR=0.00 JDR=0.00
BurstGapLoss: BLD=0.00 BD=0 GLD=0.00 GD=0 GMIN=16
Delay: RTD=65535 ESD=0
QualityEst:
DialogID: [email protected];to-
tag=1c1690611502;from-tag=1c2032864069

 To configure RTCP XR:

1. Open the RTP/RTCP Settings page (Configuration tab > VoIP menu > Media >
RTP/RTCP Settings). The RTCP XR parameters are listed under the 'RTCP XR
Settings' group, as shown below:
Figure 38-1: RTCP XR Parameters in RTP/RTCP Settings Page
2. Configure the RTCP XR parameters, as required:

• 'Enable RTCP XR' (VQMonEnable) - enables voice quality monitoring and RTCP
XR.
• 'Minimum Gap Size' (VQMonGMin) - defines the voice quality monitoring -
minimum gap size (number of frames).
• 'Burst Threshold' (VQMonBurstHR) - defines the voice quality monitoring -
excessive burst alert threshold.
• 'Delay Threshold' (VQMonDelayTHR) - defines the voice quality monitoring -
excessive delay alert threshold.
• 'R-Value Delay Threshold' (VQMonEOCRValTHR) - defines the voice quality
monitoring - end of call low quality alert threshold.
• 'RTCP XR Report Mode' (RTCPXRReportMode) - determines whether RTCP XR
reports are sent to the ESC and defines the interval in which they are sent.
• 'RTCP XR Packet Interval' (RTCPInterval) - defines the time interval between
adjacent RTCP reports.
• 'Disable RTCP XR Interval Randomization' (DisableRTCPRandomize) -
determines whether RTCP report intervals are randomized or whether each
report interval accords exactly to the parameter RTCPInterval.
• 'RTCP XR Collection Server' (RTCPXREscIP) - defines the IP address of the
Event State Compositor (ESC). Alternatively, if you want to send the RTCP XR to
a specific IP Group, use the PublicationIPGroupID ini file parameter.
• 'RTCP XR Collection Server Transport Type' (RTCPXRESCTransportType) -
determines the transport layer for outgoing SIP dialogs initiated by the device to
the RTCP XR Collection Server.
3. Click Submit.
4. Reset the device for the settings to take effect.
Version 6.6 427 MP-11x & MP-124

MP-11x & MP-124
38.2 Generating Call Detail Records

The Call Detail Record (CDR) contains vital statistic information on calls made from the
device. The device can be configured to generate and report CDRs for various stages of
the call, including SIP messages and/or media. You can configure when CDRs for a call
are generated, for example, only at the end of the call or only at the start and end of the
call. Once generated, the device sends the CDRs to a user-defined Syslog server.
The CDR Syslog message complies with RFC 3161 and is identified by Facility 17 (local1)
and Severity 6 (Informational).
For CDR in RADIUS format, see 'RADIUS Accounting CDR Attributes' on page 435.
38.2.1 Configuring CDR Reporting

The procedure below describes how to configure CDR reporting.
 To configure CDR reporting:

1. Enable the Syslog feature for sending log messages generated ty the device to a
collecting log message server. For more information, see 'Configuring Syslog' on page
448.
Definitions > Advanced Parameters). The CDR parameters appear under the 'CDR
and Debug' group, as shown below:
Figure 38-2: CDR Parameters in Advanced Parameters Page
3. Configure the parameters as required. For a description of the parameters, see

'Syslog, CDR and Debug Parameters' on page 498.
4. Click Submit.
Note: If the CDR server IP address is not configured, the CDRs are sent to the
Syslog server, configured in 'Configuring Syslog' on page 448.

38.2.2 CDR Field Description

This section describes the CDR fields that are generated by the device.
38.2.2.1 CDR Fields for Gateway/IP-to-IP Application

The CDR fields for the Gateway / IP-to-IP application are listed in the table below.
Table 38-2: CDR Fields for Gateway/IP-to-IP Application
Field Name Description
GWReportType Report type:

 CALL_START
 CALL_CONNECT
 CALL_END
Cid Port number
SessionId SIP session identifier
Trunk Physical trunk number
Note: This field is applicable only to the Gateway application.
BChan Selected B-channel (set to '0', as not applicable)
ConId SIP conference ID
TG Trunk Group ID
EPTyp Endpoint type:
 FXO
 FXS
 EANDM
 ISDN
 CAS
 DAA
 IPMEDIA
 NETANN
 STREAMING
 TRANSPARENT
 MSCML
 VXML
 IP2IP
Orig Call originator:
 LCL (Tel side)
 RMT (IP side)
SourceIp Source IP address
DestIp Destination IP address
Version 6.6 429 MP-11x & MP-124

MP-11x & MP-124
TON Source phone number type

NPI Source phone number plan
SrcPhoneNum Source phone number
SrcNumBeforeMap Source number before manipulation
TON Destination phone number type
NPI Destination phone number plan
DstPhoneNum Destination phone number
DstNumBeforeMap Destination number before manipulation
Durat Call duration
Coder Selected coder
Intrv Packet interval
RtpIp RTP IP address
Port Remote RTP port
TrmSd Initiator of call release (IP, Tel, or Unknown)
TrmReason SIP call termination reason (see 'Release Reasons in CDR' on page 432)
Fax Fax transaction during call
InPackets Number of incoming packets
OutPackets Number of outgoing packets
PackLoss Local packet loss
RemotePackLoss Number of outgoing lost packets
SIPCalld Unique SIP call ID
SetupTime Call setup time
ConnectTime Call connect time
ReleaseTime Call release time
RTPdelay RTP delay
RTPjitter RTP jitter
RTPssrc Local RTP SSRC
RemoteRTPssrc Remote RTP SSRC
RedirectReason Redirect reason
TON Redirection phone number type
NPI Redirection phone number plan

RedirectPhonNum Redirection phone number

MeteringPulses Number of generated metering pulses
SrcHost Source host name

SrcHostBeforeMap Source host name before manipulation
DstHost Destination host name
DstHostBeforeMap Destination host name before manipulation
IPG IP Group description
LocalRtpIp Remote RTP IP address
LocalRtpPort Local RTP port
Amount 0-999999
Data is stored per call and sent in the syslog as follows:
 currency-type: amount multiplier for currency charge (euro or usd)
 recorded-units: for unit charge (1-999999)
Mult 0,001-1000 (in steps of 10)

(See explanation above.)
TrmReasonCategory Termination reason category:
 Calls with duration 0 (i.e., not connected):
 NO_ANSWER - GWAPP_NORMAL_CALL_CLEAR,
GWAPP_NO_USER_RESPONDING,
GWAPP_NO_ANSWER_FROM_USER_ALERTED
 BUSY - GWAPP_USER_BUSY
 NO_RESOURCES -
GWAPP_RESOUUCE_UNAVAILABLE_UNSPECIFIED,
RELEASE_BECAUSE_NO_CONFERENCE_RESOURCES_LEFT
,
RESOURCE_BECAUSE_NO_TRANSCODING_RESOURCES_L
EFT, RELEASE_BECAUSE_GW_LOCKED
 NO_MATCH -
RELEASE_BECAUSE_UNMATCHED_CAPABILITIES
 FORWARDED - RELEASE_BECAUSE_FORWARD
 GENERAL_FAILED - any other reason
 Calls with duration:
 NORMAL_CALL_CLEAR - GWAPP_NORMAL_CALL_CLEAR
 ABNORMALLY_TERMINATED - Anything else
 N/A - Reasons not belonging to above categories
RedirectNumBeforeMap Redirect number before manipulation

SrdId SRD ID name
SIPInterfaceId SIP interface ID
ProxySetId Proxy Set ID
IpProfileId IP Profile ID name
MediaRealmId Media Realm name
Version 6.6 431 MP-11x & MP-124

MP-11x & MP-124
SigTransportType SIP signaling transport type (UDP, TCP, or TLS)

TxRTPIPDiffServ Media IP DiffServ
TxSigIPDiffServ Signaling IP DiffServ
LocalRFactor Local R-factor
RemoteRFactor Remote R-factor
LocalMosCQ Local MOS for conversation quality
RemoteMosCQ Remote MOS for conversation quality
SigSourcePort SIP source port
SigDestPort SIP destination port
MediaType Media type - audio, video, or text
SIPTrmReason SIP call termination reason (BYE, CANCEL, or SIP error codes, e.g., 404)
SipTermDesc Description of SIP termination reason:
 SIP Reason header, if exists, for example: SIP ;cause=200 ;text="Call
completed elsewhere".
 If no SIP Reason header exists, the description is taken from the
reason text, if exists, of the SIP response code, for example: "417
Unknown Resource-Priority".
 If no reason text exists in the SIP response code, the description is
taken from an internal SIP response mapping mechanism. For
example, if the device receives a SIP response "422", it sends in the
CDR "422 Session Interval Too Small method" as the description.
PstnTermReason Q.850 protocol termination reason (0-127).
LatchedRtpIp Remote IP address of the incoming RTP stream that the device "latched"
on to as a result of the RTP latching mechanism for NAT traversal.
LatchedRtpPort Remote RTP port of the incoming RTP stream that the device "latched"
on to as a result of the RTP latching mechanism for NAT traversal.
38.2.2.2 Release Reasons in CDR

The possible reasons for call termination for the Gateway / IP-to-IP application which is
represented in the CDR field TrmReason are listed below:
 "REASON N/A"
 "RELEASE_BECAUSE_NORMAL_CALL_DROP"
 "RELEASE_BECAUSE_DESTINATION_UNREACHABLE"
 "RELEASE_BECAUSE_DESTINATION_BUSY"
 "RELEASE_BECAUSE_NOANSWER"
 "RELEASE_BECAUSE_UNKNOWN_REASON"
 "RELEASE_BECAUSE_REMOTE_CANCEL_CALL"
 "RELEASE_BECAUSE_UNMATCHED_CAPABILITIES"
 "RELEASE_BECAUSE_UNMATCHED_CREDENTIALS"
 "RELEASE_BECAUSE_UNABLE_TO_HANDLE_REMOTE_REQUEST"
 "RELEASE_BECAUSE_NO_CONFERENCE_RESOURCES_LEFT"

 "RELEASE_BECAUSE_CONFERENCE_FULL"
 "RELEASE_BECAUSE_VOICE_PROMPT_PLAY_ENDED"
 "RELEASE_BECAUSE_VOICE_PROMPT_NOT_FOUND"
 "RELEASE_BECAUSE_TRUNK_DISCONNECTED"
 "RELEASE_BECAUSE_RSRC_PROBLEM"
 "RELEASE_BECAUSE_MANUAL_DISC"
 "RELEASE_BECAUSE_SILENCE_DISC"
 "RELEASE_BECAUSE_RTP_CONN_BROKEN"
 "RELEASE_BECAUSE_DISCONNECT_CODE"
 "RELEASE_BECAUSE_GW_LOCKED"
 "RELEASE_BECAUSE_NORTEL_XFER_SUCCESS"
 "RELEASE_BECAUSE_FAIL"
 "RELEASE_BECAUSE_FORWARD"
 "RELEASE_BECAUSE_ANONYMOUS_SOURCE"
 "RELEASE_BECAUSE_IP_PROFILE_CALL_LIMIT"
 "GWAPP_UNASSIGNED_NUMBER"
 "GWAPP_NO_ROUTE_TO_TRANSIT_NET"
 "GWAPP_NO_ROUTE_TO_DESTINATION"
 "GWAPP_CHANNEL_UNACCEPTABLE"
 "GWAPP_CALL_AWARDED_AND "
 "GWAPP_PREEMPTION"
 "PREEMPTION_CIRCUIT_RESERVED_FOR_REUSE"
 "GWAPP_NORMAL_CALL_CLEAR"
 "GWAPP_USER_BUSY"
 "GWAPP_NO_USER_RESPONDING"
 "GWAPP_NO_ANSWER_FROM_USER_ALERTED"
 "MFCR2_ACCEPT_CALL"
 "GWAPP_CALL_REJECTED"
 "GWAPP_NUMBER_CHANGED"
 "GWAPP_NON_SELECTED_USER_CLEARING"
 "GWAPP_INVALID_NUMBER_FORMAT"
 "GWAPP_FACILITY_REJECT"
 "GWAPP_RESPONSE_TO_STATUS_ENQUIRY"
 "GWAPP_NORMAL_UNSPECIFIED"
 "GWAPP_CIRCUIT_CONGESTION"
 "GWAPP_USER_CONGESTION"
 "GWAPP_NO_CIRCUIT_AVAILABLE"
 "GWAPP_NETWORK_OUT_OF_ORDER"
 "GWAPP_NETWORK_TEMPORARY_FAILURE"
 "GWAPP_NETWORK_CONGESTION"
 "GWAPP_ACCESS_INFORMATION_DISCARDED"
 "GWAPP_REQUESTED_CIRCUIT_NOT_AVAILABLE"
 "GWAPP_RESOURCE_UNAVAILABLE_UNSPECIFIED"
 "GWAPP_PERM_FR_MODE_CONN_OUT_OF_S"
Version 6.6 433 MP-11x & MP-124

MP-11x & MP-124
 "GWAPP_PERM_FR_MODE_CONN_OPERATIONAL"
 "GWAPP_PRECEDENCE_CALL_BLOCKED"
• "RELEASE_BECAUSE_PREEMPTION_ANALOG_CIRCUIT_RESERVED_FOR_
REUSE"
• "RELEASE_BECAUSE_PRECEDENCE_CALL_BLOCKED"
 "GWAPP_QUALITY_OF_SERVICE_UNAVAILABLE"
 "GWAPP_REQUESTED_FAC_NOT_SUBSCRIBED"
 "GWAPP_BC_NOT_AUTHORIZED"
 "GWAPP_BC_NOT_PRESENTLY_AVAILABLE"
 "GWAPP_SERVICE_NOT_AVAILABLE"
 "GWAPP_CUG_OUT_CALLS_BARRED"
 "GWAPP_CUG_INC_CALLS_BARRED"
 "GWAPP_ACCES_INFO_SUBS_CLASS_INCONS"
 "GWAPP_BC_NOT_IMPLEMENTED"
 "GWAPP_CHANNEL_TYPE_NOT_IMPLEMENTED"
 "GWAPP_REQUESTED_FAC_NOT_IMPLEMENTED"
 "GWAPP_ONLY_RESTRICTED_INFO_BEARER"
 "GWAPP_SERVICE_NOT_IMPLEMENTED_UNSPECIFIED"
 "GWAPP_INVALID_CALL_REF"
 "GWAPP_IDENTIFIED_CHANNEL_NOT_EXIST"
 "GWAPP_SUSPENDED_CALL_BUT_CALL_ID_NOT_EXIST"
 "GWAPP_CALL_ID_IN_USE"
 "GWAPP_NO_CALL_SUSPENDED"
 "GWAPP_CALL_HAVING_CALL_ID_CLEARED"
 "GWAPP_INCOMPATIBLE_DESTINATION"
 "GWAPP_INVALID_TRANSIT_NETWORK_SELECTION"
 "GWAPP_INVALID_MESSAGE_UNSPECIFIED"
 "GWAPP_NOT_CUG_MEMBER"
 "GWAPP_CUG_NON_EXISTENT"
 "GWAPP_MANDATORY_IE_MISSING"
 "GWAPP_MESSAGE_TYPE_NON_EXISTENT"
 "GWAPP_MESSAGE_STATE_INCONSISTENCY"
 "GWAPP_NON_EXISTENT_IE"
 "GWAPP_INVALID_IE_CONTENT"
 "GWAPP_MESSAGE_NOT_COMPATIBLE"
 "GWAPP_RECOVERY_ON_TIMER_EXPIRY"
 "GWAPP_PROTOCOL_ERROR_UNSPECIFIED"
 "GWAPP_INTERWORKING_UNSPECIFIED"
 "GWAPP_UKNOWN_ERROR"
 "RELEASE_BECAUSE_HELD_TIMEOUT"

38.3 Configuring RADIUS Accounting

The the RADIUS Parameters page allows you to enable RADIUS accounting of SIP calls
by a RADIUS accounting server. The device can send the accounting messages to the
RADIUS server upon call release, call connection and release, or call setup and release.
Notes:
• For RADIUS accounting settings to take effect, you must save the settings to flash
memory with a device reset.
• For a description of the RADIUS accounting parameters, see 'RADIUS
 To configure RADIUS accounting:

1. Open the RADIUS Parameters page (Configuration tab > VoIP menu > SIP
Definitions submenu > RADIUS Parameters Settings).
Figure 38-3: RADIUS Accounting Parameters Page

3. Click Submit.
The table below describes the RADIUS Accounting CDR attributes included in the
communication packets transmitted between the device and a RADIUS server.
Table 38-3: Supported RADIUS Accounting CDR Attributes
Vendor
Attribute Attribute Specific Value
Purpose Example AAA
Number Name Attribute Format
(VSA) No.
Request Attributes
1 user-name - Account number or String 5421385747 Start Acc
calling party number up to 15 Stop Acc
or blank digits
long
4 nas-ip- - IP address of the Numeric 192.168.14.43 Start Acc
address requesting device Stop Acc
6 service-type - Type of service Numeric 1: login Start Acc
requested Stop Acc
26 h323- 1 SIP call identifier Up to 32 - Start Acc
incoming- octets Stop Acc
conf-id
Version 6.6 435 MP-11x & MP-124

MP-11x & MP-124
Vendor
Purpose Example AAA
(VSA) No.
26 h323-remote- 23 IP address of the Numeric - Stop Acc

address remote gateway
26 h323-conf-id 24 H.323/SIP call Up to 32 - Start Acc
identifier octets Stop Acc
26 h323-setup- 25 Setup time in NTP String - Start Acc
time format 1 Stop Acc
26 h323-call- 26 The call’s originator: String Answer, Start Acc
origin Answering (IP) or Originate etc Stop Acc
Originator (PSTN)
26 h323-call- 27 Protocol type or String VoIP Start Acc
type family used on this Stop Acc
leg of the call
26 h323- 28 Connect time in NTP String - Stop Acc
connect-time format
26 h323- 29 Disconnect time in String - Stop Acc
disconnect- NTP format
time
26 H323- 30 Q.931 disconnect Numeric - Stop Acc
Disconnect- cause code
Cause
26 h323-gw-id 33 Name of the gateway String SIPIDString Start Acc
Stop Acc
26 sip-call-id 34 SIP Call ID String [email protected] Start Acc
Stop Acc
26 call- 35 The call's terminator: String Yes, No Stop Acc
terminator PSTN-terminated call
(Yes); IP-terminated
call (No).
30 called- - Destination phone String 8004567145 Start Acc
station-id number
31 calling- - Calling Party Number String 5135672127 Start Acc
station-id (ANI) Stop Acc
40 acct-status- - Account Request Numeric 1: start, 2: stop Start Acc
type Type (start or stop) Stop Acc
Note: ‘start’ isn’t
supported on the
Calling Card
application.
41 acct-delay- - No. of seconds tried Numeric 5 Start Acc
time in sending a particular Stop Acc
record
42 acct-input- - Number of octets Numeric - Stop Acc
octets received for that call
duration

Vendor
Purpose Example AAA
(VSA) No.
43 acct-output- - Number of octets sent Numeric - Stop Acc

octets for that call duration
44 acct-session- - A unique accounting String 34832 Start Acc
id identifier - match start Stop Acc
& stop
46 acct-session- - For how many Numeric - Stop Acc
time seconds the user
received the service
47 acct-input- - Number of packets Numeric - Stop Acc
packets received during the
call
48 acct-oputput- - Number of packets Numeric - Stop Acc
packets sent during the call
61 nas-port-type - Physical port type of String 0: Start Acc
device on which the Asynchronous Stop Acc
call is active
Response Attributes
26 h323-return- 103 The reason for failing Numeric 0 Request Stop Acc
code authentication (0 = accepted
ok, other number
failed)
44 acct-session- - A unique accounting String - Stop Acc
id identifier – match
start & stop
Below is an example of RADIUS Accounting, where the non-standard parameters are

preceded with brackets:
Accounting-Request (361)
user-name = 111
acct-session-id = 1
nas-ip-address = 212.179.22.213
nas-port-type = 0
acct-status-type = 2
acct-input-octets = 4841
acct-output-octets = 8800
acct-session-time = 1
acct-input-packets = 122
acct-output-packets = 220
called-station-id = 201
calling-station-id = 202
// Accounting non-standard parameters:
(4923 33) h323-gw-id =
(4923 23) h323-remote-address = 212.179.22.214
(4923 1) h323-ivr-out = h323-incoming-conf-id:02102944 600a1899
3fd61009 0e2f3cc5
(4923 30) h323-disconnect-cause = 22 (0x16)
(4923 27) h323-call-type = VOIP
Version 6.6 437 MP-11x & MP-124

MP-11x & MP-124
(4923 26) h323-call-origin = Originate

(4923 24) h323-conf-id = 02102944 600a1899 3fd61009 0e2f3cc5
38.4 Event Notification using X-Detect Header

The device supports the sending of notifications to a remote party notifying the occurrence
(or detection) of certain events on the media stream. Event detection and notifications is
performed using the SIP X-Detect message header and only when establishing a SIP
dialog.
For supporting some events, certain device configurations need to be performed. The table
below lists the supported event types (and subtypes) and the corresponding device
configurations, if required:
Table 38-4: Supported X-Detect Event Types
Events Type Subtype Required Configuration
CPT SIT-NC SITDetectorEnable = 1

SIT-IC UserDefinedToneDetectorEnable = 1
SIT-VC Note: Ensure that the CPT file is configured with the
required tone type.
SIT-RO
Busy
Reorder
Ringtone
FAX CED (IsFaxUsed ≠ 0) or (IsFaxUsed = 0, and
FaxTransportMode ≠ 0)
modem VxxModemTransportType = 3
PTT voice-start EnableDSPIPMDetectors = 1
voice-end
The device can detect and report the following Special Information Tones (SIT) types from
the PSTN:
 SIT-NC (No Circuit found)
 SIT-IC (Operator Intercept)
 SIT-VC (Vacant Circuit - non-registered number)
 SIT-RO (Reorder - System Busy)
There are additional three SIT tones that are detected as one of the above SIT tones:
 The NC* SIT tone is detected as NC
 The RO* SIT tone is detected as RO
 The IO* SIT tone is detected as VC

The device can map these SIT tones to a Q.850 cause and then map them to SIP 5xx/4xx
responses, using the parameters SITQ850Cause, SITQ850CauseForNC,
SITQ850CauseForIC, SITQ850CauseForVC, and SITQ850CauseForRO.
Table 38-5: Special Information Tones (SITs) Reported by the device
Special Description First Tone Second Tone Third Tone

Information Frequency Frequency Frequency
Tones (SITs) Duration Duration Duration
Name
(Hz) (ms) (Hz) (ms) (Hz) (ms)
NC1 No circuit found 985.2 380 1428.5 380 1776.7 380

IC Operator intercept 913.8 274 1370.6 274 1776.7 380
VC Vacant circuit (non 985.2 380 1370.6 274 1776.7 380
registered number)
RO1 Reorder (system 913.8 274 1428.5 380 1776.7 380
busy)
NC* - 913.8 380 1370.6 380 1776.7 380
RO* - 985.2 274 1370.6 380 1776.7 380
IO* - 913.8 380 1428.5 274 1776.7 380
For example:
INFO sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.33.45.65;branch=z9hG4bKac2042168670
Max-Forwards: 70
From: <sip:[email protected];user=phone>;tag=1c1915542705
To: <sip:[email protected];user=phone>;tag=WQJNIDDPCOKAPIDSCOTG
CSeq: 1 INFO
Contact: <sip:[email protected]>
Supported: em,timer,replaces,path,resource-priority
Content-Type: application/x-detect
Content-Length: 28
Type= CPT
SubType= SIT-IC
The X-Detect event notification process is as follows:
1. For IP-to-Tel or Tel-to-IP calls, the device receives a SIP request message (using the
X-Detect header) that the remote party wishes to detect events on the media stream.
For incoming (IP-to-Tel) calls, the request must be indicated in the initial INVITE and
responded to either in the 183 response (for early dialogs) or in the 200 OK response
(for confirmed dialogs).
2. Once the device receives such a request, it sends a SIP response message (using the
X-Detect header) to the remote party, listing all supported events that can be detected.
The absence of the X-Detect header indicates that no detections are available.
3. Each time the device detects a supported event, the event is notified to the remote
party by sending an INFO message with the following message body:
• Content-Type: application/X-DETECT
• Type = [CPT | FAX | PTT…]
• Subtype = xxx (according to the defined subtypes of each type)
Version 6.6 439 MP-11x & MP-124

MP-11x & MP-124
Below is an example of SIP messages using the X-Detect header:

INVITE sip:[email protected];user=phone SIP/2.0
Max-Forwards: 70
From: "anonymous" <sip:[email protected]>;tag=1c25298
To: <sip:[email protected];user=phone>
CSeq: 1 INVITE
X- Detect: Request=CPT,FAX
SIP/2.0 200 OK
To: <sip:[email protected];user=phone>;tag=1c19282
CSeq: 1 INVITE
X- Detect: Response=CPT,FAX
INFO sip:[email protected];user=phone SIP/2.0
Max-Forwards: 70
To: <sip:[email protected];user=phone>
CSeq: 1 INVITE
X- Detect: Response=CPT,FAX
Content-Type: Application/X-Detect
Content-Length: xxx
Type = CPT
Subtype = SIT
38.5 Querying Device Channel Resources using SIP

OPTIONS
The device reports its maximum and available channel resources in SIP 200 OK responses
upon receipt of SIP OPTIONS messages. The device sends this information in the SIP X-
Resources header with the following parameters:
 telchs: Specifies the total telephone channels and the number of free (available)
telephone channels.
 mediachs: Not applicable.
Below is an example of the X-Resources:
X-Resources: telchs= 8/4;mediachs=0/0
In the example above, "telchs" specifies the number of available channels and the number
of occupied channels (4 channels are occupied and 8 channels are available).

Part IX
Diagnostics
User's Manual 39. Syslog and Debug Recordings
39 Syslog and Debug Recordings

Syslog is an event notification protocol that enables a device to send event notification
messages across IP networks to event message collectors, also known as Syslog servers.
The device contains an embedded Syslog client, which sends error reports / events that it
generates to a remote Syslog server using the IP / UDP protocol. This information is a
collection of error, warning, and system messages that records every internal operation of
the device.
For receiving Syslog messages generated by the device, you can use any of the following
Syslog servers:
 Device's embedded Syslog server: The device provides an embedded Syslog
server, which is accessed through the Web interface. This provides limited Syslog
server functionality.
 Wireshark: Third-party network protocol analyzer (http://www.wireshark.org).
 Third-party, Syslog server: Any third-party Syslog server program that enables
filtering of messages according to parameters such as priority, IP sender address,
time, and date.
39.1 Syslog Message Format

The Syslog message is sent from the device to a Syslog server as an ASCII (American
Standard Code for Information Interchange) message. Syslog uses UDP as its underlying
transport layer mechanism. By default, UDP port 514 is assigned to Syslog, but this can be
changed (see 'Configuring Syslog' on page 448).
Below is an example of a Syslog message:
13:10:57.811 : 10.13.4.12 : NOTICE : [S=235][SID:1034099026] (
lgr_flow)(63 ) UdpTransportObject#0- Adding socket event
for address 10.33.2.42:5060 [Time: 04-19-2012@18:29:39]
Table 39-1: Syslog Message Format Description
Message Item Description
Message Types Syslog generates the following types of messages:

 ERROR: Indicates that a problem has been identified that
requires immediate handling.
 WARNING: Indicates an error that might occur if measures are
not taken to prevent it.
 NOTICE: Indicates that an unusual event has occurred.
 INFO: Indicates an operational message.
 DEBUG: Messages used for debugging.
Notes:
 The INFO and DEBUG messages are required only for advanced
debugging. Therefore, by default, they are not sent by the device.
 When viewing Syslog messages in the Web interface, these
message types are color coded.
Version 6.6 443 MP-11x & MP-124

MP-11x & MP-124
Message Item Description
Message Sequence Number Syslog messages are sequentially numbered in the format
[S=<number>] [S=<number>], for example, "[S=643]".
A skip in the number sequence of messages indicates a loss of
message packets. For example, in the below Syslog message
generation, messages 238 through 300 were not received. In other
words, 63 Syslog messages were lost (the sequential numbers are
indicated below in bold font):
18:38:14. 52 : 10.33.45.72 : NOTICE:
[S=235][SID:1034099026] (lgr_psbrdex)(619) recv
<-- DIGIT(0) Ch:0 OnTime:0 InterTime:100
Direction:0 System:1 [File: Line:-1]
18:38:14. 83 : 10.33.45.72 : NOTICE:
[S=236][SID:1034099026] (lgr_flow)(620)
#0:DIGIT_EV [File: Line:-1]
18:38:14. 83 : 10.33.45.72 : NOTICE:
[S=237][SID:1034099026] (lgr_flow)(621) |
18:38:14.958 : 10.33.45.72 : NOTICE:
[S=301][SID:1034099026] (lgr_flow)(625) |
Log Number Ignore this number; it has been replaced by the Message Sequence
(lgr)(number) Number (described previously).
Session ID Automatically assigned (random), unique session identifier (session-
id / SID) number per call in the CDR of sent Syslog messages and
debug recording packets. This enables you to filter the information
(such as SIP, Syslog, and media) according to the SID. A call
session is considered either as a Tel-to-IP leg or an IP-to-Tel leg,
where each leg is assigned a unique SID.
The benefit of this unique numbering is that it enables you to filter the
information (such as SIP, Syslog, and media) according to a specific
SID.
Note: Forked legs and alternative legs share the same SID.
Message Body Describes the message.
Timestamp When the Network Time Protocol (NTP) is enabled, a timestamp
string [hour:minutes:seconds] is added to all Syslog messages.
39.1.1 Event Representation in Syslog Messages

The Syslog message events that the device sends are represented by unique
abbreviations. An example of an abbreviated event in a Syslog message indicating packet
loss (PL) is shown below:
Apr 4 12:00:12 172.30.1.14 PL:5 [Code:3a002] [CID:3294] [Time:
20:17:00]

The table below lists these unique event abbreviations:

Table 39-2: Syslog Error Name Descriptions
Error Abbreviation Error Name Description
AA Invalid Accumulated Packets Counter

AC Invalid Channel ID
AL Invalid Header Length
AO Invalid Codec Type
AP Unknown Aggregation Payload Type
AR Invalid Routing Flag Received
AT Simple Aggregation Packets Lost
CC Command Checksum Error
CE Invalid Cell Coder Code
CS Command Sequence Error
ES 8 sec Timeout Before Disconnect
HO Host Received Overrun
IA Invalid AMR Payload
IC Invalid CID Error
IG Invalid G723 Code
IP Invalid payload length
IR Invalid RTCP Packet
IS Invalid SID Length
LC Transmitter Received Illegal Command
LF Lost Fax Frames In High Speed Mode
LM Lost Modem Frames In High Speed Mode
MI Misalignment Error
MR Modem Relay Is Not Supported
OR DSP JB Overrun
PH Packet Header Error
PL RTP Packet Loss
RB Counts the number of BFI Frames Received From The Host
RD No Available Release Descriptor
RO RTP Reorder
RP Unknown RTP Payload Type
RS RTP SSRC Error
UF Unrecognized Fax Relay Command
AA Invalid Accumulated Packets Counter
Version 6.6 445 MP-11x & MP-124

MP-11x & MP-124
Error Abbreviation Error Name Description
AC Invalid Channel ID
AL Invalid Header Length
AO Invalid Codec Type
AP Unknown Aggregation Payload Type
AR Invalid Routing Flag Received
39.1.2 Identifying AudioCodes Syslog Messages using Facility Levels

The device’s Syslog messages can easily be identified and distinguished from Syslog
messages from other equipment, by setting its Facility level. The Facility levels of the
device's Syslog messages are numerically coded with decimal values. Facility level may
use any of the "local use" facilities (0 through 7), according to RFC 3164. Implementing
Facility levels is useful, for example, if you collect the device’s as well as other equipments’
Syslog messages on the same server. Therefore, in addition to filtering Syslog messages
according to IP address, the messages can be filtered according to Facility level.
The Facility level is configured using the SyslogFacility ini file parameter, which provides
the following options:
Table 39-3: Syslog Facility Levels
Numerical Value Facility Level
16 (default) local use 0 (local0)

17 local use 1 (local1)
Syslog messages begin with a less-than ("<") character, followed by a number, which is
followed by a greater-than (">") character. This is optionally followed by a single ASCII
space. The number is known as the Priority and represents both the Facility level and the
Severity level. A Syslog message with Facility level 16 is shown below:
Facility: LOCAL0 - reserved for local use (16)

39.1.3 SNMP Alarms in Syslog Messages

SNMP alerts are sent to the Syslog server using the following formats:
 Raised Alarms: RAISE-ALARM: <Alarm Name>; Textual Description: <Textual
Description>; Severity <Alarm Severity>; Source <Alarm Source>; Unique ID: <Alarm
Unique ID >.
If additional information exists in the alarm, then these are also added: Additional
Info1:/ Additional Info2:/ Additional Info3
The Messages’ Severity is as follows:
Table 39-4: Syslog Message Severity
ITU Perceived Severity AudioCodes' Syslog Severity

(SNMP Alarm’s Severity)
Critical RecoverableMsg
Major RecoverableMsg
Minor RecoverableMsg
Warning Notice
Indeterminate Notice
Cleared Notice
 Cleared Alarms: CLEAR-ALARM: <Alarm Name>; Textual Description: <Textual

Description>; Severity <Alarm Severity>; Source <Alarm Source>; Unique ID: <Alarm
Unique ID >; If exists Additional Info1:/ Additional Info2:/ Additional Info3:
Version 6.6 447 MP-11x & MP-124

MP-11x & MP-124
39.2 Configuring Syslog Settings

The procedure below describes how to configure Syslog. This includes defining the Syslog
server address as well as selecting the activities on the device (for example, a parameter
value change) that you want reported to the server.
Notes:
• For configuring CDR reporting, see 'Configuring CDR Reporting' on page 428.
• For viewing Syslog messages in the Web interface, see 'Viewing Syslog
Messages' on page 451.
• For a detailed description on the Syslog parameters, see 'Syslog, CDR and Debug
 To configure Syslog :
1. Open the Syslog Settings page (Configuration tab > System menu > Syslog
Settings).
2. Enable the Syslog feature by setting the 'Enable Syslog' to Enable.

3. Define the Syslog server using the 'Syslog Server IP Address' and 'Syslog Server Port'
parameters.
4. Configure the debug level using the 'Debug Level' parameter.
5. Under the 'Activity Types to Report ...' group, select the activities to report.

39.3 Configuring Debug Recording

The device enables you to activate debug recording and send debug recording packets to
a defined capturing server. When the debug recording is activated, the device duplicates all
messages that are sent and/or received by it and then sends them to an external IP
address. The debug recording can be done for different types of traffic for example,
RTP/RTCP, T.38, and SIP.
Debug recording is used for advanced debugging when you need to analyze internal
messages and signals. Debug recording is also useful for recording network traffic in
environments where hub or port mirroring is unavailable and for recording internal traffic
between two endpoints on the same device.
Note: Debug recording is collected only on the device's OAMP interface.
 To configure and activate debug recording:

1. Open the Logging Settings page (Configuration tab > System menu > Logging >
Logging Settings).
Figure 39-1: Logging Settings Page
2. Configure the debug capturing server using the 'Debug Recording Destination IP' and
'Debug Recording Destination Port' parameters.
3. From the 'Debug Recording Status' drop-down list, select Start to start the debug
recording or Stop to end the recording.
39.4 Filtering Syslog Messages and Debug Recordings

The device can filter Syslog messages and debug recording (DR) packets, sent by the
device to a Syslog server and packet capturing application (such as Wireshark)
respectively. This can be useful to reduce CPU consumption and minimize negative impact
on VoIP performance.
You can configure up to 30 filtering rules, each based on a selected filtering criteria (e.g.,
an IP Group). Each filtering criteria can be configured with a range. For example, you can
filter Syslog messages for IP Groups 1 through 4. For each filter criteria, you can enable or
disable Syslog messages and debug recording.
Debug recording can also be filtered using various filtering criteria such as SIP signaling or
signaling and media.
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 To configure logging filtering rules:

1. Open the Logging Filters Table page (Configuration tab > System menu > Logging
> Logging Filters Table).
Figure 39-2: Logging Filters Table - Add Record Dialog Box
3. Configure the logging filter, as required. See the table below for a description of the
parameters.
4. Click Submit to save your changes.
Notes:
• To configure the Syslog debug level, use the 'Debug Level' parameter (see
'Configuring Syslog' on page 448).
• The Logging Filters table can also be configured using the table ini file parameter,
LoggingFilters.
Table 39-5: Logging Filters Table Parameters Description
Filter Type Defines the filter criteria.

CLI: filter-type  [1] Any (default)
[LoggingFilters_Type]  [3] Trunk Group ID = Filters according to a specified Trunk
Group ID.
 [6] Tel-to-IP = Filters according to a specified Tel-to-IP routing
rule listed in the Outbound IP Routing table.
 [7] IP-to-Tel = Filters according to a specified IP-to-Tel routing
rule listed in the Inbound IP Routing table.
 [8] IP Group = Filters according to a specified IP Group ID listed
in the IP Group table.
 [12] User = Filters according to a specified user defined by
username or user@host.
 [13] IP Trace = Filters according to a specified IP network trace
wireshark-like expression. For a detailed description on
configuring IP traces, see 'Filtering IP Network Traces' on page
451.

Value Defines the value of the selected filtering type in the 'Filter Type'
CLI: value parameter.
[LoggingFilters_Value] The value can be the following:
 A single value
 A range, using a hyphen "-" between the two values, e.g., "1-3"
 Multiple, non-contiguous values, using commas "," between
each value, e.g., "1,3,9"
 FXO/FXS pertaining to a module, using the syntax module
number/port or port, for example:
 "1/2", means module 1, port 2
 "1/[2-4]", means module 1, ports 2 through 4
 Any to indicate all
 For IP trace expressions, see e 'Filtering IP Network Traces' on
page 451
Syslog Enables Syslog messages for the defined logging filter:
[LoggingFilters_Syslog]  [0] Disable (default)
 [1] Enable
Capture Type Enables debug recordings for the defined logging filter and defines
[LoggingFilters_CaptureType] what to record:
 [0] None (default)
 [1] Signaling = Information related to signaling such as SIP
signaling messages, Syslog, and CDR.
 [2] Signaling & Media = Signaling and media
(RTP/RTCP/T.38).
 [3] Signaling & Media & PCM = Signaling, media, and PCM
(voice signals from and to TDM).
39.4.1 Filtering IP Network Traces

You can filter Syslog and debug recording messages for IP network traces, by setting the
'Filter Type' parameter to IP Trace in the Logging Filters table. IP traces are used to record
any IP stream that is not associated with media (RTP/RTCP), according to destination
and/or source IP address, or port and Layer-4 protocol (UDP, TCP or any other IP type as
defined by http://www.iana.com).
When the IP Trace option is selected, only the ‘Value’ parameter is applicable; the 'Syslog'
and 'Capture Type' parameters are not relevant. The 'Value' parameter configures the
Wireshark-like filtering expressions for your IP trace. The following Wireshark-like
expressions are supported:
Supported Wireshark-like Expressions for 'Value' Parameter
Expression Description
ip.src, ip.dst Source and destination IP address

ip.addr IP address - up to two IP addresses can be entered
ip.proto IP protocol type (PDU) entered as an enumeration value
(e.g., 1 is ICMP, 6 is TCP, 17 is UDP)
udp, tcp, icmp, sip, http, https Single expressions for protocol type
udp.port, tcp.port Transport layer
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Expression Description
udp.srcport, tcp.srcport Transport layer for source port

udp.dstport, tcp.dstport Transport layer for destination port
and, &&, ==, <, > Between expressions
Below are examples of configured expressions for the 'Value' parameter:

 udp && ip.addr==10.8.6.55
 ip.src==10.8.6.55 && udp.port>=5000 and udp.port<6000
 ip.dst==10.8.0.1/16
 ip.addr==10.8.6.40
For conditions requiring the "or" / "||" expression, add multiple table rows. For example, the
Wireshark condition "(ip.src == 1.1.1.1 or ip.src == 2.2.2.2) and ip.dst == 3.3.3.3" can be
configured using the following two table row entries:
1. ip.src == 1.1.1.1 and ip.dst == 3.3.3.3
2. ip.src == 2.2.2.2 and ip.dst == 3.3.3.3
Note: If the 'Value' field is left empty, the device will record all IP traffic types.

39.5 Viewing Syslog Messages

You can use the following tools to view the Syslog messages sent by the device:
 Web interface's Message Log page (see below).
 Any third-party Syslog server (e.g., Wireshark).
The procedure below describes how to view Syslog messages in the Web interface.
Notes:
• It's not recommended to keep a Message Log session open for a prolonged
period. This may cause the device to overload. For prolonged (and detailed)
debugging, use an external Syslog server.
• You can select the Syslog messages in this page, and copy and paste them into a
text editor such as Notepad. This text file (txt) can then be sent to AudioCodes
Technical Support for diagnosis and troubleshooting.
 To activate the Web interface's Message Log:

1. Enable Syslog (see 'Configuring Syslog' on page 448).
2. Open the Message Log page (Status & Diagnostics tab > System Status menu >
Message Log); the Message Log page is displayed and the log is activated.
Figure 39-3: Message Log Page
The displayed logged messages are color-coded as follows:

• Yellow - fatal error message
• Blue - recoverable error message (i.e., non-fatal error)
• Black - notice message
 To stop and clear the Message Log:

 Close the Message Log page by accessing any another page in the Web interface.
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39.6 Collecting Debug Recording Messages

To collect debug recording packets, the open source program Wireshark is used.
AudioCodes proprietary plug-in files for Wireshark, which are shipped in your software kit,
are also required.
Notes:
• The default debug recording port is 925. You can change the port in Wireshark
(Edit menu > Preferences > Protocols > AC DR).
• The plug-ins are per major software release and are applicable to Wireshark Ver.
1.62.
• The plug-ins are backward compatible.
• From Wireshark Ver. 99.08, the tpncp.dat file must be located in the folder,
...WireShark\tpncp.
 To install Wireshark and the plug-ins for debug recording:

1. Install Wireshark on your computer. The Wireshark program can be downloaded from
http://www.wireshark.org.
2. Copy the supplied AudioCodes plug-in files to the directory in which you installed
Wireshark, as follows:
Copy this file To this folder

...\dtds\cdr.dtd Wireshark\dtds\
...\plugins\1.6.2\*.dll Wireshark\plugins\1.6.2
...\tpncp\tpncp.dat Wireshark\tpncp
3. Start Wireshark.
4. In the Filter field, type "acdr" (see the figure below) to view the debug recording
messages. Note that the source IP address of the messages is always the OAMP IP
address of the device.

The device adds the header "AUDIOCODES DEBUG RECORDING" to each debug
recording message, as shown below:
Figure 39-4: Wireshark Page
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User's Manual 40. Self-Testing
40 Self-Testing
The device features the following self-testing modes to identify faulty hardware
components:
 Detailed Test (Configurable): This test verifies the correct functioning of the different
hardware components on the device. This test is done when the device is taken out of
service (i.e., not in regular service for processing calls). The test is performed on
startup when initialization of the device completes.
To enable this test, set the ini file parameter, EnableDiagnostics to 1 or 2, and then
reset the device. The Ready and Fail LEDs are lit while this test is running. Upon
completion of the test and if the test fails, the device sends information on the test
results of each hardware component to the Syslog server.
The following hardware components are tested:
• Flash memory - when EnableDiagnostics = 1 or 2
• DSPs - when EnableDiagnostics = 1 or 2
• Physical Ethernet ports - when EnableDiagnostics = 1 or 2
• Analog interfaces - when EnableDiagnostics = 1 or 2
Notes:
• To return the device to regular operation and service, disable the test by setting
the ini file parameter, EnableDiagnostics to 0, and then reset the device.
• While the test is enabled, ignore errors sent to the Syslog server.
 Startup Test (automatic): This hardware test has minor impact in real-time. While
this test is executed, the regular operation of the device is disabled. If an error is
detected, an error message is sent to the Syslog.
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User's Manual 41. Line Testing
41 Line Testing
41.1 FXS Line Testing
The device can test the telephone lines connected to its FXS ports, using the SNMP
acAnalogFxsLineTestTable table. These tests provide various line measurements. In
addition to these tests, a keep-alive test is also done every 100 msec on each of the
analog ports to detect communication problems with the analog equipment and
overheating of the FXS ports.
 Hardware revision number
 Temperature (above or below limit, only if a thermometer is installed)
 Hook state
 Coefficients checksum
 Message waiting indication status
 Ring state
 Reversal polarity state
For MP-118 and MP-124 only, you can also use the following Command Shell commands
to view line status and electrical measurements per FXS port or phone number:
/SIP>LineTesting Port <port number> <test type>
- or -
/SIP>LineTesting Phone <phone number> <test type>
Where <test type> can be one of the following values:
 0 = Line status, which includes the following:
• Hook status – on-hook (0) or off-hook (1)
• Message Waiting Indication (MWI) – off (0) or on (1)
• Ring – off (0) or on (1)
• Reversal polarity – off (0) or on (1)
 Line electrical measurements:
• 1 = DC Voltage Tip-Ring [V]
• 2 = DC Voltage Tip-Ground [V]
• 3 = DC Voltage Ring-Ground [V]
• 4 = AC Voltage Transmit(Tel2IP) [dbm]
• 5 = AC Voltage Receive (IP2Tel) [dbm]
• 6 = AC Voltage Transmit & Receive [dbm]
• 7 = Current [mA]
• 8 = Resistance Tip-Ring [Ohm]
• 9 = Resistance Tip-Ground [Ohm]
• 10 = Resistance Ring-Ground [Ohm]
• 11 = Capacity Tip-Ring [F]
• 12 = Capacity Tip-Ground [F]
• 13 = Capacity Ring-Ground [F]
• 14 = AC Voltage Tip-Ring [V]
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• 15 = AC Voltage Tip-Ground [V]

• 16 = AC Voltage Ring-Ground [V]
• 17 = All the above
Notes:
• Use the Analog Line testing mechanism only for monitoring and never when there
are calls in progress.
• For MP-118, the line status and electrical measurement tests are supported only if
they are done when the grounding reference is relative to the device. However, if
the grounding is done directly to the earth, the tests are supported only on specific
hardware models. For more information, please contact your AudioCodes' sales
representative.
• Line electrical measurements are supported only on certain MP-124 hardware
assemblies. For more information, contact your AudioCodes' sales representative.
41.2 FXO Line Testing

The device can test the telephone lines connected to its FXO ports, using the SNMP
acAnalogFxoLineTestTable table. These tests provide various line measurements. In
addition to these tests (detailed below), a keep-alive test is also done every 100 msec on
each of the analog ports to detect communication problems with the analog equipment.
 Line Current (mA)
 Line Voltage (V)
 Hook (0 = on-hook; 1 = off-hook)
 Ring (0 - Off; 1 - On)
 Line Connected (0 = Disconnected; 1 = Connected)
 Polarity state (0 = Normal; 1 = Reversed, 2 = N\A)
 Line polarity (0 = Positive; 1 = Negative)
 Message Waiting Indication (0 = Off; 1 = On)
Note: Use the Analog Line testing mechanism only for monitoring and never when
there are calls in progress.

User's Manual 42. Testing SIP Signaling Calls
42 Testing SIP Signaling Calls

A simulated endpoint can be configured on the device to test SIP signaling of calls between
it and a remote destination. This feature is useful in that it can remotely verify SIP message
flow without involving the remote end side in the debug process. The SIP test call
simulates the SIP signaling process - call setup, SIP 1xx responses, and through to
completing the SIP transaction with a 200 OK.
The test call sends Syslog messages to a Syslog server, showing the SIP message flow,
DTMF signals, termination reasons, as well as voice quality statistics.
42.1 Configuring Test Call Endpoints

The Test Call table enables you to test the SIP signaling (setup and registration) of calls
and media (DTMF signals) between a simulated phone on the device and a remote
endpoint. These tests involve both incoming and outgoing calls, where the test endpoint
can be configured as the caller or called party. Test calls can be dialed automatically at a
user-defined interval and/or manually when required. The simulated phone and remote
endpoints are defined as SIP URIs (user@host) and the remote destination can be defined
as an IP Group, IP address, or according to an Outbound IP Routing rule. You can also
enable automatic registration of the endpoint.
When a SIP test call is initiated, the device generates a SIP INVITE towards the remote
endpoint (e.g., a SIP proxy server or softswitch). It simulates the SIP call setup process,
managing SIP 1xx responses and completing the SIP transaction with a 200 OK.
Notes:
• By default, you can configure up to five test calls. This maximum can be increased
by installing the relevant Software License Key. For more information, contact
your AudioCodes sales representative.
• The Test Call Endpoint table can also be configured using the table ini file
parameter Test_Call (see 'SIP Test Call Parameters' on page 498).
 To configure test calls:

1. Open the Test Call Table page (Configuration tab > System menu > Test Call >
Test Call Table).
Figure 42-1: General Tab of Test Call Table
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3. Configure the test endpoint parameters as desired. See the table below for a
description of these parameters.
Test Call Table Parameters
General Tab
Endpoint URI Defines the endpoint's URI. This can be defined as a user or user@host.
[Test_Call_Endpoin The device identifies this endpoint only by the URI's user part. The URI's
tURI] host part is used in the SIP From header in REGISTER requests.
The valid value is a string of up to 150 characters. By default, this parameter
is not configured.
Called URI Defines the destination (called) URI (user@host).
[Test_Call_CalledU The valid value is a string of up to 150 characters. By default, this parameter
RI] is not configured.
CLI: called-uri
Route By Defines the type of routing method. This applies to incoming and outgoing
[Test_Call_DestTyp calls.
e]  [0] GW Tel2IP = (Default) Calls are matched by (or routed to) an SRD
and Application type (defined in the SRD and Application Type
parameters below).
 [1] IP Group = Calls are matched by (or routed to) an IP Group ID.
 [2] Dest Address = Calls are matched by (or routed to) an SRD and
application type.
Notes:
 For REGISTER messages, the option [0] cannot be used as the routing
method.
 For REGISTER messages, if option [1] is used, only Server-type IP
Groups can be used.
IP Group ID Defines the IP Group ID to which the test call is sent or from which it is
[Test_Call_IPGroup received.
ID] Notes:
 This parameter is applicable only if option [1] is configured for the 'Route
By' parameter.
 This IP Group is used for incoming and outgoing calls.
Destination Address Defines the destination host. This can be defined as an IP address[:port] or
[Test_Call_DestAd DNS name[:port].
dress] Note: This parameter is applicable only if the 'Route By' parameter is set to
[2] (Dest Address).
Destination Defines the transport type for outgoing calls.
Transport Type  [-1] Not configured (default)
[Test_Call_DestTra  [0] UDP
nsportType]
 [1] TCP
 [2] TLS
Note: This parameter is applicable only if the 'Route By' parameter is set to
[2] (Dest Address).
Application Type Defines the application type for the endpoint.
[Test_Call_Applicat  [0] GW & IP2IP (default)
ionType]

Authentication Tab
Note: These parameters are applicable only if the test endpoint is set to Caller (see the 'Call Party'
parameter).
Auto Register Enables automatic registration of the endpoint. The endpoint can register to
[Test_Call_AutoRe the device itself or to the 'Destination Address' or 'IP Group ID' parameter
gister] settings (see above).
 [0] False (default)
 [1] True
User Name Defines the authentication username.
[Test_Call_UserNa By default, no username is defined.
me]
Password Defines the authentication password.
[Test_Call_Passwo By default, no password is defined.
rd]
Test Settings Tab
Call Party Defines whether the test endpoint is the initiator or receiving side of the test
[Test_Call_CallPart call.
y]  [0] Caller (default)
 [1] Called
Maximum Channels Defines the maximum number of concurrent channels for the test session.
for Session For example, if you have configured an endpoint "101" and you set this
[Test_Call_MaxCha parameter to "3", the device automatically creates three simulated endpoints
nnels] - "101", "102" and "103" (i.e., consecutive endpoint URIs are assigned).
The default is 1.
Call Duration Defines the call duration (in seconds).
[Test_Call_CallDur The valid value is -1 to 100000. The default is 20. A value of 0 means
ation] infinite. A value of -1 means that the parameter value is automatically
calculated according to the values of the 'Calls per Second' and 'Maximum
Channels for Session' parameters.
Note: This parameter is applicable only if 'Call Party' is set to Caller.
Calls per Second Defines the number of calls per second.
[Test_Call_CallsPer Note: This parameter is applicable only if 'Call Party' is set to Caller.
Second]
Test Mode Defines the test session mode.
[Test_Call_TestMo  [0] Once = (Default) The test runs until the lowest value between the
de] following is reached:
 Maximum channels is reached for the test session, configured by
'Maximum Channels for Session'.
 Call duration ('Call Duration') multiplied by calls per second ('Calls
per Second').
 Test duration expires, configured by 'Test Duration'.
 [1] Continuous = The test runs until the configured test duration is
reached. If it reaches the maximum channels configured for the test
session (in the 'Maximum Channels for Session'), it waits until the
configured call duration of a currently established tested call expires
before making the next test call. In this way, the test session stays within
the configured maximum channels.
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Test Duration Defines the test duration (in minutes).

[Test_Call_TestDur The valid value is 0 to 100000. The default is 0 (i.e., unlimited).
ation]
Play Enables playing a user-defined DTMF signal to the answered side of the
[Test_Call_Play] call.
 [1] DTMF
To configure the played DTMF signal, see 'Configuring DTMF Tones for
Test Calls' on page 466.
Notes:
 To configure the DTMF signaling type (e.g., out-of-band or in-band) use
the 'DTMF Transport Type' parameter (see 'Configuring DTMF Transport
Types' on page 181).
 This parameter is applicable only if 'Call Party' is set to Caller.
Schedule Interval Defines the interval (in minutes) between automatic outgoing test calls.
[Test_Call_Schedul The valid value range is 0 to 100000. The default is 0 (i.e., scheduling is
eInterval] disabled).
42.1.1 Starting, Stopping and Restarting Test Calls

The procedure below describes how to start, stop, and restart test calls.
 To start, stop, and restart a test call:

1. In the Test Call table, select the required test call entry; the Actions button appears
above the table.
2. From the Actions drop-down list, choose the required command:
• Dial: starts the test call (this action is applicable only if the test call party is the
caller).
• Drop Call: stops the test call.
• Restart: ends all established calls and then starts the test call session again.
The status of the test call is displayed in the 'Test Status' field of the Test Call table:
 "Idle": test call is not active.
 "Scheduled": test call is planned to run (according to 'Schedule Interval' parameter
settings)
 "Running": test call has been started (i.e., the Dial command was clicked)
 "Receiving": test call has been automatically activated by calls received for the test
call endpoint from the remote endpoint (when all these calls end, the status returns to
"Idle")
 "Terminating": test call is in the process of terminating the currently established calls
(this occurs if the Drop Call command is clicked to stop the test)
 "Done": test call has been successfully completed (or was prematurely stopped by
clicking the Drop Call command)
A more detailed description of this field is displayed below the table when you click the
Show/Hide button (see 'Viewing Test Call Statistics' on page 465.

42.1.2 Viewing Test Call Statistics

In addition to viewing a brief status description of the test call in the 'Test Status' field (as
described in 'Starting, Stopping and Restarting Test Calls' on page 464), you can also view
a more detailed status description which includes test call statistics.
 To view statistics of a test call:

1. Open the Test Call Table page (Configuration tab > System menu > Test Call >
Test Call Table).
2. Select the test call table entry whose call statistics you want to view.
3. Click the Show/Hide button; the call statistics are displayed in the Test Statistics
pane located below the table, as shown in the figure below:
Figure 42-2: Viewing Test Call Statistics
The 'Test Statistics' pane displays the following test session information:
 Elapsed Time: Duration of the test call since it was started (or restarted).
 Active Calls: The number of currently active test calls.
 Call Attempts: The number of calls that were attempted.
 Total Established Calls: The total number of calls that were successfully established.
 Total Failed Attempts: The total number of calls that failed to be established.
 Remote Disconnections Count: Number of calls that were disconnected by the
remote side.
 Average CPS: The average calls per second.
 Test Status: Displays the status (brief description) as displayed in the 'Test Status'
field (see 'Starting, Stopping and Restarting Test Calls' on page 464).
 Detailed Status: Displays a detailed description of the test call status::
• "Idle": The test call is currently not active.
• "Scheduled - Established Calls: <established calls>, ASR: <%>": The test call is
planned to run (according to 'Schedule Interval' parameter settings) and also
shows the following summary of completed test calls:
♦ Total number of test calls that were established.
♦ Number of successfully answered calls out of the total number of calls
attempted (ASR).
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• "Running (Calls: <number of active calls>, ASR: <%>)": The test call has been
started (i.e., the Dial command was clicked) and shows the following:
♦ Number of currently active test calls.
attempted (Answer Seizure Ratio or ASR).
• "Receiving (<number of active calls>)": The test call has been automatically
activated by calls received for this configured test call endpoint from the
configured remote endpoint. When all these calls terminate, the status returns to
"Idle".
• "Terminating (<number of active calls>)": The Drop Call command has been
clicked to stop the test call and the test call is in the process of terminating the
currently active test calls.
• "Done - Established Calls: <established calls>, ASR: <%>": The test call has
been successfully completed (or was prematurely stopped by clicking the Drop
Call command) and shows the following:
♦ Total number of test calls that were established.
attempted (ASR).
Note: On the receiving side, when the first call is accepted in "Idle" state, statistics
are reset.
42.2 Configuring DTMF Tones for Test Calls

By default, no DTMF signal is played to an answered test call (incoming or outgoing).
However, you can enable this per configured test call in the Test Call table (see
'Configuring Test Call Endpoints' on page 461). If enabled, the default DTMF signal that is
played is "3212333". You can change this as described below.
Notes:
• The DTMF signaling type (e.g., out-of-band or in-band) can be configured using
the 'DTMF Transport Type' parameter. For more information, see 'Configuring
DTMF Transport Types' on page 181.
• To generate DTMF tones, the device's DSP resources are required.
 To configure the played DTMF signal to answered test call:

1. Open the Test Call Settings page (Configuration tab > System menu > Test Call >
Test Call Settings).
2. In the 'Test Call DTMF String' field, enter the DTMF string (up to 15 digits).
3. Click Submit.

42.3 Configuring Basic Test Call

The Basic Test Call feature tests incoming Gateway / IP-to-IP calls from a remote SIP
endpoint to a simulated test endpoint on the device. The only required configuration is to
assign a prefix number (test call ID) to the simulated endpoint. All incoming calls with this
called (destination) prefix number is identified as a test call and sent to the simulated
endpoint. The figure below displays a basic test call example.
Figure 42-3: Incoming Test Call Example
 To configure basic call testing:

1. Open the Test Call Settings page (Configuration tab > System menu > Test Call >
Test Call Settings).
Figure 42-4: Test Call Settings Page
2. In the 'Test Call ID' field, enter a prefix for the simulated endpoint.
Notes:
• The Basic Test Call feature tests incoming calls only and is initiated only upon
receipt of incoming calls with the configured prefix.
• For a full description of this parameter, see 'SIP Test Call Parameters' on page
498.
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42.4 Test Call Configuration Examples

Below are a few examples of test call configurations.
 Single Test Call Scenario: This example describes the configuration of a simple test
call scenario that includes a single test call between a simulated test endpoint on the
device and a remote endpoint.
Figure 42-5: Single Test Call Example
• Test Call table configuration:

♦ Endpoint URI: "101"
♦ Called URI: "201"
♦ Route By: Dest Address
♦ Destination Address: "10.30.40.01"
♦ Call Party: Caller
♦ Test Mode: Once (default)
Alternatively, if you want to route the test call using the Outbound IP Routing table for
the Gateway / IP-to-IP application, configure the following:
♦ Endpoint URI: [email protected]
♦ Route By: GW Tel2IP
♦ Called URI: [email protected]
• Outbound IP Routing table configuration:
♦ Dest. Phone Prefix: 201 (i.e., the Called URI user-part)
♦ Source Phone Prefix: 101 (i.e., the Endpoint URI user-part)
♦ Dest. IP Address: 10.30.40.1
 Batch Test Call Scenario: This example describes the configuration of a batch test
call setup for scheduled and continuous call testing of multiple endpoints. The test call
is done between two AudioCodes devices - Device A and Device B - with simulated
test endpoints. This eliminates the need for phone users, who would otherwise need
to answer and end calls many times for batch testing. The calls are initiated from
Device A, where Device B serves as the remote answering endpoint.

Figure 42-6: Batch Test Call Example
• Test Call table configuration at Device A:

♦ Called URI: "201"
♦ Destination Address: "10.13.4.12"
♦ Maximum Channels for Session: "3" (this setting configures three endpoints -
"101", "102" and "103)
♦ Call Duration: "5" (seconds)
♦ Calls per Sec: "1"
♦ Test Mode: Continuous
♦ Test Duration: "3" (minutes)
♦ Schedule Interval: "180" (minutes)
• Test Call table configuration at Device B:
♦ Maximum Channels for Session: "3" (this setting configures three endpoints -
"201", "202" and "203)
 Registration Test Call Scenario: This example describes the configuration for testing
the registration and authentication (i.e., username and password) process of a
simulated test endpoint on the device with an external proxy/registrar server. This is
useful, for example, for verifying that endpoints located in the LAN can register with an
external proxy and subsequently, communicate with one another.
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Figure 42-7: Test Call Registration Example
This example assumes that you have configured your device for communication
between LAN phone users such as IP Groups to represent the device (10.13.4.12)
and the proxy server, and IP-to-IP routing rules to route calls between these IP
Groups.
♦ Called URI: "itsp"
♦ Destination Address: "10.13.4.12" (this is the IP address of the device itself)
♦ Auto Register: Enable
♦ User Name: "testuser"
♦ Password: "12345"

Part X
Appendix
User's Manual 43. Dialing Plan Notation for Routing and Manipulation
43 Dialing Plan Notation for Routing and

Manipulation
The device supports flexible dialing plan notations for denoting the prefix and/or suffix
source and/or destination numbers and SIP URI user names in the routing and
manipulation tables.
Table 43-1: Dialing Plan Notations for Prefixes and Suffixes
x (letter "x") Denotes any single digit.

# (pound symbol)  When used at the end of a prefix, it denotes the end of a number. For
example, 54324# represents a 5-digit number that starts with the digits
54324.
 When used anywhere else in the number (not at the end), it is part of the
number (pound key). For example, 3#45 represents the prefix number
3#45.
 To denote the pound key when it appears at the end of the number, the
pound key must be enclosed in square brackets. For example, 134[#]
represents any number that starts with 134#.
* (asterisk symbol)  When used on its own, it denotes any number.
 When used as part of a number, it denotes the asterisk key.
$ (dollar sign) Denotes an empty prefix for incoming IP calls that do not have a user part in
the Request-URI, or for incoming Tel calls that do not have a called or
calling number. This is used for the following matching criteria:
 Source and Destination Phone Prefix
 Source and Destination Username
 Source and Destination Calling Name Prefix
Range of Digits
Notes:
 Dial plans denoting a prefix that is a range must be enclosed in square brackets, e.g., [4-8] or
23xx[456].
 Dial plans denoting a prefix that is not a range is not enclosed, e.g., 12345#.
 Dial plans denoting a suffix must be enclosed in parenthesis, e.g., (4) and (4-8).
 Dial plans denoting a suffix that include multiple ranges, the range must be enclosed in square
brackets, e.g., (23xx[4,5,6]).
 An example for entering a combined prefix and suffix dial plan - assume you want to match a rule
whose destination phone prefix is 4 to 8, and suffix is 234, 235, or 236. The entered value would
be the following: [4-8](23[4,5,6]).
[n-m] or (n-m) Represents a range of numbers, for example:
 To depict numbers from 5551200 to 5551300:
 Prefix: [5551200-5551300]#
 Suffix: (5551200-5551300)
 To depict numbers from 123100 to 123200:
 Prefix: 123[100-200]
 Suffix: (123[100-200])
 To depict prefix and suffix numbers together:
 03(100): for any number that starts with 03 and ends with 100.
 [100-199](100,101,105): for a number that starts with 100 to 199 and
ends with 100, 101 or 105.
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 03(abc): for any number that starts with 03 and ends with abc.
 03(5xx): for any number that starts with 03 and ends with 5xx.
 03(400,401,405): for any number that starts with 03 and ends with
400 or 401 or 405.
Notes:
 The value n must be less than the value m.
 Only numerical ranges are supported (not alphabetical letters).
 For suffix ranges, the starting (n) and ending (m) numbers in the range
must have the same number of digits. For example, (23-34) is correct,
but (3-12) is not.
[n,m,...] or (n,m,...) Represents multiple numbers. For example, to depict a one-digit number
starting with 2, 3, 4, 5, or 6:
 Prefix: [2,3,4,5,6]#
 Suffix: (2,3,4,5,6)
 Prefix with Suffix: [2,3,4,5,6](8,7,6) - prefix is denoted in square brackets;
suffix in parenthesis
For prefix only, the notations d[n,m]e and d[n-m]e can also be used:
 To depict a five-digit number that starts with 11, 22, or 33:
[11,22,33]xxx#
 To depict a six-digit number that starts with 111 or 222: [111,222]xxx#
[n1-m1,n2- Represents a mixed notation of single numbers and multiple ranges. For
m2,a,b,c,n3-m3] or example, to depict numbers 123 to 130, 455, 766, and 780 to 790:
(n1-m1,n2-  Prefix: [123-130,455,766,780-790]
m2,a,b,c,n3-m3)  Suffix: (123-130,455,766,780-790)
Note: The ranges and the single numbers used in the dial plan must have
the same number of digits. For example, each number range and single
number in the dialing plan example above consists of three digits.
Note: When configuring phone numbers or prefixes in the Web interface, enter them
only as digits without any other characters. For example, if you wish to enter the
phone number 555-1212, it must be entered as 5551212 without the hyphen (-). If the
hyphen is entered, the entry is invalid.

User's Manual 44. Configuration Parameters Reference
44 Configuration Parameters Reference

The device's configuration parameters, default values, and their descriptions are
documented in this section.
Note: Parameters and values enclosed in square brackets [...] represent the ini file
parameters and their enumeration values.
44.1 Networking Parameters

This subsection describes the device's networking parameters.
44.1.1 Ethernet Parameters

The Ethernet parameters are described in the table below.
Table 44-1: Ethernet Parameters
EMS: Physical Configuration Defines the Ethernet connection mode type.

[EthernetPhyConfiguration]  [0] = 10Base-T half-duplex (Not applicable)
 [1] = 10Base-T full-duplex
 [2] = 100Base-TX half-duplex
 [3] = 100Base-TX full-duplex
 [4] = (Default) Auto-negotiate
Note: For this parameter to take effect, a device reset is required.
44.1.2 Multiple VoIP Network Interfaces and VLAN Parameters

The IP network interfaces and VLAN parameters are described in the table below.
Table 44-2: IP Network Interfaces and VLAN Parameters
Multiple Interface Table

Web: Multiple Interface Table This table parameter configures the Multiple Interface table. The
EMS: IP Interface Settings format of this parameter is as follows:
[InterfaceTable] [InterfaceTable]
InterfaceTable_UnderlyingInterface;
[\InterfaceTable]
For example:
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InterfaceTable 0 = 0, 0, 192.168.85.14, 16, 0.0.0.0, 1,
Management;
InterfaceTable 1 = 2, 0, 200.200.85.14, 24, 0.0.0.0, 200,
Control;
InterfaceTable 2 = 1, 0, 211.211.85.14, 24, 211.211.85.1, 211,
Media;
The above example, configures three network interfaces
(OAMP, Control, and Media).
Notes:
 For a description of this parameter, see 'Configuring IP
Network Interfaces' on page 124.
Single IP Network Parameters
Web: IP Address Defines the device's source IP address of the operations,
EMS: Local IP Address administration, maintenance, and provisioning (OAMP) interface
[LocalOAMIPAddress] when operating in a single interface scenario without a Multiple
Interface table.
The default is 0.0.0.0.
Note: For this parameter to take effect, a device reset is
required.
Web: Subnet Mask Defines the device's subnet mask of the OAMP interface when
EMS: OAM Subnet Mask operating in a single interface scenario without a Multiple
[LocalOAMSubnetMask] Interface table.
The default subnet mask is 0.0.0.0.
required.
Web: Default Gateway Address Defines the Default Gateway of the OAMP interface when
EMS: Local Def GW operating in a single interface scenario without a Multiple
[LocalOAMDefaultGW] Interface table.
VLAN Parameters
Web/EMS: VLAN Mode Enables VLANs tagging (IEEE 802.1Q).
[VLANMode]  [0] Disable (default)
 [1] Enable
Notes:
 To operate with multiple network interfaces, VLANs must be
enabled.
 VLANs are available only when booting the device from
flash. When booting using BootP/DHCP protocols, VLANs
are disabled to allow easier maintenance access. In this
scenario, multiple network interface capabilities are
unavailable.
Web/EMS: Native VLAN ID Defines the Native VLAN ID. This is the VLAN ID to which
[VLANNativeVLANID] untagged incoming traffic is assigned. Outgoing packets sent to
this VLAN are sent only with a priority tag (VLAN ID = 0).
When the Native VLAN ID is equal to one of the VLAN IDs listed
in the Multiple Interface table (and VLANs are enabled),
untagged incoming traffic is considered as incoming traffic for
that interface. Outgoing traffic sent from this interface is sent
with the priority tag (tagged with VLAN ID = 0).

When the Native VLAN ID is different to any value in the 'VLAN
ID' column in the table, untagged incoming traffic is discarded
and all outgoing traffic is tagged.
The default Native VLAN ID is 1.
Note: If this parameter is not configured (i.e., default is 1) and
one of the interfaces has a VLAN ID set to 1, this interface is
still considered the ‘Native’ VLAN. If you do not wish to have a
‘Native’ VLAN ID and want to use VLAN ID 1, set this parameter
to a value other than any VLAN ID in the table.
[EnableNTPasOAM] Defines the application type for Network Time Protocol (NTP)
services.
 [1] = OAMP (default)
 [0] = Control
required.
[VLANSendNonTaggedOnNative] Determines whether to send non-tagged packets on the native
VLAN.
 [0] = (Default) Sends priority tag packets.
 [1] = Sends regular packets (with no VLAN tag).
required.
44.1.3 Routing Parameters

The IP network routing parameters are described in the table below.
Table 44-3: IP Network Routing Parameters
Web: Disable ICMP Determines whether the device accepts or ignores ICMP Redirect
Redirects messages.
[DisableICMPRedirects]  [0] Disable = (Default) ICMP Redirect messages are handled by the
device.
 [1] Enable = ICMP Redirect messages are ignored.
Static IP Routing Table
Web/EMS: IP Routing Defines up to 30 static IP routing rules for the device. These rules can be
Table associated with IP interfaces defined in the Multiple Interface table
[StaticRouteTable] (InterfaceTable parameter). The routing decision for sending the
outgoing IP packet is based on the source subnet/VLAN. If not
associated with an IP interface, the static IP rule is based on destination
IP address.
When the destination of an outgoing IP packet does not match one of the
subnets defined in the Multiple Interface table, the device searches this
table for an entry that matches the requested destination host/network. If
such an entry is found, the device sends the packet to the indicated
router (i.e., next hop). If no explicit entry is found, the packet is sent to
the default gateway according to the source interface of the packet (if
defined).
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[ StaticRouteTable ]
FORMAT StaticRouteTable_Index = StaticRouteTable_InterfaceName,
StaticRouteTable_Destination, StaticRouteTable_PrefixLength,
StaticRouteTable_Gateway, StaticRouteTable_Description;
[ \StaticRouteTable ]
Note: For a description of this parameter, see 'Configuring Static IP
Routing' on page 135.
44.1.4 Quality of Service Parameters

The Quality of Service (QoS) parameters are described in the table below.
Table 44-4: QoS Parameters
Layer-2 Class Of Service (CoS) Parameters (VLAN Tag Priority Field)

Web: Network Priority Defines the VLAN priority (IEEE 802.1p) for Network
EMS: Network Service Class Priority Class of Service (CoS) content.
[VLANNetworkServiceClassPriority] The valid range is 0 to 7. The default is 7.
Web: Media Premium Defines the VLAN priority (IEEE 802.1p) for the
EMS: Premium Service Class Media Priority Premium CoS content and media traffic.
Priority The valid range is 0 to 7. The default is 6.
[VLANPremiumServiceClassMediaPriority]
Web: Control Premium Priority Defines the VLAN priority (IEEE 802.1p) for the
EMS: Premium Service Class Control Priority Premium CoS content and control traffic.
[VLANPremiumServiceClassControlPriority] The valid range is 0 to 7. The default is 6.
Web: Gold Priority Defines the VLAN priority (IEEE 802.1p) for the
EMS: Gold Service Class Priority Gold CoS content.
[VlanGoldServiceClassPriority] The valid range is 0 to 7. The default is 4.
Web: Bronze Priority Defines the VLAN priority (IEEE 802.1p) for the
EMS: Bronze Service Class Priority Bronze CoS content.
[VLANBronzeServiceClassPriority] The valid range is 0 to 7. The default is 2.
Layer-3 Class of Service (TOS/DiffServ) Parameters
Web: Network QoS Defines the Differentiated Services (DiffServ) value
EMS: Network Service Class Diff Serv for Network CoS content.
[NetworkServiceClassDiffServ] The valid range is 0 to 63. The default is 48.
Note: For this parameter to take effect, a device
reset is required.
Web: Media Premium QoS Defines the DiffServ value for Premium Media CoS
EMS: Premium Service Class Media Diff Serv content (only if IPDiffServ is not set in the selected
[PremiumServiceClassMediaDiffServ] IP Profile).
The valid range is 0 to 63. The default is 46.
Note: The value for the Premium Control DiffServ is
determined by the following (according to priority):
 IPDiffServ value in the selected IP Profile
(IPProfile parameter).

 PremiumServiceClassMediaDiffServ.
Web: Control Premium QoS Defines the DiffServ value for Premium Control CoS
EMS: Premium Service Class Control Diff Serv content (Call Control applications) - only if
[PremiumServiceClassControlDiffServ] ControlIPDiffserv is not set in the selected IP Profile.
Notes:
 The value for the Premium Control DiffServ is
determined by the following (according to
priority):
 SiglPDiffserv value in the selected IP Profile
(IPProfile parameter).
 PremiumServiceClassControlDiffServ.
 The same value must be configured for this
parameter and the parameter MLPPDiffServ.
 Outgoing calls are tagged according to this
parameter.
Web: Gold QoS Defines the DiffServ value for the Gold CoS content
EMS: Gold Service Class Diff Serv (Streaming applications).
[GoldServiceClassDiffServ] The valid range is 0 to 63. The default is 26.
Web: Bronze QoS Defines the DiffServ value for the Bronze CoS
EMS: Bronze Service Class Diff Serv content (OAMP applications).
[BronzeServiceClassDiffServ] The valid range is 0 to 63. The default is 10.
44.1.5 NAT and STUN Parameters

The Network Address Translation (NAT) and Simple Traversal of UDP through NAT
(STUN) parameters are described in the table below.
Table 44-5: NAT and STUN Parameters
STUN Parameters
Web: Enable STUN Enables Simple Traversal of UDP through NATs (STUN).
EMS: STUN Enable  [0] Disable (default)
[EnableSTUN]  [1] Enable
When enabled, the device functions as a STUN client and
communicates with a STUN server located in the public
Internet. STUN is used to discover whether the device is located
behind a NAT and the type of NAT. It is also used to determine
the IP addresses and port numbers that the NAT assigns to
outgoing signaling messages (using SIP) and media streams
(using RTP, RTCP and T.38). STUN works with many existing
NAT types and does not require any special behavior from
them.
Notes:
 For defining the STUN server domain name, use the
parameter STUNServerDomainName.
 For more information on STUN, see Configuring STUN on
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page 148.
Web: STUN Server Primary IP Defines the IP address of the primary STUN server.
EMS: Primary Server IP The valid range is the legal IP addresses. The default is 0.0.0.0.
[STUNServerPrimaryIP]
required.
Web: STUN Server Secondary IP Defines the IP address of the secondary STUN server.
EMS: Secondary Server IP The valid range is the legal IP addresses. The default is 0.0.0.0.
[STUNServerSecondaryIP]
required.
[STUNServerDomainName] Defines the domain name for the Simple Traversal of User
Datagram Protocol (STUN) server's address (used for retrieving
all STUN servers with an SRV query). The STUN client can
perform the required SRV query to resolve this domain name to
an IP address and port, sort the server list, and use the servers
according to the sorted list.
Notes:
 Use either the STUNServerPrimaryIP or the
STUNServerDomainName parameter, with priority to the first
one.
NAT Parameters
NAT Mode Enables the NAT feature for media when the device
disable-NAT-traversal communicates with UAs located behind NAT.
[NATMode]  [0] Auto-Detect = NAT is performed only if necessary. If the
UA is identified as being located behind NAT, the device
sends the media packets to the public IP address:port
obtained from the source address of the first media packet
received from the UA. Otherwise, the packets are sent using
the IP address:port obtained from the address in the first
received SIP message. Note that if the SIP session is
established (ACK) and the device (not the UA) sends the
first packet, it sends it to the address obtained from the SIP
message and only after the device receives the first packet
from the UA, does it determine whether the UA is behind
NAT.
 [1] NAT Is Not Used = (Default) NAT feature is disabled.
The device always sends the media packets to the remote
UA using the IP address:port obtained from the first
received SIP message.
 [2] NAT Is Used = NAT is always performed. The device
always sends the media packets to the remote UA using the
source address obtained from the first media packet from
the UA. In this mode, the device does not send any packets
until it receives the first packet from the UA (in order to
obtain the IP address).
Web: NAT IP Address Defines the global (public) IP address of the device to enable
EMS: Static NAT IP Address static NAT between the device and the Internet.
[StaticNatIP] Note: For this parameter to take effect, a device reset is
required.
EMS: Binding Life Time Defines the default NAT binding lifetime in seconds. STUN
[NATBindingDefaultTimeout] refreshes the binding information after this time expires.

The valid range is 0 to 2,592,000. The default is 30.
required.
[EnableIPAddrTranslation] Enables IP address translation for RTP, RTCP, and T.38
packets.
 [0] = Disable IP address translation.
 [1] = (Default) Enable IP address translation.
 [2] = Enable IP address translation for RTP Multiplexing
(ThroughPacket™).
 [3] = Enable IP address translation for all protocols (RTP,
RTCP, T.38 and RTP Multiplexing).
When enabled, the device compares the source IP address of
the first incoming packet to the remote IP address stated in the
opening of the channel. If the two IP addresses don't match, the
NAT mechanism is activated. Consequently, the remote IP
address of the outgoing stream is replaced by the source IP
address of the first incoming packet.
Notes:
 The NAT mechanism must be enabled for this parameter to
take effect (parameter NATMode).
 For information on RTP Multiplexing, see RTP Multiplexing
(ThroughPacket).
[EnableUDPPortTranslation] Enables UDP port translation.
 [0] = (Default) Disables UDP port translation.
 [1] = Enables UDP port translation. The device compares
the source UDP port of the first incoming packet to the
remote UDP port stated in the opening of the channel. If the
two UDP ports don't match, the NAT mechanism is
activated. Consequently, the remote UDP port of the
outgoing stream is replaced by the source UDP port of the
first incoming packet.
Notes:
 The NAT mechanism and the IP address translation must be
enabled for this parameter to take effect (i.e., parameter
NATMode and the parameter EnableIpAddrTranslation to 1).
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44.1.6 NFS Parameters

The Network File Systems (NFS) configuration parameters are described in the table
below.
Table 44-6: NFS Parameters
[NFSBasePort] Defines the start of the range of numbers used for local UDP ports used
by the NFS client. The maximum number of local ports is maximum
channels plus maximum NFS servers.
NFS Table
Web: NFS Table This table parameter defines up to 16 NFS file systems so that the
EMS: NFS Settings device can access a remote server's shared files and directories for
[NFSServers] loading cmp, ini, and auxiliary files (using the Automatic Update
mechanism).
The format of this table ini file parameter is as follows:
[NFSServers]
FORMAT NFSServers_Index = NFSServers_HostOrIP,
NFSServers_RootPath, NFSServers_NfsVersion,
NFSServers_AuthType, NFSServers_UID, NFSServers_GID,
NFSServers_VlanType;
[\NFSServers]
For example:
NFSServers 1 = 101.1.13, /audio1, 3, 1, 0, 1, 1;
Note: For a detailed description of this table, see 'Configuring NFS
Settings' on page 145.
44.1.7 DNS Parameters

The Domain name System (DNS) parameters are described in the table below.
Table 44-7: DNS Parameters
Internal DNS Table

Web: Internal DNS Table This table parameter defines the internal DNS table for resolving host
EMS: DNS Information names into IP addresses. Up to four different IP addresses (in dotted-
[DNS2IP] decimal notation) can be assigned to a host name.
[Dns2Ip]
FORMAT Dns2Ip_Index = Dns2Ip_DomainName,
Dns2Ip_FirstIpAddress, Dns2Ip_SecondIpAddress,
Dns2Ip_ThirdIpAddress, Dns2Ip_FourthIpAddress;
[\Dns2Ip]
For example:
Dns2Ip 0 = DnsName, 1.1.1.1, 2.2.2.2, 3.3.3.3, 4.4.4.4;
Note: For a detailed description of this table parameter, see 'Configuring
the Internal DNS Table' on page 142.
Internal SRV Table
Web: Internal SRV Table This table parameter defines the internal SRV table for resolving host

EMS: DNS Information names into DNS A-Records. Three different A-Records can be assigned
[SRV2IP] to a host name. Each A-Record contains the host name, priority, weight,
and port. The format of this parameter is as follows:
[SRV2IP]
FORMAT SRV2IP_Index = SRV2IP_InternalDomain,
SRV2IP_TransportType, SRV2IP_Dns1, SRV2IP_Priority1,
SRV2IP_Weight1, SRV2IP_Port1, SRV2IP_Dns2, SRV2IP_Priority2,
SRV2IP_Weight2, SRV2IP_Port2, SRV2IP_Dns3, SRV2IP_Priority3,
SRV2IP_Weight3, SRV2IP_Port3;
[\SRV2IP]
For example:
SRV2IP 0 =
SrvDomain,0,Dnsname1,1,1,500,Dnsname2,2,2,501,$$,0,0,0;
Note: For a detailed description of this table parameter, see 'Configuring
the Internal SRV Table' on page 143.
44.1.8 DHCP and LLDP Parameters

The Dynamic Host Control Protocol (DHCP) and LLDP parameters are described in the
table below.
Table 44-8: DHCP Parameters
LLDP Parameters
[EnableLLDP] Enables the device to use the discovery protocol, Link Layer
Discovery Protocol (LLDP) to obtain (over the Layer-2 data link layer)
a VLAN ID for its OAMP interface (per IEEE 802.1, IEEE 802.3 and
TR-41) upon device startup (reset or power up).
 [0] = (Default) Disabled
 [1] = Enabled
For more information on LLDP, see Section VLAN ID Discovery
using LLDP on page 391.
DHCP Parameters
Web: Enable DHCP Enables Dynamic Host Control Protocol (DHCP) functionality.
EMS: DHCP Enable  [0] Disable (default)
[DHCPEnable]  [1] Enable
After the device powers up, it attempts to communicate with a BootP
server. If a BootP server does not respond and DHCP is enabled,
then the device attempts to obtain its IP address and other
networking parameters from the DHCP server.
Notes:
 After you enable the DHCP server, do the following:
a. Enable DHCP and save the configuration.
b. Perform a cold reset using the device's hardware reset button
(soft reset using the Web interface doesn't trigger the
BootP/DHCP procedure and this parameter reverts to
'Disable').
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 Throughout the DHCP procedure, the BootP/TFTP application

must be deactivated; otherwise the device receives a response
from the BootP server instead of from the DHCP server.
 This parameter is a special 'Hidden' parameter. Once defined and
saved in flash memory, its assigned value doesn't revert to its
default even if the parameter doesn't appear in the ini file.
[DhcpOption160Support] Enables the use of DHCP Option 160.
 [0] = (Default) Disable
 [1] = Enable
For more information, see "Provisioning the Device using DHCP
Option 160" on page 394.
Note: For the parameter to take effect, a device reset is required.
EMS: DHCP Speed Factor Defines the DHCP renewal speed.
[DHCPSpeedFactor]  [0] = Disable
 [1] = (Default) Normal
 [2] to [10] = Fast
When set to 0, the DHCP lease renewal is disabled. Otherwise, the
renewal time is divided by this factor. Some DHCP-enabled routers
perform better when set to 4.
[DHCPRequestTFTPParams] Determines whether the device includes DHCP options 66 and 67 in
DHCP Option 55 (Parameter Request List) for requesting the DHCP
server for TFTP provisioning parameters.
 [0] = (Default) Disable
 [1] = Enable
44.1.9 NTP and Daylight Saving Time Parameters

The Network Time Protocol (NTP) and daylight saving time parameters are described in
the table below.
Table 44-9: NTP and Daylight Saving Time Parameters
NTP Parameters
Note: For more information on Network Time Protocol (NTP), see 'Simple Network Time Protocol
Support' on page 119.
Web: NTP Server DN/IP Defines the IP address (in dotted-decimal notation or as an FQDN)
EMS: Server IP Address of the NTP server. The advantage of using an FQDN is that multiple
[NTPServerIP] IP addresses can be resolved from the DNS server, providing NTP
server redundancy.
The default IP address is 0.0.0.0 (i.e., internal NTP client is
disabled).
Web: NTP Secondary Server Defines a second NTP server's address as an FQDN or an IP
IP address (in dotted-decimal notation). This NTP is used for
[NTPSecondaryServerIP] redundancy; if the primary NTP server fails, then this NTP server is
used.

The default IP address is 0.0.0.0.
Web: NTP UTC Offset Defines the Universal Time Coordinate (UTC) offset (in seconds)
EMS: UTC Offset from the NTP server.
[NTPServerUTCOffset] The default offset is 0. The offset range is -43200 to 43200.
Web: NTP Update Interval Defines the time interval (in seconds) that the NTP client requests
EMS: Update Interval for a time update.
[NTPUpdateInterval] The default interval is 86400 (i.e., 24 hours). The range is 0 to
214783647.
Note: It is not recommend to set this parameter to beyond one
month (i.e., 2592000 seconds).
Daylight Saving Time Parameters
Web: Day Light Saving Time Enables daylight saving time.
EMS: Mode  [0] Disable (default)
[DayLightSavingTimeEnable]  [1] Enable
Web: Start Time or Day of Defines the date and time when daylight saving begins. This value
Month Start can be configured using any of the following formats:
EMS: Start  Day of year - mm:dd:hh:mm, where:
[DayLightSavingTimeStart]  mm denotes month
 dd denotes date of the month
 hh denotes hour
 mm denotes minutes
For example, "05:01:08:00" denotes daylight saving starting from
May 1 at 8 A.M.
 Day of month - mm:day/wk:hh:mm, where:
 mm denotes month (e.g., 04)
 day denotes day of week (e.g., FRI)
 wk denotes week of the month (e.g., 03)
 hh denotes hour (e.g., 23)
 mm denotes minutes (e.g., 10)
For example, "04:FRI/03:23:00" denotes Friday, the third week of
April, at 11 P.M. The week field can be 1-5, where 5 denotes the
last occurrence of the specified day in the specified month. For
example, "04:FRI/05:23:00" denotes the last Friday of April, at 11
P.M.
Web: End Time or Day of Defines the date and time when daylight saving ends. For a
Month End description of the format of this value, see the
EMS: End DayLightSavingTimeStart parameter.
[DayLightSavingTimeEnd]
Web/EMS: Offset Defines the daylight saving time offset (in minutes).
[DayLightSavingTimeOffset] The valid range is 0 to 120. The default is 60.
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44.2 Management Parameters

This subsection describes the device's Web and Telnet parameters.
44.2.1 General Parameters

The general management parameters are described in the table below.
Table 44-10: General Management Parameters
Web: Web and Telnet This table configures up to ten IP addresses that are permitted to
Access List Table access the device's Web interface and Telnet interfaces. Access from
EMS: Web Access an undefined IP address is denied. When no IP addresses are
Addresses defined in this table, this security feature is inactive (i.e., the device
[WebAccessList_x] can be accessed from any IP address).
The default is 0.0.0.0 (i.e., the device can be accessed from any IP
address).
For example:
WebAccessList_0 = 10.13.2.66
WebAccessList_1 = 10.13.77.7
For a description of this parameter, see 'Configuring Web and Telnet
Access List' on page 75.
[INIPasswordsDisplayType] Defines how passwords are displayed in the ini file.
 [0] = (default) Disable. Passwords are obscured ("encoded"). The
passwords are displayed in the following syntax: $1$<obscured
password> (e.g., $1$S3p+fno=).
 [1] = Enable. All passwords are hidden and replaced by an asterisk
(*).
44.2.2 Web Parameters

The Web parameters are described in the table below.
Table 44-11: Web Parameters
Web: Password Change Interval Defines the duration (in minutes) of the validity of Web login
[WebUserPassChangeInterval] passwords. When this duration expires, the password of the
Web user must be changed.
The valid value is 0 to 100000, where 0 means that the
password is always valid. The default is 1140.
Note: This parameter is applicable only when using the Web
Users table, where the default value of the 'Password Age'
parameter in the Web Users table inherits this parameter's
value.
Web: User inactivity timer Defines the duration (in days) for which a user has not logged in
[UserInactivityTimer] to the Web interface, after which the status of the user becomes
inactive and can no longer access the Web interface. These
users can only log in to the Web interface if their status is
changed (to New or Valid) by a System Administrator or Master
user.

The valid value is 0 to 10000, where 0 means inactive. The
default is 90.
Note: This parameter is applicable only when using the Web
Users table.
Web: Session Timeout Defines the duration (in minutes) of Web inactivity of a logged-in
[WebSessionTimeout] user, after which the user is automatically logged off the Web
interface.
The valid value is 0-100000, where 0 means no timeout. The
default is 15.
Note: This parameter can apply to all users, or per user when
set in the Web Users table.
Web: Deny Access On Fail Count Defines the maximum number of failed login attempts, after
[DenyAccessOnFailCount] which the requesting IP address is blocked.
The valid value range is 0 to 10. The values 0 and 1 mean
immediate block. The default is 3.
Web: Deny Authentication Timer Defines the duration (in seconds) for which login to the Web
EMS: WEB Deny Authentication interface is denied from a specific IP address (for all users)
Timer when the number of failed login attempts has exceeded the
[DenyAuthenticationTimer] maximum. This maximum is defined by the
DenyAccessOnFailCount parameter. Only after this time expires
can users attempt to login from this same IP address.
The valid value is 0 to 100000, where 0 means that login is not
denied regardless of number of failed login attempts. The
default is 60.
Web: Display Login Information Enables display of user's login information on each successful
[DisplayLoginInformation] login attempt.
 [0] = Disable (default)
 [1] = Enable
[EnableMgmtTwoFactorAuthenti Enables Web login authentication using a third-party, smart
cation] card.
 [1] = Enable
When enabled, the device retrieves the Web user’s login
username from the smart card, which is automatically displayed
(read-only) in the Web Login screen; the user is then required to
provide only the login password.
Typically, a TLS connection is established between the smart
card and the device’s Web interface, and a RADIUS server is
implemented to authenticate the password with the username.
Thus, this feature implements a two-factor authentication - what
the user has (the physical card) and what the user knows (i.e.,
the login password).
EMS: HTTPS Port Defines the LAN HTTP port for Web management (default is
[HTTPport] 80). To enable Web management from the LAN, configure the
desired port.
required.
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EMS: Disable WEB Config Determines whether the entire Web interface is read-only.
[DisableWebConfig]  [0] = (Default) Enables modifications of parameters.
 [1] = Web interface is read-only.
When in read-only mode, parameters can't be modified. In
addition, the following pages can't be accessed: 'Web User
Accounts', 'Certificates', 'Regional Settings', 'Maintenance
Actions' and all file-loading pages ('Load Auxiliary Files',
'Software Upgrade Wizard', and 'Configuration File').
Notes:
 To return to read/write after you have applied read-only
using this parameter (set to 1), you need to reboot your
device with an ini file that doesn't include this parameter,
using the AcBootP utility.
[ResetWebPassword] Enables the <device> to restore the default management users:
 Security Administrator user (username "Admin"; password
"Admin")
 Monitor user (username "User"; password "User")
In addition, all other users that may have been configured (in
the Web Users table) are deleted.
 [0] = (Default) Disabled. Currently configured users
(usernames and passwords) are retained.
 [1] = Enabled. Default users are restored (see description
above) and all other configured users are deleted.
Notes:
 For the parameter to take effect, a device reset is required.
 In addition to the ini file (see above), you can also restore the
default user accounts through the following management
platforms:
 SNMP (restores default users and retains other
configured users:
1) Set acSysGenericINILine to
WEBPasswordControlViaSNMP = 1, and reset the
device with a flash burn (set
acSysActionSetResetControl to 1 and
acSysActionSetReset to 1).
2) Change the username and password in the
acSysWEBAccessEntry table. Use the following format:
Username acSysWEBAccessUserName: old/pass/new
Password acSysWEBAccessUserCode:
username/old/new
[ScenarioFileName] Defines the file name of the Scenario file to be loaded to the
device. The file name must have the .dat extension and can be
up to 47 characters. For loading a Scenario using the Web
interface, see Loading a Scenario to the Device on page 57.

[WelcomeMessage] Enables and defines a Welcome message that appears on the

Web Login page for logging in to the Web interface.
[WelcomeMessage ]
FORMAT WelcomeMessage_Index = WelcomeMessage_Text
[\WelcomeMessage]
For Example:
FORMAT WelcomeMessage_Index = WelcomeMessage_Text
WelcomeMessage 1 = "**********************************" ;
WelcomeMessage 2 = "********* This is a Welcome message
***" ;
WelcomeMessage 3 = "**********************************" ;
Notes:
 Each index row represents a line of text in the Welcome
message box. Up to 20 lines (or rows) of text can be defined.
 The configured text message must be enclosed in double
quotation marks (i.e., "...").
 If this parameter is not configured, no Welcome message is
displayed.
Local Users Table
Local Users The table defines management users.
configure system > create- The format of the ini file table parameter is as follows:
users-table [ WebUsers ]
[WebUsers] FORMAT WebUsers_Index = WebUsers_Username,
WebUsers_Password, WebUsers_Status,
WebUsers_PwAgeInterval, WebUsers_SessionLimit,
WebUsers_SessionTimeout, WebUsers_BlockTime,
WebUsers_UserLevel;
[ \WebUsers ]
For more information, see Advanced User Accounts
Configuration on page 69.
44.2.3 Telnet Parameters

The Telnet parameters are described in the table below.
Table 44-12: Telnet Parameters
Web: Embedded Telnet Server Enables the device's embedded Telnet server. Telnet is disabled by
EMS: Server Enable default for security.
[TelnetServerEnable]  [0] Disable (default)
 [1] Enable Unsecured
 [2] Enable Secured (SSL)
Note: Only the primary Web User Account (which has Security
Administration access level) can access the device using Telnet
(see 'Configuring Web User Accounts' on page 66).
Web: Telnet Server TCP Port Defines the port number for the embedded Telnet server.
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EMS: Server Port The valid range is all valid port numbers. The default port is 23.
[TelnetServerPort]
Web: Telnet Server Idle Defines the timeout (in minutes) for disconnection of an idle Telnet
Timeout session. When set to zero, idle sessions are not disconnected.
EMS: Server Idle Disconnect The valid range is any value. The default is 0.
[TelnetServerIdleDisconnect]
44.2.4 SNMP Parameters

The SNMP parameters are described in the table below.
Table 44-13: SNMP Parameters
Web: Enable SNMP Enables SNMP.

[DisableSNMP]  [0] Enable = (Default) SNMP is enabled.
 [1] Disable = SNMP is disabled and no traps are sent.
Defines the device's local (LAN) UDP port used for SNMP
[SNMPPort] Get/Set commands.
The range is 100 to 3999. The default port is 161.
required.
EMS: Keep Alive Trap Port Defines the port to which keep-alive traps are sent.
[KeepAliveTrapPort] The valid range is 0 - 65534. The default is port 162.
[SendKeepAliveTrap] Enables keep-alive traps and sends them every 9/10 of the
time as defined by the NATBindingDefaultTimeout parameter.
 [0] = Disable
 [1] = Enable
required.
Defines the base product system OID.
[SNMPSysOid] The default is eSNMP_AC_PRODUCT_BASE_OID_D.
required.
[SNMPTrapEnterpriseOid] Defines the Trap Enterprise OID.
The default is eSNMP_AC_ENTERPRISE_OID.
The inner shift of the trap in the AcTrap subtree is added to
the end of the OID in this parameter.
required.
[acUserInputAlarmDescription] Defines the description of the input alarm.
[acUserInputAlarmSeverity] Defines the severity of the input alarm.

[AlarmHistoryTableMaxSize] Defines the maximum number of rows in the Alarm History

table. This parameter can be controlled by the Config Global
Entry Limit MIB (located in the Notification Log MIB).
required.
[ActiveAlarmTableMaxSize] Defines the maximum number of currently active alarms that
can be displayed in the Active Alarms table. When the table
reaches this user-defined maximum capacity (i.e., full), the
device sends the SNMP trap event,
acActiveAlarmTableOverflow. If the table is full and a new
alarm is raised by the <device>, the new alarm is not
displayed in the table.
The valid range is 15 (MP-11x) or 20 (MP-124) to 40 (MP-11x)
or 100 (MP-124). The default is 20 (MP-11x) or 60 (MP-124).
For more information on the Active Alarms table, see Viewing
Active Alarms on page 413.
Note:
 For the parameter to take effect, a <device> reset is
required.
 To clear the acActiveAlarmTableOverflow trap, you must
reset the device. The reset also deletes all the alarms in
the Active Alarms table.
No Alarm For Disabled Port Enables the device to not send the SNMP trap
[NoAlarmForDisabledPort] acBoardControllerFailureAlarm, which indicates a "disabled"
(non-configured) telephony port. A disabled port is one that is
not configured at all or that is configured but without a Trunk
Group ID (i.e., Trunk Group ID is 0), in the Endpoint Phone
Number table.
 [0] Disable = (Default) The device sends the SNMP trap for
non-configured ports.
 [1] Enable = The device does not send the SNMP trap for
non-configured ports.
Note: The parameter is applicable to all telephony (analog)
port types.
[SNMPEngineIDString] Defines the SNMP engine ID for SNMPv2/SNMPv3 agents.
This is used for authenticating a user attempting to access the
SNMP agent on the device.
The ID can be a string of up to 36 characters. The default is
00:00:00:00:00:00:00:00:00:00:00:00 (12 Hex octets
characters). The provided key must be set with 12 Hex values
delimited by a colon (":") in the format xx:xx:...:xx. For
example, 00:11:22:33:44:55:66:77:88:99:aa:bb
Notes:
 Before setting this parameter, all SNMPv3 users must be
deleted; otherwise, the parameter setting is ignored.
 If the supplied key does not pass validation of the 12 Hex
values input or it is set with the default value, the engine ID
is generated according to RFC 3411.
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Web: SNMP Trap Destination Parameters

EMS: Network > SNMP Managers Table
Note: Up to five SNMP trap managers can be defined.
SNMP Manager Determines the validity of the parameters (IP address and port
[SNMPManagerIsUsed_x] number) of the corresponding SNMP Manager used to receive
SNMP traps.
 [0] (Check box cleared) = Disabled (default)
 [1] (Check box selected) = Enabled
Web: IP Address Defines the IP address of the remote host used as an SNMP
EMS: Address Manager. The device sends SNMP traps to this IP address.
[SNMPManagerTableIP_x] Enter the IP address in dotted-decimal notation, e.g.,
108.10.1.255.
Web: Trap Port Defines the port number of the remote SNMP Manager. The
EMS: Port device sends SNMP traps to this port.
[SNMPManagerTrapPort_x] The valid SNMP trap port range is 100 to 4000. The default
port is 162.
Web: Trap Enable Enables the sending of traps to the corresponding SNMP
[SNMPManagerTrapSendingEnabl manager.
e_x]  [0] Disable = Sending is disabled.
 [1] Enable = (Default) Sending is enabled.
Web: Trap User Defines the SNMPv3 USM user or SNMPv2 user to associate
[SNMPManagerTrapUser_x] with the trap destination. This determines the trap format,
authentication level, and encryption level. By default, it is
associated with the SNMPv2 user (SNMP trap community
string).
The valid value is a string.
Web: Trap Manager Host Name Defines an FQDN of the remote host used as an SNMP
[SNMPTrapManagerHostName] manager. The resolved IP address replaces the last entry in
the Trap Manager table (defined by the
SNMPManagerTableIP parameter) and the last trap manager
entry of snmpTargetAddrTable in the snmpTargetMIB. For
example: 'mngr.corp.mycompany.com'.
The valid range is a string of up to 99 characters.
SNMP Community String Parameters
Community String Defines up to five read-only SNMP community strings (up to
[SNMPReadOnlyCommunityString 19 characters each). The default string is 'public'.
_x]
Community String Defines up to five read/write SNMP community strings (up to
[SNMPReadWriteCommunityStrin 19 characters each). The default string is 'private'.
g_x]
Trap Community String Defines the Community string used in traps (up to 19
[SNMPTrapCommunityString] characters).
The default string is 'trapuser'.
SNMP Trusted Managers Table
Web: SNMP Trusted Managers Defines up to five IP addresses of remote trusted SNMP
[SNMPTrustedMgr_x] managers from which the SNMP agent accepts and processes
SNMP Get and Set requests.
Notes:

 By default, the SNMP agent accepts SNMP Get and Set
requests from any IP address, as long as the correct
community string is used in the request. Security can be
enhanced by using Trusted Managers, which is an IP
address from which the SNMP agent accepts and
processes SNMP requests.
 If no values are assigned to these parameters any
manager can access the device.
 Trusted managers can work with all community strings.
SNMP V3 Users Table
Web/EMS: SNMP V3 Users This parameter table defines SNMP v3 users. The format of
[SNMPUsers] this parameter is as follows:
[SNMPUsers]
FORMAT SNMPUsers_Index = SNMPUsers_Username,
SNMPUsers_AuthProtocol, SNMPUsers_PrivProtocol,
SNMPUsers_AuthKey, SNMPUsers_PrivKey,
SNMPUsers_Group;
[\SNMPUsers]
For example:
SNMPUsers 1 = v3admin1, 1, 0, myauthkey, -, 1;
The example above configures user 'v3admin1' with security
level authNoPriv(2), authentication protocol MD5,
authentication text password 'myauthkey', and
ReadWriteGroup2.
Note: For a description of this table, see 'Configuring SNMP
V3 Users' on page 91.
44.2.5 TR-069 Parameters

The TR-069 parameters are described in the table below.
Table 44-14: TR-069 Parameters
Web: TR069 Enables device management using TR-069.

CLI: service  [0] Disable (default)
[TR069ServiceEnable]  [1] Enable
required.
Web: Protocol Defines the protocol used for the TR-069 connection.
CLI: protocol  [0] HTTP (default)
[TR069Protocol]  [1] HTTPS
Web: Port Defines the local HTTP/S port used for TR-069.
CLI: port The valid range is 0 to 65535. The default is 82.
[TR069HTTPPort]
required.
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Web: URL Provisioning Mode Defines the method for configuring the URL of the TR-069
CLI: acs-url-provisioning-mode ACS.
[Tr069AcsUrlProvisioningMode]  [0] Manual (default) = URL must be configured
manually on the device. The URL is configured using
the TR069ConnectionRequestUrl parameter.
 [1] Automatic = Device uses DHCP Option 43 to obtain
URL address of ACS.
Web: URL Defines the URL address of the Auto Configuration
CLI: acl-url Servers (ACS) to which the device connects. For example,
[TR069AcsUrl] http://10.4.2.1:10301/acs/.
By default, no URL is defined.
Note: This parameter is applicable only if the 'URL
Provisioning Mode' parameter is set to Manual.
Web: Username Defines the login username that the device uses for
CLI: acs-user-name authenticated access to the ACS.
[TR069AcsUsername] The valid value is a string of up to 256 characters. By
default, no username is defined.
Web: Password Defines the login password that the device uses for
CLI: acs-password authenticated access to the ACS.
[TR069AcsPassword] The valid value is a string of up to 256 characters. By
default, no password is defined.
Web: URL Defines the URL for the ACS connection request. For
CLI: connection-request-url example, http://10.31.4.115:82/tr069/.
[TR069ConnectionRequestUrl
Web: Username Defines the connection request username used by the
CLI: connection-request-user-name ACS to connect to the device.
[TR069ConnectionRequestUsername] The valid value is a string of up to 256 characters. By
default, no username is defined.
Web: Password Defines the connection request password used by the ACS
CLI: connection-request-password to connect to the device.
[TR069ConnectionRequestPassword] The valid value is a string of up to 256 characters. By
default, no password is defined.
Web: Default Inform Interval Defines the inform interval (in seconds) at which the device
CLI: inform-interval periodically communicates with the ACS. Each time the
[TR069PeriodicInformInterval] device communicates with the ACS, the ACS sends a
response indicating whether or not the ACS has an action
to execute on the device.
[TR069RetryinimumWaitInterval] Defines the minimum interval (in seconds) that the device
waits before attempting again to communicate with the
ACS after the previous communication attempt failure.
CLI: debug-mode Defines the debug mode level, which is the type of
[TR069DebugMode] messages sent to the Syslog server.
The valid value is between 0 and 3, where 0 (default)
means no debug messages are sent and 3 is all message
types are sent.

44.2.6 Serial Parameters

The RS-232 serial parameters are described in the table below.
Table 44-15: Serial Parameters
[DisableRS232] Enables the device's RS-232 (serial) port.

 [0] = (Default) Enabled
 [1] = Disabled
The RS-232 serial port can be used to change the networking
parameters and view error/notification messages. For how to establish a
serial communication with the device, refer to the Installation Manual.
EMS: Baud Rate Defines the RS-232 baud rate.
[SerialBaudRate] The valid values include the following: 1200, 2400, 9600 (default),
14400, 19200, 38400, 57600, or 115200.
EMS: Data Defines the RS-232 data bit.
[SerialData]  [7] = 7-bit
 [8] = (Default) 8-bit
EMS: Parity Defines the RS-232 polarity.
[SerialParity]  [0] = (Default) None
 [1] = Odd
 [2] = Even
EMS: Stop Defines the RS-232 stop bit.
[SerialStop]  [1] = (Default) 1-bit (default)
 [2] = 2-bit
EMS: Flow Control Defines the RS-232 flow control.
[SerialFlowControl]  [0] = (Default) None
 [1] = Hardware
44.3 Debugging and Diagnostics Parameters

This subsection describes the device's debugging and diagnostic parameters.

The general debugging and diagnostic parameters are described in the table below.
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Table 44-16: General Debugging and Diagnostic Parameters
EMS: Enable Diagnostics Determines the method for verifying correct functioning of the
[EnableDiagnostics] different hardware components on the device. On completion of the
check and if the test fails, the device sends information on the test
results of each hardware component to the Syslog server.
 [0] = (Default) Rapid and Enhanced self-test mode.
 [1] = Detailed self-test mode (full test of DSPs, PCM, Switch,
LAN, PHY and Flash).
 [2] = A quicker version of the Detailed self-test mode (full test of
DSPs, PCM, Switch, LAN, PHY, but partial test of Flash).
Web: Enable LAN Watchdog Enables the LAN watchdog feature.
[EnableLanWatchDog]  [0] Disable (default)
 [1] Enable
When LAN watchdog is enabled, the device's overall
communication integrity is checked periodically. If no
communication is detected for about three minutes, the device
performs a self test:
 If the self-test succeeds, the problem is a logical link down (i.e.,
Ethernet cable disconnected on the switch side) and the Busy
Out mechanism is activated if enabled (i.e., the parameter
EnableBusyOut is set to 1). Lifeline is activated only if it is
enabled (using the parameter LifeLineType).
 If the self-test fails, the device restarts to overcome internal fatal
communication error.
Notes:
 Enable LAN watchdog is relevant only if the Ethernet connection
is full duplex.
 LAN watchdog is not applicable to MP-118.

[LifeLineType] Defines the condition(s) upon which the Lifeline analog (FXS)
feature is activated. The Lifeline feature can be activated upon a
power outage, physical disconnection of the LAN cable, or network
failure (i.e., loss of IP connectivity). Upon any of these conditions,
the Lifeline feature provides PSTN connectivity and thus call
continuity for the FXS phone users.
If the device is in Lifeline mode and the scenario that caused it to
enter Lifeline (e.g., power outage) no longer exists (e.g., power
returns), the device exists Lifeline and operates as normal.
 [0] = (Default) Lifeline is activated upon power outage.
 [1] = Lifeline is activated upon power outage or when the link is
down (i.e., physically disconnected).
 [2] = Lifeline is activated upon a power outage, when the link is
down (physically disconnected), network failure (logical link
disconnection), or when the Trunk Group is in Busy Out state
(see the EnableBusyOut parameter).
The Lifeline (FXS) phone is connected to the following port:
 MP-11x FXS-only device: FXS Port 1
 MP-118 FXS/FXO device: FXS Ports 1 to 4
For the FXS-only device, FXS Port 1 connects to the POTS
(Lifeline) phone as well as to the PSTN / PBX, using a splitter
cable. For the combined FXS / FXO device, the FXS ports are
provided with a lifeline by their corresponding FXO ports which are
connected to the PSTN / PBX (i.e., FXO Port 5 provides a lifeline
for FXS Port 1, FXO Port 6 provides a lifeline for FXS Port 2, and
so on).
Notes:
 This parameter is applicable only to FXS interfaces.
 For optional value [2], lifeline activation upon an IP network
failure or Busy Out state is not supported by MP-118.
 To enable Lifeline upon a network failure, the LAN watch dog
must be activated (i.e., set the parameter EnableLANWatchDog
to 1).
 For information on Lifeline cabling, refer to the Installation
Manual.
Web: Delay After Reset [sec] Defines the time interval (in seconds) that the device's operation is
[GWAppDelayTime] delayed after a reset.
The valid range is 0 to 45. The default is 7 seconds.
Note: This feature helps overcome connection problems caused by
some LAN routers or IP configuration parameters' modifications by
a DHCP server.
[EnableAutoRAITransmitBER] Enables the device to send a remote alarm indication (RAI) when
the bit error rate (BER) is greater than 0.001.
 [1] Enable
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44.3.2 SIP Test Call Parameters

The SIP Signaling Test Call parameters are described in the table below.
Table 44-17: SIP Test Call Parameters
Web: Test Call DTMF Defines the DTMF tone that is played for answered test calls (incoming
String and outgoing).
[TestCallDtmfString] The DTMF string can be up to 15 strings. The default is "3212333". An
empty string means that no DTMF is played.
Web: Test Call ID Defines the test call prefix number (ID) of the simulated phone on the
[TestCallID] device. Incoming calls received with this called prefix number are
identified as test calls.
This can be any string of up to 15 characters. By default, no number is
defined.
Note: This parameter is only for testing incoming calls destined to this
prefix number.
Test Call Table
Web: Test Call Table Defines the local and remote endpoints to be tested.
[Test_Call] FORMAT Test_Call_Index = Test_Call_EndpointURI,
Test_Call_CalledURI, Test_Call_RouteBy, Test_Call_IPGroupID,
Test_Call_DestAddress, Test_Call_DestTransportType, Test_Call_SRD,
Test_Call_ApplicationType, Test_Call_AutoRegister,
Test_Call_UserName, Test_Call_Password, Test_Call_CallParty,
Test_Call_MaxChannels, Test_Call_CallDuration,
Test_Call_CallsPerSecond, Test_Call_TestMode,
Test_Call_TestDuration, Test_Call_Play, Test_Call_ScheduleInterval;
Note: For a description of this table, see 'Configuring Test Calls' on
page 461.
44.3.3 Syslog, CDR and Debug Parameters

The Syslog, CDR and debug parameters are described in the table below.
Table 44-18: Syslog, CDR and Debug Parameters
Web: Enable Syslog Determines whether the device sends logs and error messages (e.g.,
EMS: Syslog enable CDRs) generated by the device to a Syslog server.
[EnableSyslog]  [0] Disable (default)
 [1] Enable
Notes:
 If you enable Syslog, you must enter an IP address of the Syslog
server (using the SyslogServerIP parameter).
 Syslog messages may increase the network traffic.
 To configure Syslog SIP message logging levels, use the
GwDebugLevel parameter.
 By default, logs are also sent to the RS-232 serial port. For how to
establish serial communication with the device, refer to the
Installation Manual.

Web/EMS: Syslog Server IP Defines the IP address (in dotted-decimal notation) of the computer on
Address which the Syslog server is running. The Syslog server is an application
[SyslogServerIP] designed to collect the logs and error messages generated by the
device.
Web: Syslog Server Port Defines the UDP port of the Syslog server.
EMS: Syslog Server Port The valid range is 0 to 65,535. The default port is 514.
Number
[SyslogServerPort]
Defines the maximum size (in bytes) threshold of logged Syslog
[MaxBundleSyslogLength] messages bundled into a single UDP packet, after which they are sent
to a Syslog server.
The valid value range is 0 to 1220 (where 0 indicates that no bundling
occurs). The default is 1220.
Note: This parameter is applicable only if the GWDebugLevel
parameter is set to 7.
Web: CDR Server IP Defines the destination IP address to where CDR logs are sent.
Address The default is a null string, which causes CDR messages to be sent
EMS: IP Address of CDR with all Syslog messages to the Syslog server.
Server
Notes:
[CDRSyslogServerIP]
 The CDR messages are sent to UDP port 514 (default Syslog port).
 This mechanism is active only when Syslog is enabled (i.e., the
parameter EnableSyslog is set to 1).
Web/EMS: CDR Report Enables media- and signaling-related CDRs to be sent to a Syslog
Level server and determines the call stage at which they are sent.
[CDRReportLevel]  [0] None = (Default) CDRs are not used.
 [1] End Call = CDR is sent to the Syslog server at the end of each
call.
 [2] Start & End Call = CDR report is sent to Syslog at the start and
end of each call.
 [3] Connect & End Call = CDR report is sent to Syslog at
connection and at the end of each call.
 [4] Start & End & Connect Call = CDR report is sent to Syslog at
the start, at connection, and at the end of each call.
Notes:
 The CDR Syslog message complies with RFC 3161 and is
identified by: Facility = 17 (local1) and Severity = 6 (Informational).
 This mechanism is active only when Syslog is enabled (i.e., the
parameter EnableSyslog is set to 1).
Web/EMS: Debug Level Defines the Syslog debug logging level.
[GwDebugLevel]  [0] 0 = (Default) Debug is disabled.
 [1] 1 = Flow debugging is enabled.
 [5] 5 = Flow, device interface, stack interface, session manager,
and device interface expanded debugging are enabled.
 [7] 7 = This option is recommended when the device is running
under "heavy" traffic. In this mode:
 The Syslog debug level automatically changes between level
5, level 1, and level 0, depending on the device's CPU
consumption so that VoIP traffic isn’t affected.
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 Syslog messages are bundled into a single UDP packet, after
which they are sent to a Syslog server (bundling size is
determined by the MaxBundleSyslogLength parameter).
Bundling reduces the number of UDP Syslog packets, thereby
improving CPU utilization.
Note that when this option is used, in order to read Syslog
messages with Wireshark, a special plug-in (i.e., acsyslog.dll) must
be used. Once the plug-in is installed, the Syslog messages are
decoded as "AC SYSLOG" and are dispalyed using the ‘acsyslog’
filter instead of the regular ‘syslog’ filter.
Notes:
 This parameter is typically set to 5 if debug traces are required.
However, in cases of heavy traffic, option 7 is recommended.
 Options 2, 3, 4, and 6 are not recommended.

Web: Syslog Facility Number Defines the Facility level (0 through 7) of the device’s Syslog
EMS: SyslogFacility messages, according to RFC 3164. This allows you to identify Syslog
[SyslogFacility] messages generated by the device. This is useful, for example, if you
collect the device’s and other equipments’ Syslog messages, at one
single server. The device’s Syslog messages can easily be identified
and distinguished from other Syslog messages by its Facility level.
Therefore, in addition to filtering Syslog messages according to IP
address, the messages can be filtered according to Facility level.
 [16] = (Default) local use 0 (local0)
 [17] = local use 1 (local1)
Web: Activity Types to Defines the Activity Log mechanism of the device, which sends log
Report via Activity Log messages to a Syslog server for reporting certain types of Web
Messages operations according to the below user-defined filters.
[ActivityListToLog]  [pvc] Parameters Value Change = Changes made on-the-fly to
parameters. Note that the ini file parameter,
EnableParametersMonitoring can also be used to set this option,
using values [0] (disable) or [1] (enable).
 [afl] Auxiliary Files Loading = Loading of auxiliary files.
 [dr] Device Reset = Reset of device via the 'Maintenance Actions
page.
Note: For this option to take effect, a device reset is required.
 [fb] Flash Memory Burning = Burning of files or parameters to flash
(in 'Maintenance Actions page).
 [swu] Device Software Update = cmp file loading via the Software
Upgrade Wizard.
 [ard] Access to Restricted Domains = Access to restricted
domains, which include the following Web pages:
 (1) ini parameters (AdminPage)
 (2) General Security Settings
 (3) Configuration File
 (4) IP Security Proposal / IP Security Associations Tables
 (5) Software Upgrade Key Status
 (6) Firewall Settings
 (7) Web & Telnet Access List
 (8) WEB User Accounts
 [naa] Non-Authorized Access = Attempt to access the Web
interface with a false or empty user name or password.
 [spc] Sensitive Parameters Value Change = Changes made to
sensitive parameters:
 (1) IP Address
 (2) Subnet Mask
 (3) Default Gateway IP Address
 (4) ActivityListToLog
 [ll] Login and Logout = Every login and logout attempt.
For example: ActivityListToLog = 'pvc', 'afl', 'dr', 'fb', 'swu', 'ard', 'naa',
'spc'
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Note: For the ini file, values must be enclosed in single quotation
marks.
Web: Debug Recording Defines the IP address of the server for capturing debug recording.
Destination IP
[DebugRecordingDestIP]
Web: Debug Recording Defines the UDP port of the server for capturing debug recording. The
Destination Port default is 925.
[DebugRecordingDestPort]
Debug Recording Status Activates or de-activates debug recording.
[DebugRecordingStatus]  [0] Stop (default)
 [1] Start
Logging Filters Table

Web: Logging Filters Table This table parameter defines logging filtering rules for Syslog
[LoggingFilters] messages and debug recordings. The format of this parameter is as
follows:
[ LoggingFilters ]
FORMAT LoggingFilters_Index = LoggingFilters_Type,
LoggingFilters_Value, LoggingFilters_Syslog,
LoggingFilters_CaptureType;
[ \LoggingFilters ]
Note: For a detailed description of this table, see 'Filtering Syslog
Messages and Debug Recordings' on page 449.
44.3.4 Resource Allocation Indication Parameters

The Resource Allocation Indication (RAI) parameters are described in the table below.
Table 44-19: RAI Parameters
[EnableRAI] Enables RAI alarm generation if the device's busy endpoints exceed a
user-defined threshold.
 [0] = (Default) Disable RAI (Resource Available Indication) service.
 [1] = RAI service enabled and an SNMP
'acBoardCallResourcesAlarm' Alarm Trap is sent.
Note: For this parameter to take effect, a device reset is required
[RAIHighThreshold] Defines the high threshold percentage of total calls that are active (busy
endpoints). When the percentage of the device's busy endpoints
exceeds this high threshold, the device sends the SNMP
acBoardCallResourcesAlarm alarm trap with a 'major' alarm status.
The range is 0 to 100. The default is 90.
Note: The percentage of busy endpoints is calculated by dividing the
number of busy endpoints by the total number of “enabled” endpoints.

[RAILowThreshold] Defines the low threshold percentage of total calls that are active (busy
endpoints).
When the percentage of the device's busy endpoints falls below this low
threshold, the device sends an SNMP acBoardCallResourcesAlarm
alarm trap with a 'cleared' alarm status.
The range is 0 to 100%. The default is 90%.
[RAILoopTime] Defines the time interval (in seconds) that the device periodically checks
call resource availability.
44.3.5 BootP Parameters

The BootP parameters are described in the table below. The BootP parameters are special
'hidden' parameters. Once defined and saved in the device's flash memory, they are used
even if they don't appear in the ini file.
Table 44-20: BootP Parameters
[BootPRetries] Note: For this parameter to take effect, a device reset is required.
This parameter is used to:
Defines the number of BootP Defines the number of DHCP
requests that the device sends packets that the device sends. If
during start-up. The device stops after all packets are sent there's
sending BootP requests when still no reply, the device loads from
either BootP reply is received or flash.
number of retries is reached.  [1] = 4 DHCP packets
 [1] = 1 BootP retry, 1 sec.  [2] = 5 DHCP packets
 [2] = 2 BootP retries, 3 sec.  [3] = (Default) 6 DHCP packets
 [3] = (Default) 3 BootP retries,  [4] = 7 DHCP packets
6 sec.  [5] = 8 DHCP packets
 [4] = 10 BootP retries, 30 sec.  [6] = 9 DHCP packets
 [7] = 100 BootP retries, 300
sec.
 [15] = BootP retries indefinitely.
[BootPSelectiveEnable] Enables the Selective BootP mechanism.
 [1] = Enabled
 [0] = Disabled (default)
The Selective BootP mechanism (available from Boot version 1.92)
enables the device's integral BootP client to filter unsolicited
BootP/DHCP replies (accepts only BootP replies that contain the text
'AUDC' in the vendor specific information field). This option is useful in
environments where enterprise BootP/DHCP servers provide undesired
responses to the device's BootP requests.
Notes:
 When working with DHCP (i.e., the parameter DHCPEnable is set to
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1), the selective BootP feature must be disabled.
[BootPDelay] Defines the interval between the device's startup and the first
BootP/DHCP request that is issued by the device.
 [1] = (Default) 1 second
 [2] = 3 second
 [3] = 6 second
 [4] = 30 second
 [5] = 60 second
[ExtBootPReqEnable] Determines whether the device uses the Vendor Specific Information
field in the BootP request to provide device-related initial startup
information.
 [0] = (Default) Disabled.
 [1] = Enables extended information to be sent in BootP requests. The
device uses the Vendor Specific Information field in the BootP
request to provide device-related initial startup information such as
device type, current IP address, software version. For a full list of the
Vendor Specific Information fields, refer to the AcBootP Utility User's
Guide. The AcBootP utility displays this information in the 'Client Info'
column.
Notes:
 This option is not available on DHCP servers.
44.4 Security Parameters

This subsection describes the device's security parameters.

The general security parameters are described in the table below.
Table 44-21: General Security Parameters
Web: Voice Menu Defines the password for accessing the device's FXS Voice menu used for
Password configuring and monitoring the device.
[VoiceMenuPassword] The default is 12345.
Notes:
 To activate the menu, connect a POTS telephone to an FXS port and
dial *** (three stars) followed by the password.
 To disable the Voice menu, do any of the following:
 Set the VoiceMenuPassword parameter to 'disable'.
 Change the Web login password for the Admin user from its default
value (i.e., 'Admin') to any other value, and then reset the device.
 For more information on the Voice menu, see FXS Voice Menu
Guidance on page 32.

[EnableSecureStartup] Enables the Secure Startup mode. In this mode, downloading the ini file to
the device is restricted to a URL provided in initial configuration (see the
parameter IniFileURL) or using DHCP.
 [0] Disable (default).
 [1] Enable = disables TFTP and allows secure protocols such as
HTTPS to fetch the device configuration.
Firewall Table
Web/EMS: Internal This table parameter defines the device's access list (firewall), which
Firewall Parameters defines network traffic filtering rules.
[AccessList] The format of this parameter is as follows:
[AccessList]
FORMAT AccessList_Index = AccessList_Source_IP,
AccessList_Source_Port, AccessList_PrefixLen, AccessList_Source_Port,
AccessList_Start_Port, AccessList_End_Port, AccessList_Protocol,
AccessList_Use_Specific_Interface, AccessList_Interface_ID,
AccessList_Packet_Size, AccessList_Byte_Rate, AccessList_Byte_Burst,
AccessList_Allow_Type;
[\AccessList]
For example:
AccessList 10 = mgmt.customer.com, , , 32, 0, 80, tcp, 1, OAMP, 0, 0, 0,
allow;
AccessList 22 = 10.4.0.0, , , 16, 4000, 9000, any, 0, , 0, 0, 0, block;
In the example above, Rule #10 allows traffic from the host
‘mgmt.customer.com’ destined to TCP ports 0 to 80 on interface OAMP
(OAMP). Rule #22 blocks traffic from the subnet 10.4.xxx.yyy destined to
ports 4000 to 9000.
Note: For a description of this table, see 'Configuring Firewall Settings' on
page 153.
44.4.2 HTTPS Parameters

The Secure Hypertext Transport Protocol (HTTPS) parameters are described in the table
below.
Table 44-22: HTTPS Parameters
Web: Secured Web Connection Determines the protocol used to access the Web interface.
(HTTPS)  [0] HTTP and HTTPS (default).
EMS: HTTPS Only  [1] HTTPs Only = Unencrypted HTTP packets are blocked.
[HTTPSOnly]
required.
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MP-11x & MP-124
EMS: HTTPS Port Defines the local Secured HTTPS port of the device. This
[HTTPSPort] parameter allows secure remote device Web management from
the LAN. To enable secure Web management from the LAN,
configure the desired port.
The valid range is 1 to 65535 (other restrictions may apply within
this range). The default port is 443.
required.
Web/EMS: HTTPS Cipher String Defines the Cipher string for HTTPS (in OpenSSL cipher list
[HTTPSCipherString] format). For the valid range values, refer to URL
http://www.openssl.org/docs/apps/ciphers.html.
The default is ‘RC4:EXP’ (Export encryption algorithms). For
example, use ‘ALL’ for all ciphers suites (e.g., for ARIA
encryption for TLS). The only ciphers available are RC4 and
DES, and the cipher bit strength is limited to 56 bits.
required.
Web: HTTP Authentication Mode Determines the authentication mode used for the Web interface.
EMS: Web Authentication Mode  [0] Basic Mode = Basic authentication (clear text) is used.
[WebAuthMode]  [1] Web Based Authentication = (Default) Digest
authentication (MD5) is used.
Note: If you enable RADIUS login (i.e., the WebRADIUSLogin
parameter is set to 1), you must set the WebAuthMode
parameter to Basic Mode [0].
Web: Requires Client Certificates Determines whether client certificates are required for HTTPS
for HTTPS connection connection.
[HTTPSRequireClientCertificate]  [0] Disable = (Default) Client certificates are not required.
 [1] Enable = Client certificates are required. The client
certificate must be preloaded to the device and its matching
private key must be installed on the managing PC. Time and
date must be correctly set on the device for the client
certificate to be verified.
Notes:
 For a description on implementing client certificates, see
'Client Certificates' on page 114.
[HTTPSRootFileName] Defines the name of the HTTPS trusted root certificate file to be
loaded using TFTP. The file must be in base64-encoded PEM
(Privacy Enhanced Mail) format.
The valid range is a 47-character string.
Note: This parameter is applicable only when the device is
loaded using BootP/TFTP.
[HTTPSPkeyFileName] Defines the name of a private key file (in unencrypted PEM
format) to be loaded from the TFTP server.
[HTTPSCertFileName] Defines the name of the HTTPS server certificate file to be
loaded using TFTP. The file must be in base64-encoded PEM
format.
Note: This parameter is only applicable when the device is
loaded using BootP/TFTP.

44.4.3 SRTP Parameters

The Secure Real-Time Transport Protocol (SRTP) parameters are described in the table
below.
Table 44-23: SRTP Parameters
Web: Media Security Enables Secure Real-Time Transport Protocol (SRTP).

EMS: Enable Media Security  [0] Disable = (Default) SRTP is disabled.
[EnableMediaSecurity]  [1] Enable = SRTP is enabled.
Notes:
 SRTP reduces the number of available channels.
 MP-124: 18 available channels
 MP-112: No reduction
Web/EMS: Media Security Determines the device's mode of operation when SRTP is used
Behavior (i.e., when the parameter EnableMediaSecurity is set to 1).
[MediaSecurityBehaviour]  [0] Preferable = (Default) The device initiates encrypted calls.
However, if negotiation of the cipher suite fails, an
unencrypted call is established. Incoming calls that don't
include encryption information are accepted.
 [1] Mandatory = The device initiates encrypted calls, but if
negotiation of the cipher suite fails, the call is terminated.
Incoming calls that don't include encryption information are
rejected.
 [2] Disable = The IP Profile for which this parameter is set
does not support encrypted calls (i.e., SRTP).
 [3] Preferable - Single Media = The device sends SDP with a
single media ('m=') line only (e.g., m=audio 6000 RTP/AVP 4 0
70 96) with RTP/AVP and crypto keys. The remote UA can
respond with SRTP or RTP parameters:
 If the remote SIP UA does not support SRTP, it uses RTP
and ignores the crypto lines.
 In the opposite direction, if the device receives an SDP
offer with a single media (as shown above), it responds
with SRTP (RTP/SAVP) if the EnableMediaSecurity
parameter is set to 1. If SRTP is not supported (i.e.,
EnableMediaSecurity is set to 0), it responds with RTP.
Notes:
 Before configuring this parameter, set the
EnableMediaSecurity parameter to 1.
 If this parameter is set to Preferable [3] and two 'm=' lines are
received in the SDP offer, the device prefers the SAVP
(secure audio video profile) regardless of the order in the SDP.
 Option [2] Disable is applicable only to IP Profiles.
 This parameter can also be configured per IP Profile, using the
IPProfile parameter (see 'Configuring IP Profiles' on page
225).
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Web: Master Key Identifier (MKI) Defines the size (in bytes) of the Master Key Identifier (MKI) in
Size SRTP Tx packets.
EMS: Packet MKI Size The range is 0 to 4. The default is 0 (i.e., new keys are generated
[SRTPTxPacketMKISize] without MKI).
Notes:
 The device only initiates the MKI size.
 You can also configure MKI size in an IP Profile.
Web: Symmetric MKI Enables symmetric MKI negotiation.
Negotiation  [0] Disable = (Default) The device includes the MKI in its 200
EMS: Enable Symmetric MKI OK response according to the SRTPTxPacketMKISize
[EnableSymmetricMKI] parameter (if set to 0, then it is not included; if set to any other
value, it is included with this value).
 [1] Enable = The answer crypto line contains (or excludes) an
MKI value according to the selected crypto line in the offer. For
example, assume that the device receives an INVITE
containing the following two crypto lines in SDP:
inline:TAaxNnQt8/qLQMnDuG4vxYfWl6K7eBK/ufk04pR
4|2^31|1:1
inline:bnuYZnMxSfUiGitviWJZmzr7OF3AiRO0l5Vnh0k
H|2^31
The first crypto line includes the MKI parameter "1:1". In the
200 OK response, the device selects one of the crypto lines
(i.e., '2' or '3'). Typically, it selects the first line that supports
the crypto suite. If the device selects crypto line '2', it includes
the MKI parameter in its answer SDP, for example:
inline:R1VyA1xV/qwBjkEklu4kSJyl3wCtYeZLq1/QFux
w|2^31|1:1
If the device selects a crypto line that does not contain the MKI
parameter, then the MKI parameter is not included in the
crypto line in the SDP answer (even if the
SRTPTxPacketMKISize parameter is set to any value other
than 0).
Notes:
 To enable symmetric MKI, the SRTPTxPacketMKISize
parameter must be set to any value other than 0.
 You can also enable MKI negotiation per IP Profile.
Web/EMS: SRTP offered Suites Defines the offered crypto suites (cipher encryption algorithms)
[SRTPofferedSuites] for SRTP.
 [0] = (Default) All available crypto suites.
 [1] CIPHER SUITES AES CM 128 HMAC SHA1 80 = device
uses AES-CM encryption with a 128-bit key and HMAC-SHA1
message authentication with a 80-bit tag.
 [2] CIPHER SUITES AES CM 128 HMAC SHA1 32 = device
uses AES-CM encryption with a 128-bit key and HMAC-SHA1
message authentication with a 32-bit tag.
Note: This parameter also affects the selection of the crypto in
the device's answer. For example, if the device receives an offer
with two crypto lines containing HMAC_SHA1_80 and
HMAC_SHA_32, it uses the HMAC_SHA_32 key in its SIP 200

OK response if the parameter is set to 2.
Web: Disable Authentication On Enables authentication on transmitted RTP packets in a secured
Transmitted RTP Packets RTP session.
EMS: RTP AuthenticationDisable  [0] Enable (default)
Tx
 [1] Disable
[RTPAuthenticationDisableTx]
Web: Disable Encryption On Enables encryption on transmitted RTP packets in a secured RTP
Transmitted RTP Packets session.
EMS: RTP EncryptionDisable Tx  [0] Enable (default)
[RTPEncryptionDisableTx]  [1] Disable
Web: Disable Encryption On Enables encryption on transmitted RTCP packets in a secured
Transmitted RTCP Packets RTP session.
EMS: RTCP EncryptionDisable  [0] Enable (default)
Tx  [1] Disable
[RTCPEncryptionDisableTx]
[ResetSRTPStateUponRek Enables synchronization of the SRTP state between the device and
ey] a server when a new SRTP key is generated upon a SIP session
expire. This feature ensures that the roll-over counter (ROC), one of
the parameters used in the SRTP encryption/decryption process of
the SRTP packets, is synchronized on both sides for transmit and
receive packets.
 [0] = (Default) Disabled. ROC is not reset on the device side.
 [1] = Enabled. If the session expires causing a session refresh
through a re-INVITE, the device or server generates a new key
and the device resets the ROC index (and other SRTP fields) as
done by the server, resulting in a synchronized SRTP.
Notes:
 This feature can also be configured for an IP Profile.
 If this feature is disabled and the server resets the ROC upon a
re-key generation, one-way voice may occur.
44.4.4 TLS Parameters

The Transport Layer Security (TLS) parameters are described in the table below.
Table 44-24: TLS Parameters
Web/EMS: TLS Version Defines the supported SSL/TLS protocol version. Clients
[TLSVersion] attempting to communicate with the device using a different
TLS version are rejected.
 [0] Any - Including SSLv3 = (Default) SSL 3.0 and all TLS
versions are supported.
 [1] TLSv1.0 = Only TLS 1.0.
 [2] TLSv1.1 = Only TLS 1.1.
 [3] TLSv1.0 and TLSv1.1 = Only TLS 1.0 and TLS 1.1.
 [4] TLSv1.2 = Only TLS 1.2.
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[7] TLSv1.0 TLSv1.1 and TLSv1.2 = Only TLS 1.0, TLS 1.1 and
TLS 1.2 (excludes SSL 3.0).
required.
Web: TLS Client Re-Handshake Defines the time interval (in minutes) between TLS Re-
Interval Handshakes initiated by the device.
EMS: TLS Re Handshake Interval The interval range is 0 to 1,500 minutes. The default is 0 (i.e.,
[TLSReHandshakeInterval] no TLS Re-Handshake).
Web: TLS Mutual Authentication Determines the device's behavior when acting as a server for
EMS: SIPS Require Client TLS connections.
Certificate  [0] Disable = (Default) The device does not request the
[SIPSRequireClientCertificate] client certificate.
 [1] Enable = The device requires receipt and verification of
the client certificate to establish the TLS connection.
Notes:
 The SIPS certificate files can be changed using the
parameters HTTPSCertFileName and
HTTPSRootFileName.
Web/EMS: Peer Host Name Determines whether the device verifies the Subject Name of a
Verification Mode remote certificate when establishing TLS connections.
[PeerHostNameVerificationMode]  [0] Disable (default).
 [1] Server Only = Verify Subject Name only when acting as
a client for the TLS connection.
 [2] Server & Client = Verify Subject Name when acting as a
server or client for the TLS connection.
When a remote certificate is received and this parameter is not
disabled, the value of SubjectAltName is compared with the list
of available Proxies. If a match is found for any of the
configured Proxies, the TLS connection is established.
The comparison is performed if the SubjectAltName is either a
DNS name (DNSName) or an IP address. If no match is found
and the SubjectAltName is marked as ‘critical’, the TLS
connection is not established. If DNSName is used, the
certificate can also use wildcards (‘*’) to replace parts of the
domain name.
If the SubjectAltName is not marked as ‘critical’ and there is no
match, the CN value of the SubjectName field is compared with
the parameter TLSRemoteSubjectName. If a match is found,
the connection is established. Otherwise, the connection is
terminated.
Note: If you set this parameter to [2] (Server & Client), for this
functionality to operate, you also need to set the
SIPSRequireClientCertificate parameter to [1] (Enable).
Web: TLS Client Verify Server Determines whether the device, when acting as a client for TLS
Certificate connections, verifies the Server certificate. The certificate is
EMS: Verify Server Certificate verified with the Root CA information.
[VerifyServerCertificate]  [0] Disable (default)
 [1] Enable
Note: If Subject Name verification is necessary, the parameter
PeerHostNameVerificationMode must be used as well.

Web: Strict Certificate Extension Enables the validation of the extensions (keyUsage and
Validation extentedKeyUsage) of peer certificates. This validation ensures
[RequireStrictCert] that the signing CA is authorized to sign certificates and that the
end-entity certificate is authorized to negotiate a secure TLS
connection.
 [1] Enable
Web/EMS: TLS Remote Subject Defines the Subject Name that is compared with the name
Name defined in the remote side certificate when establishing TLS
[TLSRemoteSubjectName] connections.
If the SubjectAltName of the received certificate is not equal to
any of the defined Proxies Host names/IP addresses and is not
marked as 'critical', the Common Name (CN) of the Subject field
is compared with this value. If not equal, the TLS connection is
not established. If the CN uses a domain name, the certificate
can also use wildcards (‘*’) to replace parts of the domain
name.
The valid range is a string of up to 49 characters.
PeerHostNameVerificationMode is set to 1 or 2.
Web: Client Cipher String Defines the cipher-suite string for TLS clients.
[TLSClientCipherString] The valid value is up to 255 strings. The default is "ALL:!ADH".
For example: TLSClientCipherString = 'EXP'
This parameter complements the HTTPSCipherString
parameter (which affects TLS servers). For possible values and
additional details, visit the OpenSSL website at
https://www.openssl.org/docs/man1.0.2/apps/ciphers.html.
[TLSPkeySize] Defines the key size (in bits) for RSA public-key encryption for
newly self-signed generated keys for SSH.
 [512]
 [768]
 [1024] (default)
 [2048]
Web: TLS FIPS 140 Mode Enables FIPS 140-2 conformance mode for TLS.
[TLS_Fips140_Mode]  [0] Disable (default)
 [1] Enable
44.4.5 SSH Parameters

Secure Shell (SSH) parameters are described in the table below.
Table 44-25: SSH Parameters
Web/EMS: Enable SSH Server Enables the device's embedded SSH server.
[SSHServerEnable]  [0] Disable (default)
 [1] Enable
Web/EMS: Server Port Defines the port number for the embedded SSH server.
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[SSHServerPort] Range is any valid port number. The default port is 22.
Web/EMS: SSH Admin Key Defines the RSA public key for strong authentication for logging in
[SSHAdminKey] to the SSH interface (if enabled).
The value should be a base64-encoded string. The value can be
a maximum length of 511 characters.
Web: Require Public Key Enables RSA public keys for SSH.
EMS: EMS: SSH Require Public  [0] = (Default) RSA public keys are optional if a value is
Key configured for the parameter SSHAdminKey.
[SSHRequirePublicKey]  [1] = RSA public keys are mandatory.
Note: To define the key size, use the TLSPkeySize parameter.
Web: Max Payload Size Defines the maximum uncompressed payload size (in bytes) for
EMS: SSH Max Payload Size SSH packets.
[SSHMaxPayloadSize] The valid value is 550 to 32768. The default is 32768.
Web: Max Binary Packet Size Defines the maximum packet size (in bytes) for SSH packets.
EMS: SSH Max Binary Packet The valid value is 582 to 35000. The default is 35000.
Size
[SSHMaxBinaryPacketSize]
EMS: Telnet SSH Max Sessions Defines the maximum number of simultaneous SSH sessions.
[SSHMaxSessions] The valid range is 1 to 2. The default is 2 sessions.
Web: Enable Last Login Message Enables message display in SSH sessions of the time and date of
[SSHEnableLastLoginMessage] the last SSH login. The SSH login message displays the number
of unsuccessful login attempts since the last successful login.
 [0] Disable
 [1] Enable (default)
Note: The last SSH login information is cleared when the device
is reset.
Web: Max Login Attempts Defines the maximum SSH login attempts allowed for entering an
[SSHMaxLoginAttempts] incorrect password by an administrator before the SSH session is
rejected.
Note: The new setting takes effect only for new subsequent SSH
connections
44.4.6 IPSec Parameters

The Internet Protocol security (IPSec) parameters are described in the table below.
Table 44-26: IPSec Parameters
IPSec Parameters
Web: Enable IP Security Enables IPSec on the device.
EMS: IPSec Enable  [0] Disable (default)
[EnableIPSec]  [1] Enable
Web: IKE Certificate Ext Enables the validation of the extensions (keyUsage and

Validate extentedKeyUsage) of peer certificates. This validation ensures that the
[IKEcertificateExtValidate] signing CA is authorized to sign certificates and that the end-entity
certificate is authorized to negotiate a secure IPSec connection.
 [[0] Disable (default)
 [1] Enable
IPSec Associations Table
Web: IP Security This table parameter defines the IPSec SA table. This table allows you
Associations Table to configure the Internet Key Exchange (IKE) and IP Security (IPSec)
EMS: IPSec SA Table protocols. You can define up to 20 IPSec peers.
[IPSecSATable] The format of this parameter is as follows:
[ IPsecSATable ]
FORMAT IPsecSATable_Index =
IPsecSATable_RemoteEndpointAddressOrName,
IPsecSATable_AuthenticationMethod, IPsecSATable_SharedKey,
IPsecSATable_SourcePort, IPsecSATable_DestPort,
IPsecSATable_Protocol, IPsecSATable_Phase1SaLifetimeInSec,
IPsecSATable_Phase2SaLifetimeInSec,
IPsecSATable_Phase2SaLifetimeInKB, IPsecSATable_DPDmode,
IPsecSATable_IPsecMode, IPsecSATable_RemoteTunnelAddress,
IPsecSATable_RemoteSubnetIPAddress,
IPsecSATable_RemoteSubnetPrefixLength,
IPsecSATable_InterfaceName;
[ \IPsecSATable ]
For example:
IPsecSATable 1 = 0, 10.3.2.73, 0, 123456789, 0, 0, 0, 0, 28800, 3600, ;
In the above example, a single IPSec/IKE peer (10.3.2.73) is
configured. Pre-shared key authentication is selected, with the pre-
shared key set to 123456789. In addition, a lifetime of 28800 seconds
is selected for IKE and a lifetime of 3600 seconds is selected for IPSec.
Note: For a detailed description of this table, see 'Configuring IP
Security Associations Table' on page 161.
IPSec Proposal Table
Web: IP Security Proposal This table parameter defines up to four IKE proposal settings, where
Table each proposal defines an encryption algorithm, an authentication
EMS: IPSec Proposal Table algorithm, and a Diffie-Hellman group identifier.
[IPSecProposalTable] [ IPsecProposalTable ]
FORMAT IPsecProposalTable_Index =
IPsecProposalTable_EncryptionAlgorithm,
IPsecProposalTable_AuthenticationAlgorithm,
IPsecProposalTable_DHGroup;
[ \IPsecProposalTable ]
For example:
IPsecProposalTable 0 = 3, 2, 1;
IPsecProposalTable 1 = 2, 2, 1;
In the example above, two proposals are defined:
 Proposal 0: AES, SHA1, DH group 2
 Proposal 1: 3DES, SHA1, DH group 2
Note: For a detailed description of this table, see 'Configuring IP
Security Proposal Table' on page 159.
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44.4.7 802.1X Parameters

The 802.1X parameters are described in the table below.
Table 44-27: 802.1X Parameters
Web: 802.1x Mode Enables support for IEEE 802.1x physical port security. The device
EMS: Mode can function as an IEEE 802.1X supplicant. IEEE 802.1X is a standard
[802.1xMode] for port-level security on secure Ethernet switches; when a unit is
connected to a secure port, no traffic is allowed until the identity of the
unit is authenticated.
 [0] Disabled (default)
 [1] EAP-MD5 = Authentication is performed using a user name and
password configured by the parameters 802.1xUsername and
802.1xPassword.
 [2] Protected EAP = Authentication is performed using a user name
and password configured by the parameters 802.1xUsername and
802.1xPassword. In addition, the protocol used is MSCHAPv2 over
an encrypted TLS tunnel.
 [3] EAP-TLS = The device's certificate is used to establish a
mutually-authenticated TLS session with the Access Server. This
requires prior configuration of the server certificate and root CA
(see Configuring the Certificates on page 111). The parameter
802.1xUsername is used to identify the device, however
802.1xPassword is ignored.
Note: The configured mode must match the configuration of the
Access server (e.g., RADIUS server).
Web: 802.1x Username Defines the username for IEEE 802.1x support.
EMS: User Name The valid value is a string of up to 32 characters. The default is an
[802.1xUsername] empty string.
Web: 802.1x Password Defines the password for IEEE 802.1x support.
EMS: Password The valid value is a string of up to 32 characters. The default is an
[802.1xPassword] empty string.
Web: 802.1x Verify Peer Determines whether the device verifies the Peer Certificate for IEEE
Certificate 802.1x support.
EMS: Verify Peer Certificate  [0] Disable (default)
[802.1xVerifyPeerCertificate]  [1] Enable

44.4.8 OCSP Parameters

The Online Certificate Status Protocol (OCSP) parameters are described in the table
below.
Table 44-28: OCSP Parameters
Web: Enable OCSP Server Enables or disables certificate checking using OCSP.
EMS: OCSP Enable  [0] Disable (default)
[OCSPEnable]  [1] Enable
Web: Primary Server IP Defines the IP address of the OCSP server.
EMS: OCSP Server IP The default IP address is 0.0.0.0.
[OCSPServerIP]
Web: Secondary Server IP Defines the IP address (in dotted-decimal notation) of the secondary
[OCSPSecondaryServerIP] OCSP server (optional).
Web: Server Port Defines the OCSP server's TCP port number.
EMS: OCSP Server Port The default port number is 2560.
[OCSPServerPort]
Web: Default Response Determines the default OCSP behavior when the server cannot be
When Server Unreachable contacted.
EMS: OCSP Default  [0] Reject = (Default) Rejects peer certificate.
Response  [1] Allow = Allows peer certificate.
[OCSPDefaultResponse]
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44.5 RADIUS Parameters

The RADIUS parameters are described in the table below. For supported RADIUS
attributes, see 'RADIUS Accounting CDR Attributes' on page 435.
Table 44-29: RADIUS Parameters
RADIUS Accounting Parameters

Web: Enable RADIUS Access Enables the RADIUS application.
Control  [0] Disable (Default)
[EnableRADIUS]  [1] Enable
Web: Accounting Server IP Defines the IP address of the RADIUS accounting server.
Address
[RADIUSAccServerIP]
Web: Accounting Port Defines the port of the RADIUS accounting server.
[RADIUSAccPort] The default is 1646.
Web/EMS: RADIUS Accounting Determines when the RADIUS accounting messages are sent to
Type the RADIUS accounting server.
[RADIUSAccountingType]  [0] At Call Release = (Default) Sent at call release only.
 [1] At Connect & Release = Sent at call connect and release.
 [2] At Setup & Release = Sent at call setup and release.
Web: AAA Indications Determines the Authentication, Authorization and Accounting
EMS: Indications (AAA) indications.
[AAAIndications]  [0] None = (Default) No indications.
 [3] Accounting Only = Only accounting indications are used.
General RADIUS Parameters
Web: Use RADIUS for Enables RADIUS queries for Web and Telnet login authentication.
Web/Telnet Login When enabled, logging into the device's Web and Telnet
EMS: Web Use Radius Login embedded servers is done through a RADIUS server. The device
[WebRADIUSLogin] communicates with a user-defined RADIUS server and verifies the
given username and password against a remote database, in a
secure manner.
 [1] Enable
Notes:
 For RADIUS login authentication to function, you also need to
set the following parameters:
 EnableRADIUS = 1 (Enable)
 WebAuthMode = 0 (Basic Mode)
 RADIUS authentication requires HTTP basic authentication,
where the username and password are transmitted in clear text
over the network. Therefore, it's recommended to set the
HTTPSOnly parameter to 1 in order to force the use of HTTPS,
since the transport is encrypted.
 If using RADIUS authentication to log into the CLI, only the
primary Web User Account, which has Security Administration
access level, can access the device's CLI (see 'Configuring
Web User Accounts' on page 66).

Web: RADIUS Authentication Defines the IP address of the RADIUS authentication server.
Server IP Address Note: For this parameter to take effect, a device reset is required.
EMS: RADIUS Auth Server IP
[RADIUSAuthServerIP]
Web: RADIUS Authentication Defines the port of the RADIUS Authentication Server.
Server Port Note: For this parameter to take effect, a device reset is required.
EMS: RADIUS Auth Server Port
[RADIUSAuthPort]
Web: RADIUS Shared Secret Defines the 'Secret' used to authenticate the device to the
EMS: RADIUS Auth Server RADIUS server. This should be a cryptically strong password.
Secret
[SharedSecret]
RADIUS Authentication Parameters
Web: Default Access Level Defines the default access level for the device when the RADIUS
[DefaultAccessLevel] (authentication) response doesn't include an access level
attribute.
The valid range is 0 to 255. The default is 200 (i.e., Security
Administrator).
Web: Behavior upon = Defines the mode of operation regarding user login
Authentication Server Timeout authentication if connection with the LDAP server fails (due to a
[MgmtBehaviorOnTimeout] timeout, temporary network malfunction or AD server problem).
 [0] Deny Access
 [1] Verify Access Locally = (Default) Device verifies the user's
credentials (username/password) locally in its user database
and grants access if correct; otherwise, it denies access.
Web: Local RADIUS Password Determines the device's mode of operation regarding the timer
Cache Mode (configured by the parameter RadiusLocalCacheTimeout) that
[RadiusLocalCacheMode] determines the validity of the user name and password (verified by
the RADIUS server).
 [0] Absolute Expiry Timer = When you access a Web page, the
timeout doesn't reset, instead it continues decreasing.
 [1] Reset Timer Upon Access = (Default) Upon each access to
a Web page, the timeout always resets (reverts to the initial
value configured by RadiusLocalCacheTimeout).
Web: Local RADIUS Password Defines the time (in seconds) the locally stored user name and
Cache Timeout password (verified by the RADIUS server) are valid. When this
[RadiusLocalCacheTimeout] time expires, the user name and password become invalid and a
must be re-verified with the RADIUS server.
The valid range is 1 to 0xFFFFFF. The default is 300 (5 minutes).
 [-1] = Never expires.
 [0] = Each request requires RADIUS authentication.
Web: RADIUS VSA Vendor ID Defines the vendor ID that the device accepts when parsing a
[RadiusVSAVendorID] RADIUS response packet.
The valid range is 0 to 0xFFFFFFFF. The default is 5003.
Web: RADIUS VSA Access Defines the code that indicates the access level attribute in the
Level Attribute Vendor Specific Attributes (VSA) section of the received RADIUS
[RadiusVSAAccessAttribute] packet.
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[MaxRADIUSSessions] Defines the number of concurrent calls that can communicate with
the RADIUS server (optional).
EMS: RADIUS Auth Number of Defines the number of retransmission retries.
Retries The valid range is 1 to 10. The default is 3.
[RADIUSRetransmission]
[RadiusTO] Defines the time interval (measured in seconds) that the device
waits for a response before a RADIUS retransmission is issued.
44.6 SIP Media Realm Parameters

The Media Realm parameters are described in the table below.
Table 44-30: Media Realm Parameters
Media Realm Table

Web: Media Realm Table This table parameter defines the Media Realm table. The Media Realm
EMS: VoIP > Media > table allows you to divide a Media-type interface (defined in the Multiple
Media Realm Interface table) into several realms, where each realm is specified by a
[CpMediaRealm] UDP port range.
[CpMediaRealm]
FORMAT CpMediaRealm_Index = CpMediaRealm_MediaRealmName,
CpMediaRealm_IPv4IF, CpMediaRealm_IPv6IF,
CpMediaRealm_PortRangeStart, CpMediaRealm_MediaSessionLeg,
CpMediaRealm_PortRangeEnd, CpMediaRealm_TransRateRatio,
CpMediaRealm_IsDefault;
[\CpMediaRealm]
For example,
CpMediaRealm 1 = Mrealm1, Voice, , 6600, 20, 6790, , 1;
CpMediaRealm 2 = Mrealm2, Voice, , 6800, 10, 6890; , 0;
Notes:
 For a detailed description of this table, see 'Configuring Media
Realms' on page 188.
Quality of Experience Parameters
Web: Server IP Defines the IP address of AudioCodes Session Experience Manager
CLI: server-ip (SEM) server to where the quality experience reports are sent.
[QOEServerIP] Note: For this parameter to take effect, a device reset is required.
Web: Port Defines the port of the SEM server.
[QOEPort] The valid value range is 0 to 65534. The default is 5000.
Web: Interface Name Defines the IP network interface on which the quality experience reports
[QOEInterfaceName] are sent.
The default is the OAMP interface.

Media Realm > Quality of Experience Table

Web: Media Realm > This table configures Quality of Experience parameters per Media
Quality Of Experience Realm.
EMS: Media > Media [ QOERules ]
Realm > Voice Quality ORMAT QOERules_Index = QOERules_MediaRealmIndex,
Rules QOERules_RuleIndex, QOERules_MonitoredParam,
[QOERules] QOERules_Direction, QOERules_Profile,
QOERules_GreenYellowThreshold, QOERules_GreenYellowHysteresis,
QOERules_YellowRedThreshold, QOERules_YellowRedHysteresis,
QOERules_GreenYellowOperation,
QOERules_GreenYellowOperationDetails,
QOERules_YellowRedOperation,
QOERules_YellowRedOperationDetails;
[ \QOERules ]
Note: For a detailed description of this table, see Configuring Quality of
Experience Parameters per Media Realm on page 190.
44.7 Control Network Parameters

44.7.1 IP Group, Proxy, Registration and Authentication Parameters
The proxy server, registration and authentication SIP parameters are described in the table
below.
Table 44-31: Proxy, Registration and Authentication SIP Parameters
IP Group Table
Web: IP Group Table This table configures IP Groups.
EMS: Endpoints > IP Group The ini file format of this parameter is as follows:
[IPGroup]
[IPGroup]
FORMAT IPGroup_Index = IPGroup_Type,
IPGroup_Description, IPGroup_ProxySetId,
IPGroup_SIPGroupName, IPGroup_ContactUser,
IPGroup_EnableSurvivability, IPGroup_ServingIPGroup,
IPGroup_SipReRoutingMode,
IPGroup_AlwaysUseRouteTable, IPGroup_RoutingMode,
IPGroup_SRD, IPGroup_MediaRealm,
IPGroup_ClassifyByProxySet, IPGroup_ProfileId,
IPGroup_MaxNumOfRegUsers, IPGroup_InboundManSet,
IPGroup_OutboundManSet, IPGroup_RegistrationMode,
IPGroup_AuthenticationMode, IPGroup_MethodList,
IPGroup_EnableSBCClientForking, IPGroup_SourceUriInput,
IPGroup_DestUriInput, IPGroup_ContactName;
[/IPGroup]
Notes:
 For a description of this table, see 'Configuring IP Groups'
on page 205.
Authentication per Port Table
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Web: Authentication Table This table parameter defines a user name and password for
EMS: SIP Endpoints > authenticating each device port. The format of this parameter is
Authentication as follows:
[Authentication] [Authentication]
FORMAT Authentication_Index = Authentication_UserId,
Authentication_UserPassword;
[\Authentication]
Where,
 Index = port number, where 0 denotes the Port 1
For example:
Authentication 1 = lee,1552; (user name "lee" with password
1552 for authenticating Port 2)
Note: For a description o this table, see Configuring
Authentication on page 304.
Account Table
Web: Account Table This table parameter configures the Account table for
EMS: SIP Endpoints > Account registering and/or authenticating (digest) Hunt Groups(e.g., an
[Account] IP-PBX) to another IP Group (e.g., an Internet Telephony
Service Provider - ITSP). The format of this parameter is as
follows:
[Account]
FORMAT Account_Index = Account_ServedTrunkGroup,
Account_ServedIPGroup, Account_ServingIPGroup,
Account_Username, Account_Password, Account_HostName,
Account_Register, Account_ContactUser,
Account_ApplicationType;
[\Account]
For example:
Account 1 = 1, -1, 1, user, 1234, acl, 1, ITSP1;
Note: For a detailed description of this table, see 'Configuring
Account Table' on page 213.
Proxy Registration Parameters
Web: Use Default Proxy Enables the use of a SIP proxy server.
EMS: Proxy Used  [0] No = (Default) Proxy isn't used and instead, the internal
[IsProxyUsed] routing table is used.
 [1] Yes = Proxy server is used. Define the IP address of the
proxy server in the Proxy Sets table (see 'Configuring Proxy
Sets Table' on page 208).
Note: If you are not using a proxy server, you must define
outbound IP call routing rules in the Tel to IP Routing
(described in Configuring Tel to IP Routing on page 256).
Web/EMS: Proxy Name Defines the Home Proxy domain name. If specified, this name
[ProxyName] is used as the Request-URI in REGISTER, INVITE and other
SIP messages, and as the host part of the To header in INVITE
messages. If not specified, the Proxy IP address is used
instead.
The valid value is a string of up to 49 characters.
Note: This parameter functions together with the
UseProxyIPasHost parameter.
Web: Use Proxy IP as Host Enables the use of the proxy server's IP address (in dotted-

[UseProxyIPasHost] decimal notation) as the host name in SIP From and To
headers in REGISTER requests.
 [1] Enable
If this parameter is disabled and the device registers to an IP
Group (i.e., proxy server), it uses the string configured by the
ProxyName parameter as the host name in the REGISTER's
Request-URI and uses the string configured by the IP Group
table parameter, SIPGroupName as the host name in the To
and From headers. If the IP Group is configured with a Proxy
Set that has multiple IP addresses, all the REGISTER
messages sent to these proxies are sent with the same host
name.
Note: If this parameter is disabled and the ProxyName
parameter is not configured, the proxy's IP address is used as
the host name in the REGISTER Request-URI.
Web: Redundancy Mode Determines whether the device switches back to the primary
EMS: Proxy Redundancy Mode Proxy after using a redundant Proxy.
[ProxyRedundancyMode]  [0] Parking = (Default) The device continues working with a
redundant (now active) Proxy until the next failure, after
which it works with the next redundant Proxy.
 [1] Homing = The device always tries to work with the
primary Proxy server (i.e., switches back to the primary
Proxy whenever it's available).
Note: To use this Proxy Redundancy mechanism, you need to
enable the keep-alive with Proxy option, by setting the
parameter EnableProxyKeepAlive to 1 or 2.
Web: Proxy IP List Refresh Time Defines the time interval (in seconds) between each Proxy IP
EMS: IP List Refresh Time list refresh.
[ProxyIPListRefreshTime] The range is 5 to 2,000,000. The default interval is 60.
Web: Enable Fallback to Routing Determines whether the device falls back to the Tel to IP
Table Routing for call routing when Proxy servers are unavailable.
EMS: Fallback Used  [0] Disable = (Default) Fallback is not used.
[IsFallbackUsed]  [1] Enable = The Tel to IP Routing is used when Proxy
servers are unavailable.
When the device falls back to the Tel to IP Routing, it continues
scanning for a Proxy. When the device locates an active Proxy,
it switches from internal routing back to Proxy routing.
Note: To enable the redundant Proxies mechanism, set the
parameter EnableProxyKeepAlive to 1 or 2.
Web/EMS: Prefer Routing Table Determines whether the device's internal routing table takes
[PreferRouteTable] precedence over a Proxy for routing calls.
 [0] No = (Default) Only a Proxy server is used to route calls.
 [1] Yes = The device checks the routing rules in the Tel to
IP Routing for a match with the Tel-to-IP call. Only if a
match is not found is a Proxy used.
Web/EMS: Always Use Proxy Determines whether the device sends SIP messages and
[AlwaysSendToProxy] responses through a Proxy server.
 [0] Disable = (Default) Use standard SIP routing rules.
 [1] Enable = All SIP messages and responses are sent to
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the Proxy server.
Note: This parameter is applicable only if a Proxy server is
used (i.e., the parameter IsProxyUsed is set to 1).
Web: SIP ReRouting Mode Determines the routing mode after a call redirection (i.e., a 3xx
EMS: SIP Re-Routing Mode SIP response is received) or transfer (i.e., a SIP REFER
[SIPReroutingMode] request is received).
 [0] Standard = (Default) INVITE messages that are
generated as a result of Transfer or Redirect are sent
directly to the URI, according to the Refer-To header in the
REFER message, or Contact header in the 3xx response.
 [1] Proxy = Sends a new INVITE to the Proxy.
Note: This option is applicable only if a Proxy server is used
and the parameter AlwaysSendtoProxy is set to 0.
 [2] Routing Table = Uses the Routing table to locate the
destination and then sends a new INVITE to this
destination.
Notes:
 When this parameter is set to [1] and the INVITE sent to the
Proxy fails, the device re-routes the call according to the
Standard mode [0].
 When this parameter is set to [2] and the INVITE fails, the
device re-routes the call according to the Standard mode
[0]. If DNS resolution fails, the device attempts to route the
call to the Proxy. If routing to the Proxy also fails, the
Redirect/Transfer request is rejected.
 When this parameter is set to [2], the XferPrefix parameter
can be used to define different routing rules for redirect
calls.
 This parameter is disregarded if the parameter
AlwaysSendToProxy is set to 1.
Web/EMS: DNS Query Type Enables the use of DNS Naming Authority Pointer (NAPTR)
[DNSQueryType] and Service Record (SRV) queries to resolve Proxy and
Registrar servers and to resolve all domain names that appear
in the SIP Contact and Record-Route headers.
 [0] A-Record (default)
 [1] SRV
 [2] NAPTR
If set to A-Record [0], no NAPTR or SRV queries are
performed.
If set to SRV [1] and the Proxy/Registrar IP address
parameter, Contact/Record-Route headers, or IP address
defined in the Routing tables contain a domain name, an SRV
query is performed. The device uses the first host name
received from the SRV query. The device then performs a DNS
A-record query for the host name to locate an IP address.
If set to NAPTR [2], an NAPTR query is performed. If it is
successful, an SRV query is sent according to the information
received in the NAPTR response. If the NAPTR query fails, an
SRV query is performed according to the configured transport
type.
If the Proxy/Registrar IP address parameter, the domain name
in the Contact/Record-Route headers, or the IP address
defined in the Routing tables contain a domain name with port

definition, the device performs a regular DNS A-record query.
If a specific Transport Type is defined, a NAPTR query is not
performed.
Note: To enable NAPTR/SRV queries for Proxy servers only,
use the parameter ProxyDNSQueryType.
Web: Proxy DNS Query Type Enables the use of DNS Naming Authority Pointer (NAPTR)
[ProxyDNSQueryType] and Service Record (SRV) queries to discover Proxy servers.
 [0] A-Record (default)
 [1] SRV
 [2] NAPTR
If set to A-Record [0], no NAPTR or SRV queries are
performed.
If set to SRV [1] and the Proxy IP address parameter contains
a domain name without port definition (e.g., ProxyIP =
domain.com), an SRV query is performed. The SRV query
returns up to four Proxy host names and their weights. The
device then performs DNS A-record queries for each Proxy
host name (according to the received weights) to locate up to
four Proxy IP addresses. Therefore, if the first SRV query
returns two domain names and the A-record queries return two
IP addresses each, no additional searches are performed.
If set to NAPTR [2], an NAPTR query is performed. If it is
successful, an SRV query is sent according to the information
received in the NAPTR response. If the NAPTR query fails, an
SRV query is performed according to the configured transport
type.
If the Proxy IP address parameter contains a domain name
with port definition (e.g., ProxyIP = domain.com:5080), the
device performs a regular DNS A-record query.
If a specific Transport Type is defined, a NAPTR query is not
performed.
Note: When enabled, NAPTR/SRV queries are used to
discover Proxy servers even if the parameter DNSQueryType
is disabled.
Web/EMS: Use Gateway Name for Determines whether the device uses its IP address or gateway
OPTIONS name in keep-alive SIP OPTIONS messages.
[UseGatewayNameForOptions]  [0] No = (Default) Use the device's IP address in keep-alive
OPTIONS messages.
 [1] Yes = Use 'Gateway Name' (SIPGatewayName) in
keep-alive OPTIONS messages.
 [2] Server = Device's IP address is used in the From and To
headers in keep-alive OPTIONS messages.
The OPTIONS Request-URI host part contains either the
device's IP address or a string defined by the parameter
SIPGatewayName. The device uses the OPTIONS request as
a keep-alive message to its primary and redundant Proxies
(i.e., the parameter EnableProxyKeepAlive is set to 1).
Web/EMS: User Name Defines the user name used for registration and Basic/Digest
[UserName] authentication with a Proxy/Registrar server.
The default is an empty string.
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Notes:
 This parameter is applicable only if single device
registration is used (i.e., the parameter AuthenticationMode
is set to authentication per gateway).
 Instead of configuring this parameter, the Authentication
table can be used (see Authentication on page 304).
Web/EMS: Password Defines the password for Basic/Digest authentication with a
[Password] Proxy/Registrar server. A single password is used for all device
ports.
The default is 'Default_Passwd'.
Note: Instead of configuring this parameter, the Authentication
table can be used (see Authentication on page 304).
Web/EMS: Cnonce Defines the Cnonce string used by the SIP server and client to
[Cnonce] provide mutual authentication.
The value is free format, i.e., 'Cnonce = 0a4f113b'. The default
is 'Default_Cnonce'.
Web/EMS: Mutual Authentication Determines the device's mode of operation when
Mode Authentication and Key Agreement (AKA) Digest
[MutualAuthenticationMode] Authentication is used.
 [0] Optional = (Default) Incoming requests that don't include
AKA authentication information are accepted.
 [1] Mandatory = Incoming requests that don't include AKA
authentication information are rejected.
Web/EMS: Challenge Caching Determines the mode for Challenge Caching, which reduces
Mode the number of SIP messages transmitted through the network.
[SIPChallengeCachingMode] The first request to the Proxy is sent without authorization. The
Proxy sends a 401/407 response with a challenge. This
response is saved for further uses. A new request is re-sent
with the appropriate credentials. Subsequent requests to the
Proxy are automatically sent with credentials (calculated from
the saved challenge). If the Proxy doesn't accept the new
request and sends another challenge, the old challenge is
replaced with the new one.
 [0] None = (Default) Challenges are not cached. Every new
request is sent without preliminary authorization. If the
request is challenged, a new request with authorization data
is sent.
 [1] INVITE Only = Challenges issued for INVITE requests
are cached. This prevents a mixture of REGISTER and
INVITE authorizations.
 [2] Full = Caches all challenges from the proxies.
Note: Challenge Caching is used with all proxies and not only
with the active one.
Proxy IP Table
Web: Proxy IP Table This table parameter configures the Proxy Set table with Proxy
EMS: Proxy IP Set IDs, each with up to five Proxy server IP addresses (or fully
[ProxyIP] qualified domain name/FQDN). Each Proxy Set can be defined
with a transport type (UDP, TCP, or TLS). The format of this
parameter is as follows:
[ProxyIP]
FORMAT ProxyIp_Index = ProxyIp_IpAddress,

ProxyIp_TransportType, ProxyIp_ProxySetId;
[\ProxyIP]
For example:
ProxyIp 0 = 10.33.37.77, -1, 0;
ProxyIp 1 = 10.8.8.10, 0, 2;
ProxyIp 2 = 10.5.6.7, -1, 1;
Notes:
 To assign various attributes (such as Proxy Load Balancing)
per Proxy Set ID, use the parameter ProxySet.
 For a description of this table, see 'Configuring Proxy Sets
Table' on page 208.
Proxy Set Table
Web: Proxy Set Table This table parameter configures the Proxy Set ID table. It is
EMS: Proxy Set used in conjunction with the ProxyIP table ini file parameter,
[ProxySet] which defines the IP addresses per Proxy Set ID.
The ProxySet table ini file parameter defines additional
attributes per Proxy Set ID. This includes, for example, Proxy
keep-alive and load balancing and redundancy mechanisms (if
a Proxy Set contains more than one proxy address).
[ ProxySet ]
FORMAT ProxySet_Index = ProxySet_EnableProxyKeepAlive,
ProxySet_ProxyKeepAliveTime,
ProxySet_ProxyLoadBalancingMethod,
ProxySet_IsProxyHotSwap, ProxySet_SRD,
ProxySet_ClassificationInput,
ProxySet_ProxyRedundancyMode,
ProxySet_KeepAliveFailureResp,
ProxySet_HomingSuccessDetectionRetries;
[\ProxySet]
For example:
ProxySet 0 = 0, 60, 0, 0, 0, , 1,0;
ProxySet 1 = 1, 60, 1, 0, 1, , 0,0;
Notes:
 For configuring the Proxy Set IDs and their IP addresses,
use the parameter ProxyIP.
 For a description of this table, see 'Configuring Proxy Sets
Table' on page 208.
Registrar Parameters
Web: Enable Registration Enables the device to register to a Proxy/Registrar server.
EMS: Is Register Needed  [0] Disable = (Default) The device doesn't register to
[IsRegisterNeeded] Proxy/Registrar server.
 [1] Enable = The device registers to Proxy/Registrar server
when the device is powered up and at every user-defined
interval (configured by the parameter RegistrationTime).
Note: The device sends a REGISTER request for each
channel or for the entire device (according to the
AuthenticationMode parameter).
Web/EMS: Registrar Name Defines the Registrar domain name. If specified, the name is
[RegistrarName] used as the Request-URI in REGISTER messages. If it isn't
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specified (default), the Registrar IP address, or Proxy name or
IP address is used instead.
The valid range is up to 100 characters.
Web: Registrar IP Address Defines the IP address (or FQDN) and port number (optional)
EMS: Registrar IP of the Registrar server. The IP address is in dotted-decimal
[RegistrarIP] notation, e.g., 201.10.8.1:<5080>.
Notes:
 If not specified, the REGISTER request is sent to the
primary Proxy server.
 When a port number is specified, DNS NAPTR/SRV queries
aren't performed, even if the parameter DNSQueryType is
set to 1 or 2.
 If the parameter RegistrarIP is set to an FQDN and is
resolved to multiple addresses, the device also provides
real-time switching (hotswap mode) between different
Registrar IP addresses (the parameter IsProxyHotSwap is
set to 1). If the first Registrar doesn't respond to the
REGISTER message, the same REGISTER message is
sent immediately to the next Proxy. To allow this
mechanism, the parameter EnableProxyKeepAlive must be
set to 0.
 When a specific transport type is defined using the
parameter RegistrarTransportType, a DNS NAPTR query is
not performed even if the parameter DNSQueryType is set
to 2.
Web/EMS: Registrar Transport Determines the transport layer used for outgoing SIP dialogs
Type initiated by the device to the Registrar.
[RegistrarTransportType]  [-1] Not Configured (default)
 [0] UDP
 [1] TCP
 [2] TLS
Note: When set to ‘Not Configured’, the value of the parameter
Web/EMS: Registration Time Defines the time interval (in seconds) for registering to a Proxy
[RegistrationTime] server. The value is used in the SIP Expires header. This
parameter also defines the time interval between Keep-Alive
messages when the parameter EnableProxyKeepAlive is set to
2 (REGISTER).
Typically, the device registers every 3,600 sec (i.e., one hour).
The device resumes registration according to the parameter
RegistrationTimeDivider.
Web: Re-registration Timing [%] Defines the re-registration timing (in percentage). The timing is
EMS: Time Divider a percentage of the re-register timing set by the Registrar
[RegistrationTimeDivider] server.
For example: If this parameter is set to 70% and the
Registration Expires time is 3600, the device re-sends its
registration request after 3600 x 70% (i.e., 2520 sec).
Note: This parameter may be overridden if the parameter
RegistrationTimeThreshold is greater than 0.

Web/EMS: Registration Retry Time Defines the time interval (in seconds) after which a registration
[RegistrationRetryTime] request is re-sent if registration fails with a 4xx response or if
there is no response from the Proxy/Registrar server.
The default is 30 seconds. The range is 10 to 3600.
Web: Registration Time Threshold Defines a threshold (in seconds) for re-registration timing. If
EMS: Time Threshold this parameter is greater than 0, but lower than the computed
[RegistrationTimeThreshold] re-registration timing (according to the parameter
RegistrationTimeDivider), the re-registration timing is set to the
following: timing set by the Registration server in the SIP
Expires header minus the value of the parameter
RegistrationTimeThreshold.
Web: Re-register On INVITE Failure Enables immediate re-registration if no response is received for
EMS: Register On Invite Failure an INVITE request sent by the device.
[RegisterOnInviteFailure]  [0] Disable (default)
 [1] Enable
When enabled, the device immediately expires its re-
registration timer and commences re-registration to the same
Proxy upon any of the following scenarios:
 The response to an INVITE request is 407 (Proxy
Authentication Required) without an authentication header
included.
 The remote SIP UA abandons a call before the device has
received any provisional response (indicative of an
outbound proxy server failure).
 The remote SIP UA abandons a call and the only
provisional response the device has received for the call is
100 Trying (indicative of a home proxy server failure, i.e.,
the failure of a proxy in the route after the outbound proxy).
 The device terminates a call due to the expiration of RFC
3261 Timer B or due to the receipt of a 408 (Request
Timeout) response and the device has not received any
provisional response for the call (indicative of an outbound
proxy server failure).
 The device terminates a call due to the receipt of a 408
(Request Timeout) response and the only provisional
response the device has received for the call is the 100
Trying provisional response (indicative of a home proxy
server failure).
Web: ReRegister On Connection Enables the device to perform SIP re-registration upon
Failure TCP/TLS connection failure.
EMS: Re Register On Connection  [0] Disable (default)
Failure  [1] Enable
[ReRegisterOnConnectionFailure]
Web: Gateway Registration Name Defines the user name that is used in the From and To
EMS: Name headers in SIP REGISTER messages. If no value is specified
[GWRegistrationName] (default) for this parameter, the UserName parameter is used
instead.
Note: This parameter is applicable only for single registration
per device (i.e., AuthenticationMode is set to 1). When the
device registers each channel separately (i.e.,
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AuthenticationMode is set to 0), the user name is set to the
channel's phone number.
Web/EMS: Registration Mode Determines the device's registration and authentication
[AuthenticationMode] method.
 [0] Per Endpoint = Registration and authentication is
performed separately for each endpoint. This is typically
used for FXS interfaces, where each endpoint registers
(and authenticates) separately with its user name and
password.
 [1] Per Gateway = (Default) Single registration and
authentication for the entire device. This is typically used for
FXO interfaces.
 [3] Per FXS = Registration and authentication for FXS
endpoints.
Web: Set Out-Of-Service On Enables setting the endpoint or entire device (i.e., all
Registration Failure endpoints) to out-of-service if registration fails.
EMS: Set OOS On Registration Fail  [0] Disable (default)
[OOSOnRegistrationFail]  [1] Enable
If the registration is per endpoint (i.e., AuthenticationMode is
set to 0) or per Account (see Configuring Hunt Group Settings
on page 237) and a specific endpoint/Account registration fails
(SIP 4xx or no response), then that endpoint is set to out-of-
service until a success response is received in a subsequent
registration request. When the registration is per the entire
device (i.e., AuthenticationMode is set to 1) and registration
fails, all endpoints are set to out-of-service.
Note: The out-of-service method is configured using the
FXSOOSBehavior parameter.
[UnregistrationMode] Enables the device to perform explicit unregisters.
 [1] Enable = The device sends an asterisk ("*") value in the
SIP Contact header, instructing the Registrar server to
remove all previous registration bindings. The device
removes SIP User Agent (UA) registration bindings in a
Registrar, according to RFC 3261. Registrations are soft
state and expire unless refreshed, but they can also be
explicitly removed. A client can attempt to influence the
expiration interval selected by the Registrar. A UA requests
the immediate removal of a binding by specifying an
expiration interval of "0" for that contact address in a
REGISTER request. UA's should support this mechanism
so that bindings can be removed before their expiration
interval has passed. Use of the "*" Contact header field
value allows a registering UA to remove all bindings
associated with an address-of-record (AOR) without
knowing their precise values.
Note: The REGISTER-specific Contact header field value of "*"
applies to all registrations, but it can only be used if the Expires
header field is present with a value of "0".
Web/EMS: Add Empty Authorization Enables the inclusion of the SIP Authorization header in initial
Header registration (REGISTER) requests sent by the device.
[EmptyAuthorizationHeader]  [0] Disable (default)

 [1] Enable
The Authorization header carries the credentials of a user
agent (UA) in a request to a server. The sent REGISTER
message populates the Authorization header with the following
parameters:
 username - set to the value of the private user identity
 realm - set to the domain name of the home network
 uri - set to the SIP URI of the domain name of the home
network
 nonce - set to an empty value
 response - set to an empty value
For example:
Authorization: Digest
[email protected],
realm=”home1.net”, nonce=””,
response=”e56131d19580cd833064787ecc”
Note: This registration header is according to the IMS 3GPP
TS24.229 and PKT-SP-24.220 specifications.
Web: Add initial Route Header Enables the inclusion of the SIP Route header in initial
[InitialRouteHeader] registration or re-registration (REGISTER) requests sent by the
device.
 [1] Enable
When the device sends a REGISTER message, the Route
header includes either the Proxy's FQDN, or IP address and
port according to the configured Proxy Set, for example:
Route: <sip:10.10.10.10;lr;transport=udp>
or
Route: <sip: pcscf-
gm.ims.rr.com;lr;transport=udp>
EMS: Ping Pong Keep Alive Enables the use of the carriage-return and line-feed sequences
[UsePingPongKeepAlive] (CRLF) Keep-Alive mechanism, according to RFC 5626
“Managing Client-Initiated Connections in the Session Initiation
Protocol (SIP)” for reliable, connection-orientated transport
types such as TCP.
 [1] Enable
The SIP user agent/client (i.e., device) uses a simple periodic
message as a keep-alive mechanism to keep their flow to the
proxy or registrar alive (used for example, to keep NAT
bindings open). For connection-oriented transports such as
TCP/TLS this is based on CRLF. This mechanism uses a
client-to-server "ping" keep-alive and a corresponding server-
to-client "pong" message. This ping-pong sequence allows the
client, and optionally the server, to tell if its flow is still active
and useful for SIP traffic. If the client does not receive a pong
in response to its ping, it declares the flow “dead” and opens a
new flow in its place. In the CRLF Keep-Alive mechanism the
client periodically (defined by the PingPongKeepAliveTime
parameter) sends a double-CRLF (the "ping") then waits to
receive a single CRLF (the "pong"). If the client does not
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receive a "pong" within an appropriate amount of time, it
considers the flow failed.
Note: The device sends a CRLF message to the Proxy Set
only if the Proxy Keep-Alive feature (EnableProxyKeepAlive
parameter) is enabled and its transport type is set to TCP or
TLS. The device first sends a SIP OPTION message to
establish the TCP/TLS connection and if it receives any SIP
response, it continues sending the CRLF keep-alive
sequences.
EMS: Ping Pong Keep Alive Time Defines the periodic interval (in seconds) after which a “ping”
[PingPongKeepAliveTime] (double-CRLF) keep-alive is sent to a proxy/registrar, using the
CRLF Keep-Alive mechanism.
The default range is 5 to 2,000,000. The default is 120.
The device uses the range of 80-100% of this user-defined
value as the actual interval. For example, if the parameter
value is set to 200 sec, the interval used is any random time
between 160 to 200 seconds. This prevents an “avalanche” of
keep-alive by multiple SIP UAs to a specific server.
44.8 General SIP Parameters

The general SIP parameters are described in the table below.
Table 44-32: General SIP Parameters
[IgnoreAuthorizationStale] Enables the device to re-send REGISTER messages even if it receives

a SIP 407 with "state=FALSE" from the proxy in response to its
REGISTER message.
 [0] = (Default) Disable. The device stops registering with the proxy
server if it receives a SIP 407 (with "stale=FALSE") response.
 [1] = Enable. The device re-sends the REGISTER message even it
was previously rejected by a 407 with "stale=FALSE".
[GwSDPConnectionMode Defines how the device displays the Connection ("c=") line ("c=") in the
] SDP Offer/Answer model.
 [0] = (Default) The Connection ("c=") line is displayed as follows:
 Offer: In the session description only.
 Answer: In the session description and in each media ("m=")
description.
 [1] = For Offer and Answer, the Connection ("c=") line is displayed
only in the session description; not in any media ("m=") descriptions.
 [2] = The Connection ("c=") line is displayed only in media ("m=")
descriptions.
Web: SIP Remote Reset Enables a specific device action upon the receipt of a SIP NOTIFY
CLI: sip-remote-reset request, where the action depends on the value received in the Event
header.
[EnableSIPRemoteReset
]  [0] Disable (default)
 [1] Enable
The action depends on the Event header value:
 'check-sync;reboot=false': triggers the regular Automatic Update

feature (if Automatic Update has been enabled on the device)
 'check-sync;reboot=true': triggers a device reset
 'cwmp-connect': triggers connection with TR-069
Note: The Event header value is proprietary to AudioCodes.
Web/EMS: Max SIP Defines the maximum size (in Kbytes) for each SIP message that can
Message Length [KB] be sent over the network. The device rejects messages exceeding this
[MaxSIPMessageLength user-defined size.
] The valid value range is 1 to 50. The default is 50.
[SIPForceRport] Determines whether the device sends SIP responses to the UDP port
from where SIP requests are received even if the 'rport' parameter is not
present in the SIP Via header.
 [0] = (Default) Disabled. The device sends the SIP response to the
UDP port defined in the Via header. If the Via header contains the
'rport' parameter, the response is sent to the UDP port from where
the SIP request is received.
 [1] = Enabled. SIP responses are sent to the UDP port from where
SIP requests are received even if the 'rport' parameter is not present
in the Via header.
Web: Reject Cancel after Determines whether the device accepts or rejects a SIP CANCEL
Connect request received after the receipt of a 200 OK, during an established
CLI: reject-cancel-after- call.
connect  [0] = (Default) Accepts the CANCEL, by responding with a 200 OK
[RejectCancelAfterConn and terminating the call session.
ect]  [1] = Rejects the CANCEL, by responding with a SIP 481
Call/Transaction Does Not Exist, and maintaining the call session.
Web: Verify Received Enables the device to reject SIP requests (such as ACK, BYE, or re-
RequestURI INVITE) whose user part in the Request-URI is different from the user
CLI: verify-rcvd-requri part received in the Contact header of the last sent SIP request.
[VerifyReceevedReques  [0] Disable = (Default) Even if the user is different, the device
tUri] accepts the SIP request.
 [1] Enable = If the user is different, the device rejects the SIP
request (BYE is responded with 481; re-INVITE is responded with
404; ACK is ignored).
Web: Max Number of Defines the maximum number of simultaneous active calls supported by
Active Calls the device. If the maximum number of calls is reached, new calls are
EMS: Maximum not established.
Concurrent Calls The valid range is 1 to the maximum number of supported channels.
[MaxActiveCalls] The default is the maximum available channels (i.e., no restriction on
the maximum number of calls).
Web: QoS statistics in Enables the device to include call quality of service (QoS) statistics in
SIP Release Call SIP BYE and SIP 200 OK response to BYE, using the proprietary SIP
[QoSStatistics] header X-RTP-Stat.
 [1] = Enable
The X-RTP-Stat header provides the following statistics:
 Number of received and sent voice packets
 Number of received and sent voice octets
 Received packet loss, jitter (in ms), and latency (in ms)
The X-RTP-Stat header contains the following fields:
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 PS=<voice packets sent>
 OS=<voice octets sent>
 PR=<voice packets received>
 OR=<voice octets received>
 PL=<receive packet loss>
 JI=<jitter in ms>
 LA=<latency in ms>
Below is an example of the X-RTP-Stat header in a SIP BYE message:
BYE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP
10.33.4.126;branch=z9hG4bKac2127550866
Max-Forwards: 70
From:
<sip:[email protected];user=phone>;tag=1c2113553324
CSeq: 1 BYE
X-RTP-Stat:
PS=207;OS=49680;;PR=314;OR=50240;PL=0;JI=600;LA=40;
Supported: em,timer,replaces,path,resource-priority
Allow:
REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK
,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Sip-Gateway-/v.6.2A.008.006
Reason: Q.850 ;cause=16 ;text="local"
Content-Length: 0
Web/EMS: PRACK Mode Determines the PRACK (Provisional Acknowledgment) mechanism
[PrackMode] mode for SIP 1xx reliable responses.
 [0] Disable
 [1] Supported (default)
 [2] Required
Notes:
 The Supported and Required headers contain the '100rel' tag.
 The device sends PRACK messages if 180/183 responses are
received with '100rel' in the Supported or Required headers.
Web/EMS: Enable Early Enables the Early Media feature.
Media
[EnableEarlyMedia]
Enables the device to send a 183 Session Progress response with SDP
instead of a 180 Ringing, allowing the media stream to be established
prior to the answering of the call.
 [1] Enable
Notes:
 To send a 183 response, you must also set the parameter
ProgressIndicator2IP to 1. If it is equal to 0, 180 Ringing response is
sent.
 This feature can also be configured as an IP Profile and/or Tel
Profile.
Web: 183 Message Defines the response of the device upon receipt of a SIP 183 response.

Behavior  [0] Progress = (Default) A 183 response (without SDP) does not
EMS: SIP 183 Behaviour cause the device to play a ringback tone.
[SIP183Behaviour]  [1] Alert = 183 response is handled by the device as if a 180 Ringing
response is received, and the device plays a ringback tone.
Web: Session-Expires Defines the numerical value sent in the Session-Expires header in the
Time first INVITE request or response (if the call is answered).
EMS: Sip Session Expires The valid range is 1 to 86,400 sec. The default is 0 (i.e., the Session-
[SIPSessionExpires] Expires header is disabled).
Web: Minimum Session- Defines the time (in seconds) that is used in the Min-SE header. This
Expires header defines the minimum time that the user agent refreshes the
EMS: Minimal Session session.
Refresh Value The valid range is 10 to 100,000. The default is 90.
[MinSE]
Web/EMS: Session Defines a session expiry timeout. The device disconnects the session
Expires Disconnect Time (sends a SIP BYE) if the refresher does not send a refresh request
CLI: session-exp- before one-third (1/3) of the session expires time, or before the time
disconnect-time configured by this parameter (the minimum of the two).
[SessionExpiresDiscon The valid range is 0 to 32 (in seconds). The default is 32.
nectTime]
Web/EMS: Session Determines the SIP method used for session-timer updates.
Expires Method  [0] Re-INVITE = (Default) Uses Re-INVITE messages for session-
[SessionExpiresMethod timer updates.
]  [1] UPDATE = Uses UPDATE messages.
Notes:
 The device can receive session-timer refreshes using both methods.
 The UPDATE message used for session-timer is excluded from the
SDP body.
[RemoveToTagInFailure Determines whether the device removes the ‘to’ header tag from final
Response] SIP failure responses to INVITE transactions.
 [0] = (Default) Do not remove tag.
 [1] = Remove tag.
[EnableRTCPAttribute] Enables the use of the 'rtcp' attribute in the outgoing SDP.
 [1] = Enable
EMS: Options User Part Defines the user part value of the Request-URI for outgoing SIP
[OPTIONSUserPart] OPTIONS requests. If no value is configured, the endpoint number is
used.
A special value is ‘empty’, indicating that no user part in the Request-
URI (host part only) is used.
The valid range is a 30-character string. The default is an empty string
(‘’).
Web: Fax Signaling Determines the SIP signaling method for establishing and transmitting a
Method fax session after a fax is detected.
EMS: Fax Used  [0] No Fax = (Default) No fax negotiation using SIP signaling. Fax
[IsFaxUsed] transport method is according to the parameter FaxTransportMode.
 [1] T.38 Relay = Initiates T.38 fax relay.
 [2] G.711 Transport = Initiates fax/modem using the coder G.711 A-
law/Mu-law with adaptations (see Note below).
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 [3] Fax Fallback = Initiates T.38 fax relay. If the T.38 negotiation
fails, the device re-initiates a fax session using the coder G.711 A-
law/µ-law with adaptations (see the Note below).
Notes:
 Fax adaptations (for options 2 and 3):
 Echo Canceller = On
 Silence Compression = Off
 Echo Canceller Non-Linear Processor Mode = Off
 Dynamic Jitter Buffer Minimum Delay = 40
 Dynamic Jitter Buffer Optimization Factor = 13
 If the device initiates a fax session using G.711 (option 2 and
possibly 3), a 'gpmd' attribute is added to the SDP in the following
format:
 For A-law: 'a=gpmd:8 vbd=yes;ecan=on'
 For µ-law: 'a=gpmd:0 vbd=yes;ecan=on'
 When this parameter is set to 1, 2, or 3, the parameter
FaxTransportMode is ignored.
 When this parameter is set to 0, T.38 might still be used without the
control protocol's involvement. To completely disable T.38, set
FaxTransportMode to a value other than 1.
 This parameter can also be configured per IP Profile (using the
IPProfile parameter).
 For more information on fax transport methods, see 'Fax/Modem
Transport Modes' on page 169.
[HandleG711asVBD] Enables the handling of G.711 as G.711 VBD coder.
 [0] = (Default) Disable. The device negotiates G.711 as a regular
audio coder and sends an answer only with G.729 coder. For
example, if the device is configured with G.729 and G.711 VBD
coders and it receives an INVITE with an SDP offer containing
G.729 and “regular” G.711 coders, it sends an SDP answer
containing only the G.729 coder.
 [1] = Enable. The device assumes that the G.711 coder received in
the INVITE SDP offer is a VBD coder. For example, if the device is
configured with G.729 and G.711 VBD coders and it receives an
INVITE with an SDP offer containing G.729 and “regular” G.711
coders, it sends an SDP answer containing G.729 and G.711 VBD
coders, allowing a subsequent bypass (passthrough) session if
fax/modem signals are detected during the call.
Note: This parameter is applicable only if G.711 VBD coder(s) with
regular G.711 payload types 0 or 8 are configured for the device (using
the CodersGroup parameter).
[FaxVBDBehavior] Determines the device's fax transport behavior when G.711 VBD coder
is negotiated at call start.
 [0] = (Default) If the device is configured with a VBD coder (see the
CodersGroup parameter) and is negotiated OK at call start, then
both fax and modem signals are sent over RTP using the bypass
payload type (and no mid-call VBD or T.38 Re-INVITEs occur).
 [1] = If the IsFaxUsed parameter is set to 1, the channel opens with
the FaxTransportMode parameter set to 1 (relay). This is required to
detect mid-call fax tones and to send T.38 Re-INVITE messages
upon fax detection. If the remote party supports T.38, the fax is
relayed over T.38.
Notes:

 If VBD coder negotiation fails at call start and if the IsFaxUsed
parameter is set to 1 (or 3), then the channel opens with the
FaxTransportMode parameter set to 1 (relay) to allow future
detection of fax tones and sending of T.38 Re-INVITES. In such a
scenario, the FaxVBDBehavior parameter has no effect.
 This feature can be used only if the remote party supports T.38 fax
relay; otherwise, the fax fails.
[NoAudioPayloadType] Defines the payload type of the outgoing SDP offer.
The valid value range is 96 to 127 (dynamic payload type). The default
is 0 (i.e. NoAudio is not supported). For example, if set to 120, the
following is added to the INVITE SDP:
a=rtpmap:120 NoAudio/8000\r\n
Note: For incoming SDP offers, NoAudio is always supported.
Web: SIP Transport Type Determines the default transport layer for outgoing SIP calls initiated by
EMS: Transport Type the device.
[SIPTransportType]  [0] UDP (default)
 [1] TCP
 [2] TLS (SIPS)
Notes:
 It's recommended to use TLS for communication with a SIP Proxy
and not for direct device-to-device communication.
 For received calls (i.e., incoming), the device accepts all these
protocols.
 The value of this parameter is also used by the SAS application as
the default transport layer for outgoing SIP calls.
Web: SIP UDP Local Port Defines the local UDP port for SIP messages.
EMS: Local SIP Port The valid range is 1 to 65534. The default is 5060.
[LocalSIPPort]
Web: SIP TCP Local Port Defines the local TCP port for SIP messages.
EMS: TCP Local SIP Port The valid range is 1 to 65535. The default is 5060.
[TCPLocalSIPPort]
Web: SIP TLS Local Port Defines the local TLS port for SIP messages.
EMS: TLS Local SIP Port The valid range is 1 to 65535. The default is 5061.
[TLSLocalSIPPort]
Note: The value of this parameter must be different from the value of
the parameter TCPLocalSIPPort.
Web/EMS: Enable SIPS Enables secured SIP (SIPS URI) connections over multiple hops.
[EnableSIPS]  [0] Disable (default)
 [1] Enable
When the SIPTransportType parameter is set to 2 (i.e., TLS) and the
parameter EnableSIPS is disabled, TLS is used for the next network
hop only. When the parameter SIPTransportType is set to 2 or 1 (i.e.,
TCP or TLS) and EnableSIPS is enabled, TLS is used through the
entire connection (over multiple hops).
Note: If this parameter is enabled and the parameter SIPTransportType
is set to 0 (i.e., UDP), the connection fails.
Web/EMS: Enable TCP Enables the reuse of the same TCP connection for all calls to the same
Connection Reuse destination.
[EnableTCPConnection  [0] Disable = Uses a separate TCP connection for each call.
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Reuse]  [1] Enable = (Default) Uses the same TCP connection for all calls.
Note: For the SAS application, this feature is configured using the
SASConnectionReuse parameter.
Web: Fake TCP alias Enables the re-use of the same TCP/TLS connection for sessions with
[FakeTCPalias] the same user, even if the "alias" parameter is not present in the SIP
Via header of the first INVITE.
 [0] Disable = (Default) TCP/TLS connection reuse is done only if the
"alias" parameter is present in the Via header of the first INVITE.
 [1] Enable
Note: To enable TCP/TLS connection re-use, set the
EnableTCPConnectionReuse parameter to 1.
Web/EMS: Reliable Enables setting of all TCP/TLS connections as persistent and therefore,
Connection Persistent not released.
Mode  [0] = (Default) Disable. All TCP connections (except those that are
[ReliableConnectionPer set to a proxy IP) are released if not used by any SIP
sistentMode] dialog\transaction.
 [1] = Enable - TCP connections to all destinations are persistent and
not released unless the device reaches 70% of its maximum TCP
resources.
While trying to send a SIP message connection, reuse policy
determines whether live connections to the specific destination are re-
used.
Persistent TCP connection ensures less network traffic due to fewer
setting up and tearing down of TCP connections and reduced latency
on subsequent requests due to avoidance of initial TCP handshake. For
TLS, persistent connection may reduce the number of costly TLS
handshakes to establish security associations, in addition to the initial
TCP connection set up.
Note: If the destination is a Proxy server, the TCP/TLS connection is
persistent regardless of the settings of this parameter.
Web/EMS: TCP Timeout Defines the Timer B (INVITE transaction timeout timer) and Timer F
[SIPTCPTimeout] (non-INVITE transaction timeout timer), as defined in RFC 3261, when
the SIP Transport Type is TCP.
The valid range is 0 to 40 sec. The default is 64 multiplied by the
SipT1Rtx parameter value. For example, if SipT1Rtx is set to 500 msec,
then the default of SIPTCPTimeout is 32 sec.
Web: SIP Destination Port Defines the SIP destination port for sending initial SIP requests.
EMS: Destination Port The valid range is 1 to 65534. The default port is 5060.
[SIPDestinationPort] Note: SIP responses are sent to the port specified in the Via header.
Web: Use user=phone in Determines whether the 'user=phone' string is added to the SIP URI
SIP URL and SIP To header.
EMS: Is User Phone  [0] No = 'user=phone' string is not added.
[IsUserPhone]  [1] Yes = (Default) 'user=phone' string is part of the SIP URI and SIP
To header.
Web: Use user=phone in Determines whether the 'user=phone' string is added to the From and
From Header Contact SIP headers.
EMS: Is User Phone In  [0] No = (Default) Doesn't add 'user=phone' string.
From  [1] Yes = 'user=phone' string is part of the From and Contact
[IsUserPhoneInFrom] headers.
Web: Use Tel URI for Determines the format of the URI in the P-Asserted-Identity and P-

Asserted Identity Preferred-Identity headers.
[UseTelURIForAssertedI  [0] Disable = (Default) 'sip:'
D]  [1] Enable = 'tel:'
Web: Tel to IP No Answer Defines the time (in seconds) that the device waits for a 200 OK
Timeout response from the called party (IP side) after sending an INVITE
EMS: IP Alert Timeout message. If the timer expires, the call is released.
[IPAlertTimeout] The valid range is 0 to 3600. The default is 180.
Web: Enable Remote Enables Remote-Party-Identity headers for calling and called numbers
Party ID for Tel-to-IP calls.
EMS: Enable RPI Header  [0] Disable (default).
[EnableRPIheader]  [1] Enable = Remote-Party-Identity headers are generated in SIP
INVITE messages for both called and calling numbers.
Web: Enable History-Info Enables usage of the History-Info header.
Header  [0] Disable (default)
EMS: Enable History Info
 [1] Enable
[EnableHistoryInfo]
User Agent Client (UAC) Behavior:
 Initial request: The History-Info header is equal to the Request-URI.
If a PSTN Redirect number is received, it is added as an additional
History-Info header with an appropriate reason.
 Upon receiving the final failure response, the device copies the
History-Info as is, adds the reason of the failure response to the last
entry, and concatenates a new destination to it (if an additional
request is sent). The order of the reasons is as follows:
a. Q.850 Reason
b. SIP Reason
c. SIP Response code
 Upon receiving the final response (success or failure), the device
searches for a Redirect reason in the History-Info (i.e., 3xx/4xx SIP
reason). If found, it is passed to ISDN according to the following
table:
SIP Reason Code ISDN Redirecting Reason
302 - Moved Temporarily Call Forward Universal (CFU)
408 - Request Timeout Call Forward No Answer (CFNA)
480 - Temporarily Unavailable

487 - Request Terminated
486 - Busy Here Call Forward Busy (CFB)
600 - Busy Everywhere
 If history reason is a Q.850 reason, it is translated to the SIP reason

(according to the SIP-ISDN tables) and then to ISDN Redirect
reason according to the table above.
User Agent Server (UAS) Behavior:
 The History-Info header is sent only in the final response.
 Upon receiving a request with History-Info, the UAS checks the
policy in the request. If a 'session', 'header', or 'history' policy tag is
found, the (final) response is sent without History-Info; otherwise, it
is copied from the request.
Web: Use Tgrp Determines whether the SIP 'tgrp' parameter is used. This SIP
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Information parameter specifies the Hunt Group to which the call belongs
EMS: Use SIP Tgrp (according to RFC 4904). For example, the SIP message below
[UseSIPTgrp] indicates that the call belongs to Hunt Group ID 1:
INVITE sip::+16305550100;tgrp=1;trunk-
[email protected];user=phone SIP/2.0
 [0] Disable = (Default) The 'tgrp' parameter isn't used.
 [1] Send Only = The Hunt Group number or name (configured in the
Hunt Group Settings) is added to the 'tgrp' parameter value in the
Contact header of outgoing SIP messages. If a Hunt Group number /
name is not associated with the call, the 'tgrp' parameter isn't
included. If a 'tgrp' value is specified in incoming messages, it is
ignored.
 [2] Send and Receive = The functionality of outgoing SIP messages
is identical to the functionality described for option [1]. In addition, for
incoming SIP INVITEs, if the Request-URI includes a 'tgrp'
parameter, the device routes the call according to that value (if
possible). The Contact header in the outgoing SIP INVITE (Tel-to-IP
call) contains "tgrp=<source trunk group ID>;trunk-
context=<gateway IP address>”. The <source trunk group ID> is the
Hunt Group ID where incoming calls from Tel is received. For IP-Tel
calls, the SIP 200 OK device's response contains “tgrp=<destination
trunk group ID>;trunk-context=<gateway IP address>”. The
<destination trunk group ID> is the Hunt Group ID used for outgoing
Tel calls. The <gateway IP address> in “trunk-context” can be
configured using the SIPGatewayName parameter.
Note: IP-to-Tel configuration (using the PSTNPrefix parameter)
overrides the 'tgrp' parameter in incoming INVITE messages.
Web/EMS: TGRP Routing Determines the precedence method for routing IP-to-Tel calls -
Precedence according to the IP to Hunt Group Routing Table or according to the SIP
[TGRProutingPrecedenc 'tgrp' parameter.
e]  [0] = (Default) IP-to-Tel routing is determined by the IP to Hunt
Group Routing Table (PSTNPrefix parameter). If a matching rule is
not found in this table, the device uses the Hunt Group parameters
for routing the call.
 [1] = The device first places precedence on the 'tgrp' parameter for
IP-to-Tel routing. If the received INVITE Request-URI does not
contain the 'tgrp' parameter or if the Hunt Group number is not
defined, then the IP to Hunt Group Routing Table is used for routing
the call.
Below is an example of an INVITE Request-URI with the 'tgrp'
parameter, indicating that the IP call should be routed to Hunt Group 7:
INVITE sip:200;tgrp=7;trunk-
[email protected];user=phone SIP/2.0
Notes:
 For enabling routing based on the 'tgrp' parameter, the UseSIPTgrp
parameter must be set to 2.
 For IP-to-Tel routing based on the 'dtg' parameter (instead of the
'tgrp' parameter), use the parameter UseBroadsoftDTG.
Determines whether the device uses the 'dtg' parameter for routing IP-
[UseBroadsoftDTG] to-Tel calls to a specific Hunt Group.
 [1] Enable
When this parameter is enabled, if the Request-URI in the received SIP

INVITE includes the 'dtg' parameter, the device routes the call to the
Hunt Group according to its value. This parameter is used instead of the
'tgrp/trunk-context' parameters. The 'dtg' parameter appears in the
INVITE Request-URI (and in the To header).
For example, the received SIP message below routes the call to Hunt
Group ID 56:
INVITE sip:[email protected];dtg=56;user=phone SIP/2.0
Note: If the Hunt Group is not found based on the 'dtg' parameter, the
IP to Hunt Group Routing Table is used instead for routing the call to
the appropriate Hunt Group.
Web/EMS: Enable GRUU Determines whether the Globally Routable User Agent URIs (GRUU)
[EnableGRUU] mechanism is used, according to RFC 5627. This is used for obtaining a
GRUU from a registrar and for communicating a GRUU to a peer within
a dialog.
 [1] Enable
A GRUU is a SIP URI that routes to an instance-specific UA and can be
reachable from anywhere. There are a number of contexts in which it is
desirable to have an identifier that addresses a single UA (using GRUU)
rather than the group of UA’s indicated by an Address of Record (AOR).
For example, in call transfer where user A is talking to user B, and user
A wants to transfer the call to user C. User A sends a REFER to user C:
REFER sip:[email protected] SIP/2.0
From: sip:[email protected];tag=99asd
To: sip:[email protected]
Refer-To: (URI that identifies B's UA)
The Refer-To header needs to contain a URI that user C can use to
place a call to user B. This call needs to route to the specific UA
instance that user B is using to talk to user A. User B should provide
user A with a URI that has to be usable by anyone. It needs to be a
GRUU.
 Obtaining a GRUU: The mechanism for obtaining a GRUU is
through registrations. A UA can obtain a GRUU by generating a
REGISTER request containing a Supported header field with the
value “gruu”. The UA includes a “+sip.instance” Contact header
parameter of each contact for which the GRUU is desired. This
Contact parameter contains a globally unique ID that identifies the
UA instance. The global unique ID is created from one of the
following:
 If the REGISTER is per the device’s client (endpoint), it is the
MAC address concatenated with the phone number of the client.
 If the REGISTER is per device, it is the MAC address only.
 When using TP, “User Info” can be used for registering per
endpoint. Thus, each endpoint can get a unique id – its phone
number. The globally unique ID in TP is the MAC address
concatenated with the phone number of the endpoint.
If the remote server doesn’t support GRUU, it ignores the parameters of
the GRUU. Otherwise, if the remote side also supports GRUU, the
REGISTER responses contain the “gruu” parameter in each Contact
header. This parameter contains a SIP or SIPS URI that represents a
GRUU corresponding to the UA instance that registered the contact.
The server provides the same GRUU for the same AOR and instance-id
when sending REGISTER again after registration expiration. RFC 5627
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specifies that the remote target is a GRUU target if its’ Contact URL has
the "gr" parameter with or without a value.
 Using GRUU: The UA can place the GRUU in any header field that
can contain a URI. It must use the GRUU in the following messages:
INVITE request, its 2xx response, SUBSCRIBE request, its 2xx
response, NOTIFY request, REFER request and its 2xx response.
EMS: Is CISCO Sce Determines whether a Cisco gateway exists at the remote side.
Mode  [0] = (Default) No Cisco gateway exists at the remote side.
[IsCiscoSCEMode]  [1] = A Cisco gateway exists at the remote side.
When a Cisco gateway exists at the remote side, the device must set
the value of the 'annexb' parameter of the fmtp attribute in the SDP to
'no'. This logic is used if Silence Suppression for the used coder is
configured to 2 (enable without adaptation). In this case, Silence
Suppression is used on the channel but not declared in the SDP.
Note: The IsCiscoSCEMode parameter is applicable only when the
selected coder is G.729.
Web: User-Agent Defines the string that is used in the SIP User-Agent and Server
Information response headers. When configured, the string <UserAgentDisplayInfo
EMS: User Agent Display value>/software version' is used, for example:
Info User-Agent: myproduct/v.6.40.010.006
[UserAgentDisplayInfo]
If not configured, the default string, <AudioCodes product-
name>/software version' is used, for example:
User-Agent: Audiocodes-Sip-Gateway-
MediaPack/v.6.40.010.006
The maximum string length is 50 characters.
Note: The software version number and preceding forward slash (/)
cannot be modified. Therefore, it is recommended not to include a
forward slash in the parameter's value (to avoid two forward slashes in
the SIP header, which may cause problems).
Web/EMS: SDP Session Defines the value of the Owner line ('o' field) in outgoing SDP
Owner messages.
[SIPSDPSessionOwner] The valid range is a string of up to 39 characters. The default is
'AudiocodesGW'.
For example:
o=AudiocodesGW 1145023829 1145023705 IN IP4
10.33.4.126
Enables the device to ignore new SDP re-offers (from the media
[EnableSDPVersionNeg negotiation perspective) in certain scenarios (such as session expires).
otiation] According to RFC 3264, once an SDP session is established, a new
SDP offer is considered a new offer only when the SDP origin value is
incremented. In scenarios such as session expires, SDP negotiation is
irrelevant and thus, the origin field is not changed.
Even though some SIP devices don’t follow this behavior and don’t
increment the origin value even in scenarios where they want to re-
negotiate, the device can assume that the remote party operates
according to RFC 3264, and in cases where the origin field is not
incremented, the device does not re-negotiate SDP capabilities.
 [0] Disable = (Default) The device negotiates any new SDP re-offer,
regardless of the origin field.
 [1] Enable = The device negotiates only an SDP re-offer with an
incremented origin field.

Web/EMS: Subject Defines the Subject header value in outgoing INVITE messages. If not
[SIPSubject] specified, the Subject header isn't included (default).
The maximum length is up to 50 characters.
[CoderPriorityNegotiatio Defines the priority for coder negotiation in the incoming SDP offer,
n] between the device's or remote UA's coder list.
 [0] = (Default) Coder negotiation is given higher priority to the
remote UA's list of supported coders.
 [1] = Coder negotiation is given higher priority to the device's (local)
supported coders list.
 Note: This parameter is applicable only to the Gateway/IP-to-IP
application.
Web: Send All Coders on Enables coder re-negotiation in the sent re-INVITE for retrieving an on-
Retrieve hold call.
[SendAllCodersOnRetri  [0] Disable = (Default) Sends only the initially chosen coder when
eve] the call was first established and then put on-hold.
 [1] Enable = Includes all supported coders in the SDP of the re-
INVITE sent to the call made un-hold (retrieved). The used coder is
therefore, re-negotiated.
This parameter is useful in the following call scenario example:
1 Party A calls party B and coder G.711 is chosen.
2 Party B is put on-hold while Party A blind transfers Party B to Party
C.
3 Party C answers and Party B is made un-hold. However, as Party C
supports only G.729 coder, re-negotiation of the supported coder is
required.
Web: Multiple Determines whether the 'mptime' attribute is included in the outgoing
Packetization Time SDP.
Format  [0] None = (Default) Disabled.
EMS: Multi Ptime Format
 [1] PacketCable = Includes the 'mptime' attribute in the outgoing
[MultiPtimeFormat] SDP - PacketCable-defined format.
The mptime' attribute enables the device to define a separate
packetization period for each negotiated coder in the SDP. The 'mptime'
attribute is only included if this parameter is enabled even if the remote
side includes it in the SDP offer. Upon receipt, each coder receives its
'ptime' value in the following precedence: from 'mptime' attribute, from
'ptime' attribute, and then from default value.
EMS: Enable P Time Determines whether the 'ptime' attribute is included in the SDP.
[EnablePtime]  [0] = Remove the 'ptime' attribute from SDP.
 [1] = (Default) Include the 'ptime' attribute in SDP.
Web/EMS: 3xx Behavior Determines the device's behavior regarding call identifiers when a 3xx
[3xxBehavior] response is received for an outgoing INVITE request. The device can
either use the same call identifiers (Call-ID, To, and From tags) or
change them in the new initiated INVITE.
 [0] Forward = (Default) Use different call identifiers for a redirected
INVITE message.
 [1] Redirect = Use the same call identifiers.
Web/EMS: Enable P- Enables the inclusion of the P-Charging-Vector header to all outgoing
Charging Vector INVITE messages.
[EnablePChargingVecto  [0] Disable (default)
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r]  [1] Enable
Web/EMS: Retry-After Defines the time (in seconds) used in the Retry-After header when a
Time 503 (Service Unavailable) response is generated by the device.
[RetryAfterTime] The time range is 0 to 3,600. The default is 0.
Web/EMS: Fake Retry Determines whether the device, upon receipt of a SIP 503 response
After [sec] without a Retry-After header, behaves as if the 503 response included a
[FakeRetryAfter] Retry-After header and with the period (in seconds) specified by this
parameter.
 Any positive value (in seconds) for defining the period
When enabled, this feature allows the device to operate with Proxy
servers that do not include the Retry-After SIP header in SIP 503
(Service Unavailable) responses to indicate an unavailable service.
The Retry-After header is used with the 503 (Service Unavailable)
response to indicate how long the service is expected to be unavailable
to the requesting SIP client. The device maintains a list of available
proxies, by using the Keep-Alive mechanism. The device checks the
availability of proxies by sending SIP OPTIONS every keep-alive
timeout to all proxies.
If the device receives a SIP 503 response to an INVITE, it also marks
that the proxy is out of service for the defined "Retry-After" period.
Web/EMS: Enable P- Determines the device usage of the P-Associated-URI header. This
Associated-URI Header header can be received in 200 OK responses to REGISTER requests.
[EnablePAssociatedURI When enabled, the first URI in the P-Associated-URI header is used in
Header] subsequent requests as the From/P-Asserted-Identity headers value.
 [1] Enable
Note: P-Associated-URIs in registration responses is handled only if the
device is registered per endpoint (using the User Information file).
Web/EMS: Source Determines from which SIP header the source (calling) number is
Number Preference obtained in incoming INVITE messages.
[SourceNumberPreferen  If not configured (i.e., empty string) or if any string other than "From"
ce] or "Pai2" is configured, the calling number is obtained from a specific
header using the following logic:
a. P-Preferred-Identity header.
b. If the above header is not present, then the first P-Asserted-
Identity header is used.
c. If the above header is not present, then the Remote-Party-ID
header is used.
d. If the above header is not present, then the From header is
used.
 "From" = The calling number is obtained from the From header.
 "Pai2" = The calling number is obtained using the following logic:
a. If a P-Preferred-Identity header is present, the number is
obtained from it.
b. If no P-Preferred-Identity header is present and two P-Asserted-
Identity headers are present, the number is obtained from the
second P-Asserted-Identity header.
c. If only one P-Asserted-Identity header is present, the calling
number is obtained from it.
Notes:

 The "From" and "Pai2" values are not case-sensitive.
 Once a URL is selected, all the calling party parameters are set from
this header. If P-Asserted-Identity is selected and the Privacy header
is set to 'id', the calling number is assumed restricted.
Determines the SIP header used for obtaining the called number
[SelectSourceHeaderFo (destination) for IP-to-Tel calls.
rCalledNumber]  [0] Request-URI header = (Default) Obtains the destination number
from the user part of the Request-URI.
 [1] To header = Obtains the destination number from the user part of
the To header.
 [2] P-Called-Party-ID header = Obtains the destination number from
the P-Called-Party-ID header.
Web/EMS: Forking Determines how the device handles the receipt of multiple SIP 18x
Handling Mode forking responses for Tel-to-IP calls. The forking 18x response is the
[ForkingHandlingMode] response with a different SIP to-tag than the previous 18x response.
These responses are typically generated (initiated) by Proxy /
Application servers that perform call forking, sending the device's
originating INVITE (received from SIP clients) to several destinations,
using the same CallID.
 [0] Parallel handling = (Default) If SIP 18x with SDP is received, the
device opens a voice stream according to the received SDP and
disregards any subsequently received 18x forking responses (with or
without SDP). If the first response is 180 without SDP, the device
responds according to the PlayRBTone2TEL parameter and
disregards the subsequent forking 18x responses.
 [1] Sequential handling = If 18x with SDP is received, the device
opens a voice stream according to the received SDP. The device re-
opens the stream according to subsequently received 18x responses
with SDP, or plays a ringback tone if 180 response without SDP is
received. If the first received response is 180 without SDP, the
device responds according to the PlayRBTone2TEL parameter and
processes the subsequent 18x forking responses.
Note: Regardless of this parameter setting, once a SIP 200 OK
response is received, the device uses the RTP information and re-
opens the voice stream, if necessary.
Web: Forking Timeout Defines the timeout (in seconds) that is started after the first SIP 2xx
[ForkingTimeOut] response has been received for a User Agent when a Proxy server
performs call forking (Proxy server forwards the INVITE to multiple SIP
User Agents). The device sends a SIP ACK and BYE in response to
any additional SIP 2xx received from the Proxy within this timeout. Once
this timeout elapses, the device ignores any subsequent SIP 2xx.
The number of supported forking calls per channel is 4. In other words,
for an INVITE message, the device can receive up to 4 forking
responses from the Proxy server.
[ForkingDelayTimeForInvite] Defines the interval (in seconds) to wait before sending INVITE
messages to the other members of the forking group. The
INVITE is immediately sent to the first member.
The valid value range is 0 to 40. The default is 0 (i.e., sends
immediately).
Web: Tel2IP Call Forking Enables Tel-to-IP call forking, whereby a Tel call can be routed to
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Mode multiple IP destinations.
[Tel2IPCallForkingMode  [0] Disable (default)
]  [1] Enable
Note: Once enabled, routing rules must be assigned Forking Groups in
the Outbound IP Routing table.
Web/EMS: Enable Enables the usage of the SIP Reason header.
Reason Header  [0] Disable
[EnableReasonHeader]  [1] Enable (default)
Web/EMS: Gateway Defines a name for the device (e.g., device123.com). This name is used
Name as the host part of the SIP URI in the From header. If not specified, the
CLI: gw-name device's IP address is used instead (default).
[SIPGatewayName] Notes:
 Ensure that the parameter value is the one with which the Proxy has
been configured with to identify the device.
 This parameter can also be configured for an IP Group (in the IP
Group table).
[ZeroSDPHandling] Determines the device's response to an incoming SDP that includes an
IP address of 0.0.0.0 in the SDP's Connection Information field (i.e.,
"c=IN IP4 0.0.0.0").
 [0] = (Default) Sets the IP address of the outgoing SDP's c= field to
0.0.0.0.
 [1] = Sets the IP address of the outgoing SDP c= field to the IP
address of the device. If the incoming SDP doesn’t contain the
"a=inactive" line, the returned SDP contains the "a=recvonly" line.
Web/EMS: Enable Determines whether the device sends the initial INVITE message with
Delayed Offer or without an SDP. Sending the first INVITE without SDP is typically
[EnableDelayedOffer] done by clients for obtaining the far-end's full list of capabilities before
sending their own offer. (An alternative method for obtaining the list of
supported capabilities is by using SIP OPTIONS, which is not supported
by every SIP agent.)
 [0] Disable = (Default) The device sends the initial INVITE message
with an SDP.
 [1] Enable = The device sends the initial INVITE message without an
SDP.
[DisableCryptoLifeTimeI Enables the device to send "a=crypto" lines without the lifetime
nSDP] parameter in the SDP. For example, if the SDP contains "a=crypto:12
AES_CM_128_HMAC_SHA1_80
inline:hhQe10yZRcRcpIFPkH5xYY9R1de37ogh9G1MpvNp|2^31", it
removes the lifetime parameter "2^31".
 [1] Enable
Web/EMS: Enable Determines whether the device sets the Contact header of outgoing
Contact Restriction INVITE requests to ‘anonymous’ for restricted calls.
[EnableContactRestricti  [0] Disable (default)
on]  [1] Enable
Determines whether the device's IP address is used as the URI host
[AnonymousMode] part instead of "anonymous.invalid" in the INVITE's From header for
Tel-to-IP calls.
 [0] = (Default) If the device receives a call from the Tel with blocked
caller ID, it sends an INVITE with

From: “anonymous”<[email protected]>
 [1] = The device's IP address is used as the URI host part instead of
"anonymous.invalid".
This parameter may be useful, for example, for service providers who
identify their SIP Trunking customers by their source phone number or
IP address, reflected in the From header of the SIP INVITE. Therefore,
even customers blocking their Caller ID can be identified by the service
provider. Typically, if the device receives a call with blocked Caller ID
from the PSTN side (e.g., Trunk connected to a PBX), it sends an
INVITE to the IP with a From header as follows: From: “anonymous”
<[email protected]>. This is in accordance with RFC
3325. However, when this parameter is set to 1, the device replaces the
"anonymous.invalid" with its IP address.
EMS: P Asserted User Defines a 'representative number' (up to 50 characters) that is used as
Name the user part of the Request-URI in the P-Asserted-Identity header of an
[PAssertedUserName] outgoing INVITE for Tel-to-IP calls.
The default is null.
EMS: Use URL In Refer Defines the source for the SIP URI set in the Refer-To header of
To Header outgoing REFER messages.
[UseAORInReferToHead  [0] = (Default) Use SIP URI from Contact header of the initial call.
er]  [1] = Use SIP URI from To/From header of the initial call.
Web: Enable User- Enables the usage of the User Information, which is loaded to the
Information Usage device in the User Information auxiliary file. For a description on User
[EnableUserInfoUsage] Information, see 'Loading Auxiliary Files' on page 369.
 [1] Enable
[HandleReasonHeader] Determines whether the device uses the value of the incoming SIP
Reason header for Release Reason mapping.
 [0] = Disregard Reason header in incoming SIP messages.
 [1] = (Default) Use the Reason header value for Release Reason
mapping.
[EnableSilenceSuppInS Determines the device's behavior upon receipt of SIP Re-INVITE
DP] messages that include the SDP's 'silencesupp:off' attribute.
 [0] = (Default) Disregard the 'silecesupp' attribute.
 [1] = Handle incoming Re-INVITE messages that include the
'silencesupp:off' attribute in the SDP as a request to switch to the
Voice-Band-Data (VBD) mode. In addition, the device includes the
attribute 'a=silencesupp:off' in its SDP offer.
Note: This parameter is applicable only if the G.711 coder is used.
[EnableRport] Enables the usage of the 'rport' parameter in the Via header.
 [1] = Enabled
The device adds an 'rport' parameter to the Via header of each outgoing
SIP message. The first Proxy that receives this message sets the 'rport'
value of the response to the actual port from where the request was
received. This method is used, for example, to enable the device to
identify its port mapping outside a NAT.
If the Via header doesn't include the 'rport' parameter, the destination
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port of the response is obtained from the host part of the Via header.
If the Via header includes the 'rport' parameter without a port value, the
destination port of the response is the source port of the incoming
request.
If the Via header includes 'rport' with a port value (e.g., rport=1001), the
destination port of the response is the port indicated in the 'rport'
parmeter.
Determines whether the SIP X-Channel header is added to SIP
EMS: X Channel Header messages for providing information on the physical channel on which
[XChannelHeader] the call is received or placed.
 [0] Disable = (Default) X-Channel header is not used.
 [1] Enable = X-Channel header is generated by the device and sent
in INVITE messages and 180, 183, and 200 OK SIP responses. The
header includes the channel, and the device's IP address.
For example, 'x-channel: DS/DS1-1/8;IP=192.168.13.1', where:
 'DS/DS-1' is a constant string
 '1' is a constant string
 '8' is the channel (port)
 'IP=192.168.13.1' is the device's IP address
Web/EMS: Progress For Analog (FXS/FXO) interfaces:
Indicator to IP  [-1] Not Configured = (Default) Default values are used. The default
[ProgressIndicator2IP] for FXO interfaces is 1; The default for FXS interfaces is 0.
 [0] No PI = For IP-to-Tel calls, the device sends a 180 Ringing
response to IP after placing a call to a phone (FXS) or PBX (FXO).
 [1] PI = 1, [8] PI = 8: For IP-to-Tel calls, if the parameter
EnableEarlyMedia is set to 1, the device sends a 183 Session
Progress message with SDP immediately after a call is placed to a
phone/PBX. This is used to cut-through the voice path before the
remote party answers the call. This allows the originating party to
listen to network Call Progress Tones (such as ringback tone or
other network announcements).
Note: This parameter can also be configured per IP Profile (using the
IPProfile parameter) and Tel Profile (using the TelProfile parameter).
[EnableRekeyAfter181] Enables the device to send a re-INVITE with a new (different) SRTP key
(in the SDP) if a SIP 181 response is received ("call is being
forwarded"). The re-INVITE is sent immediately upon receipt of the 200
OK (when the call is answered).
 [1] = Enable
Note: This parameter is applicable only if SRTP is used.
[NumberOfActiveDialog Defines the maximum number of concurrent, outgoing SIP REGISTER
s] dialogs. This parameter is used to control the registration rate.
Notes:
 Once a 200 OK is received in response to a REGISTER message,
the REGISTER message is not considered in this maximum count
limit.
 This parameter applies only to outgoing REGISTER messages (i.e.,
incoming is unlimited).
Web/EMS: Default Defines the default Release Cause (sent to IP) for IP-to-Tel calls when
Release Cause the device initiates a call release and an explicit matching cause for this

[DefaultReleaseCause] release is not found.
The default release cause is NO_ROUTE_TO_DESTINATION (3).
Other common values include NO_CIRCUIT_AVAILABLE (34),
DESTINATION_OUT_OF_ORDER (27), etc.
Notes:
 The default release cause is described in the Q.931 notation and is
translated to corresponding SIP 40x or 50x values (e.g., 3 to SIP
404, and 34 to SIP 503).
 For information on mapping PSTN release causes to SIP responses,
see Mapping PSTN Release Cause to SIP Response.
 For a list of SIP responses-Q.931 release cause mapping, see
'Alternative Routing to Trunk upon Q.931 Call Release Cause Code'
on page 271.
Web: Enable Microsoft Enables the modification of the called and calling number for numbers
Extension received with Microsoft's proprietary "ext=xxx" parameter in the SIP
[EnableMicrosoftExt] INVITE URI user part. Microsoft Office Communications Server
sometimes uses this proprietary parameter to indicate the extension
number of the called or calling party.
 [1] Enable
For example, if a calling party makes a call to telephone number
622125519100 Ext. 104, the device receives the SIP INVITE (from
Microsoft's application) with the URI user part as INVITE
sip:622125519100;[email protected] (or INVITE
tel:622125519100;ext=104). If the parameter EnableMicrosofExt is
enabled, the device modifies the called number by adding an "e" as the
prefix, removing the "ext=" parameter, and adding the extension
number as the suffix (e.g., e622125519100104). Once modified, the
device can then manipulate the number further, using the Number
Manipulation tables to leave only the last 3 digits (for example) for
sending to a PBX.
EMS: Use SIP URI For Defines the URI format in the SIP Diversion header.
Diversion Header  [0] = 'tel:' (default)
[UseSIPURIForDiversio  [1] = 'sip:'
nHeader]
[TimeoutBetween100An Defines the timeout (in msec) between receiving a 100 Trying response
d18x] and a subsequent 18x response. If a 18x response is not received
within this timeout period, the call is disconnected.
The valid range is 0 to 180,000 (i.e., 3 minutes). The default is 32000
(i.e., 32 sec).
[IgnoreRemoteSDPMKI] Determines whether the device ignores the Master Key Identifier (MKI)
if present in the SDP received from the remote side.
 [1] Enable
Web: Comfort Noise Enables negotiation and usage of Comfort Noise (CN).
Generation Negotiation  [0] Disable
EMS: Comfort Noise  [1] Enable (default)
Generation
[ComfortNoiseNegotiati The use of CN is indicated by including a payload type for CN on the
on] media description line of the SDP. The device can use CN with a codec
whose RTP time stamp clock rate is 8,000 Hz (G.711/G.726). The static
payload type 13 is used. The use of CN is negotiated between sides.
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Therefore, if the remote side doesn't support CN, it is not used.
Regardless of the device's settings, it always attempts to adapt to the
remote SIP UA's request for CNG, as described below.
To determine CNG support, the device uses the
ComfortNoiseNegotiation parameter and the codec’s SCE (silence
suppression setting) using the CodersGroup parameter.
If the ComfortNoiseNegotiation parameter is enabled, then the following
occurs:
 If the device is the initiator, it sends a “CN” in the SDP only if the
SCE of the codec is enabled. If the remote UA responds with a “CN”
in the SDP, then CNG occurs; otherwise, CNG does not occur.
 If the device is the receiver and the remote SIP UA does not send a
“CN” in the SDP, then no CNG occurs. If the remote side sends a
“CN”, the device attempts to be compatible with the remote side and
even if the codec’s SCE is disabled, CNG occurs.
If the ComfortNoiseNegotiation parameter is disabled, then the device
does not send “CN” in the SDP. However, if the codec’s SCE is
enabled, then CNG occurs.
Defines the echo canceller format in the outgoing SDP. The 'ecan'
[SDPEcanFormat] attribute is used in the SDP to indicate the use of echo cancellation.
 [0] = (Default) The 'ecan' attribute appears on the 'a=gpmd' line.
 [1] = The 'ecan' attribute appears as a separate attribute.
 [2] = The 'ecan' attribute is not included in the SDP.
 [3] = The 'ecan' attribute and the 'vbd' parameter are not included in
the SDP.
Note: This parameter is applicable only when the IsFaxUsed parameter
is set to 2, and for re-INVITE messages generated by the device as
result of modem or fax tone detection.
Web/EMS: First Call Defines the index of the first ringback tone in the CPT file. This option
Ringback Tone ID enables an Application server to request the device to play a distinctive
[FirstCallRBTId] ringback tone to the calling party according to the destination of the call.
The tone is played according to the Alert-Info header received in the
180 Ringing SIP response (the value of the Alert-Info header is added
to the value of this parameter).
The valid range is -1 to 1,000. The default is -1 (i.e., play standard
ringback tone).
Notes:
 It is assumed that all ringback tones are defined in sequence in the
CPT file.
 In case of an MLPP call, the device uses the value of this parameter
plus 1 as the index of the ringback tone in the CPT file (e.g., if this
value is set to 1, then the index is 2, i.e., 1 + 1).
Web: Reanswer Time Defines the time interval from when the user hangs up the phone until
EMS: Regret Time the call is disconnected (FXS). This allows the user to hang up and then
[RegretTime] pick up the phone (before this timeout) to continue the call conversation.
Thus, it's also referred to as regret time.
The valid range is 0 to 255 (in seconds). The default is 0.
Web: Enable Enables the device to send a SIP INFO message with the On-Hook/Off-
Reanswering Info Hook parameter when the FXS phone goes on-hook during an ongoing
[EnableReansweringINF call and then off-hook again, within the user-defined regret timeout
O] (configured by the parameter RegretTime). Therefore, the device

notifies the far-end that the call has been re-answered.
 [1] Enable
This parameter is typically implemented for incoming IP-to-Tel collect
calls to the FXS port. If the FXS user does not wish to accept the collect
call, the user disconnects the call by on-hooking the phone. The device
notifies the softswitch (or Application server) of the unanswered collect
call (on-hook) by sending a SIP INFO message. As a result, the
softswitch disconnects the call (sends a BYE message to the device). If
the call is a regular incoming call and the FXS user on-hooks the phone
without intending to disconnect the call, the softswitch does not
disconnect the call (during the regret time).
The INFO message format is as follows:
INFO sip:[email protected]:5082 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bK_05_905924040-90579
From:
<sip:[email protected]:5080;user=phone>;tag=008
277765
To: <sip:[email protected]>;tag=svw-0-1229428367
Call-ID: ConorCCR-0-LU-1229417827103300@dtas-
stdn.fs5000group0-000.l
CSeq: 1 INFO
Contact: sip:10.20.7.70:5060
Content-Type: application/On-Hook (application/Off-Hook)
Content-Length: 0
Notes:
 This parameter is applicable only if the parameter RegretTime is
configured.
Web: PSTN Alert Timeout Defines the Alert Timeout (in seconds) for calls to the Tel side. This
EMS: Trunk PSTN Alert timer is used between the time a ring is generated (FXS) or a line is
Timeout seized (FXO), until the call is connected. For example: If the FXS device
[PSTNAlertTimeout] receives an INVITE, it generates a ring to the phone and sends a SIP
180 Ringing response to the IP. If the phone is not answered within the
time interval set by this parameter, the device cancels the call by
sending a SIP 408 response.
The valid value range is 1 to 600 (in seconds). The default is 180.
Web/EMS: RTP Only Enables the device to send and receive RTP packets to and from
Mode remote endpoints without the need to establish a SIP session. The
[RTPOnlyMode] remote IP address is determined according to the Outbound IP Routing
table (Prefix parameter). The port is the same port as the local RTP port
(configured by the BaseUDPPort parameter and the channel on which
the call is received).
 [1] Transmit & Receive = Send and receive RTP packets.
 [2] Transmit Only= Send RTP packets only.
 [3] Receive Only= Receive RTP packets only.
Notes:
 To configure the RTP Only mode per trunk, use the
RTPOnlyModeForTrunk_ID parameter.
 If per trunk configuration (using the RTPOnlyModeForTrunk_ID
parameter) is set to a value other than the default, the
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RTPOnlyMode parameter value is ignored.
Web/EMS: SIT Q850 Defines the Q.850 cause value specified in the SIP Reason header that
Cause is included in a 4xx response when a Special Information Tone (SIT) is
[SITQ850Cause] detected on an IP-to-Tel call.
Notes:
 For mapping specific SIT tones, you can use the
SITQ850CauseForNC, SITQ850CauseForIC, SITQ850CauseForVC,
and SITQ850CauseForRO parameters.
 This parameter is applicable only to FXO interfaces.
Cause For NC is included in a 4xx response when SIT-NC (No Circuit Found Special
[SITQ850CauseForNC] Information Tone) is detected from the TelPSTN for IP-to-Tel calls.
Notes:
 When not configured (i.e., default), the SITQ850Cause parameter is
used.
Cause For IC is included in a 4xx response when SIT-IC (Operator Intercept Special
[SITQ850CauseForIC] Information Tone) is detected from the Tel for IP-to-Tel calls.
The valid range is 0 to 127. The default is -1 (not configured).
Notes:
used.
Cause For VC is included in a 4xx response when SIT-VC (Vacant Circuit - non-
[SITQ850CauseForVC] registered number Special Information Tone) is detected from the Tel
Notes:
used.
Cause For RO is included in a 4xx response when SIT-RO (Reorder - System Busy
[SITQ850CauseForRO] Special Information Tone) is detected from the Tel for IP-to-Tel calls.
Notes:
used.
Out-of-Service (Busy Out) Parameters
Web/EMS: Enable Busy Enables the Busy Out feature.
Out  [0] Disable (Default)
[EnableBusyOut]  [1] Enable
When Busy Out is enabled and certain scenarios exist, the device does

the following:
The FXS port behaves according to the settings of the
FXSOOSBehavior parameter such as playing a reorder tone when the
phone is off-hooked, or changing the line polarity.
These behaviors are done upon one of the following scenarios:
 The device is physically disconnected from the network (i.e.,
Ethernet cable is disconnected).
 The Ethernet cable is connected, but the device is unable to
communicate with any host. For this scenario, the LAN Watch-Dog
must be activated (i.e., set the EnableLANWatchDog parameter to
1).
 The device can't communicate with the proxy (according to the Proxy
Keep-Alive mechanism) and no other alternative route exists to send
the call.
 The IP Connectivity mechanism is enabled (using the
AltRoutingTel2IPEnable parameter) and there is no connectivity to
any destination IP address.
Notes:
 The FXSOOSBehavior parameter determines the behavior of the
FXS endpoints when a Busy Out or Graceful Lock occurs.
 FXO endpoints during Busy Out and Lock are inactive.
 See the LifeLineType parameter for complementary optional
behavior.
Web: Out-Of-Service Determines the behavior of FXS endpoints when a Busy Out condition
Behavior exists.
EMS:FXS OOS Behavior  [0] None = Silence is heard when the FXS endpoint goes off-hook.
[FXSOOSBehavior]  [1] Reorder Tone = (Default) The device plays a reorder tone to the
connected phone / PBX.
 [2] Polarity Reversal = The device reverses the polarity of the
endpoint making it unusable (relevant, for example, for PBX DID
lines).
 [3] Reorder Tone + Polarity Reversal = Same as options [1] and [2].
 [4] Current Disconnect = The device disconnects the current to the
FXS endpoint.
Notes:
 A device reset is required for this parameter to take effect when it is
set to [2], [3], or [4].
Retransmission Parameters
Web: SIP T1 Defines the time interval (in msec) between the first transmission of a
Retransmission Timer SIP message and the first retransmission of the same message.
[msec] The default is 500.
EMS: T1 RTX
Note: The time interval between subsequent retransmissions of the
[SipT1Rtx]
same SIP message starts with SipT1Rtx. For INVITE requests, it is
multiplied by two for each new retransmitted message. For all other SIP
messages, it is multiplied by two until SipT2Rtx. For example, assuming
SipT1Rtx = 500 and SipT2Rtx = 4000:
 The first retransmission is sent after 500 msec.
 The second retransmission is sent after 1000 (2*500) msec.
 The third retransmission is sent after 2000 (2*1000) msec.
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 The fourth retransmission and subsequent retransmissions until
SIPMaxRtx are sent after 4000 (2*2000) msec.
Web: SIP T2 Defines the maximum interval (in msec) between retransmissions of SIP
Retransmission Timer messages (except for INVITE requests).
[msec] The default is 4000.
EMS: T2 RTX
Note: The time interval between subsequent retransmissions of the
[SipT2Rtx]
same SIP message starts with SipT1Rtx and is multiplied by two until
SipT2Rtx.
Web: SIP Maximum RTX Defines the maximum number of UDP transmissions of SIP messages
EMS: Max RTX (first transmission plus retransmissions).
[SIPMaxRtx] The range is 1 to 30. The default is 7.
Web: Number of RTX Defines the number of retransmitted INVITE/REGISTER messages
Before Hot-Swap before the call is routed (hot swap) to another Proxy/Registrar.
EMS: Proxy Hot Swap The valid range is 1 to 30. The default is 3.
Rtx
Note: This parameter is also used for alternative routing. If a domain
[HotSwapRtx]
name in the Tel to IP Routing is resolved into two IP addresses, and if
there is no response for HotSwapRtx retransmissions to the INVITE
message that is sent to the first IP address, the device immediately
initiates a call to the second IP address.
44.9 Coders and Profile Parameters

The profile parameters are described in the table below.
Table 44-33: Profile Parameters
Coders Table / Coder Groups Table

Web: Coders This table parameter defines the device's coders. Each group can consist of
Table/Coder Group up to 10 coders. The first Coder Group is the default coder list and the default
Settings Coder Group.
EMS: Coders Group The format of this parameter is as follows:
[CodersGroup0]
[ CodersGroup<0-9> ]
[CodersGroup1]
FORMAT CodersGroup0_Index = CodersGroup0_Name,
[CodersGroup2]
CodersGroup0_pTime, CodersGroup0_rate, CodersGroup0_PayloadType,
[CodersGroup3]
CodersGroup0_Sce;
[CodersGroup4]
[ \CodersGroup<0-9 ]
[CodersGroup5]
[CodersGroup6] For example, below are defined two Coder Groups (0 and 1):
[CodersGroup7] [ CodersGroup0 ]
[CodersGroup8] FORMAT CodersGroup0_Index = CodersGroup0_Name,
[CodersGroup9] CodersGroup0_pTime, CodersGroup0_rate,
CodersGroup0_PayloadType, CodersGroup0_Sce;
CodersGroup0 0 = g711Alaw64k, 20, 0, 255, 0;
[ \CodersGroup0 ]
[ CodersGroup1 ]
FORMAT CodersGroup1_Index = CodersGroup1_Name,
CodersGroup1_pTime, CodersGroup1_rate,
CodersGroup1_PayloadType, CodersGroup1_Sce;
CodersGroup1 0 = Transparent, 20, 0, 56, 0;

CodersGroup1 1 = g726, 20, 0, 23, 0;
[ \CodersGroup1 ]
Notes:
 For a list of supported coders and a detailed description of this table, see
Configuring Coders on page 219.
 The coder name is case-sensitive.
IP Profile Table
Web: IP Profile This table parameter configures the IP Profile table. Each IP Profile ID
Settings includes a set of parameters (which are typically configured separately using
EMS: Protocol their individual "global" parameters). You can later assign these IP Profiles to
Definition > IP Profile Tel-to-IP routing rules (Prefix parameter), IP-to-Tel routing rules and IP
[IPProfile] Groups.
[IPProfile]
FORMAT IpProfile_Index = IpProfile_ProfileName, IpProfile_IpPreference,
IpProfile_CodersGroupID, IpProfile_IsFaxUsed, IpProfile_JitterBufMinDelay,
IpProfile_JitterBufOptFactor, IpProfile_IPDiffServ, IpProfile_SigIPDiffServ,
IpProfile_SCE, IpProfile_RTPRedundancyDepth,
IpProfile_RemoteBaseUDPPort, IpProfile_CNGmode,
IpProfile_VxxTransportType, IpProfile_NSEMode, IpProfile_IsDTMFUsed,
IpProfile_PlayRBTone2IP, IpProfile_EnableEarlyMedia,
IpProfile_ProgressIndicator2IP, IpProfile_EnableEchoCanceller,
IpProfile_CopyDest2RedirectNumber, IpProfile_MediaSecurityBehaviour,
IpProfile_CallLimit, IpProfile_DisconnectOnBrokenConnection,
IpProfile_FirstTxDtmfOption, IpProfile_SecondTxDtmfOption,
IpProfile_RxDTMFOption, IpProfile_EnableHold, IpProfile_InputGain,
IpProfile_VoiceVolume, IpProfile_AddIEInSetup,
IpProfile_SBCExtensionCodersGroupID,
IpProfile_MediaIPVersionPreference, IpProfile_TranscodingMode,
IpProfile_SBCAllowedCodersGroupID, IpProfile_SBCAllowedCodersMode,
IpProfile_SBCMediaSecurityBehaviour, IpProfile_SBCRFC2833Behavior,
IpProfile_SBCAlternativeDTMFMethod, IpProfile_SBCAssertIdentity,
IpProfile_AMDSensitivityParameterSuit, IpProfile_AMDSensitivityLevel,
IpProfile_AMDMaxGreetingTime,
IpProfile_AMDMaxPostSilenceGreetingTime, IpProfile_SBCDiversionMode,
IpProfile_SBCHistoryInfoMode, IpProfile_EnableQSIGTunneling,
IpProfile_SBCFaxCodersGroupID, IpProfile_SBCFaxBehavior,
IpProfile_SBCFaxOfferMode, IpProfile_SBCFaxAnswerMode,
IpProfile_SbcPrackMode, IpProfile_SBCSessionExpiresMode,
IpProfile_SBCRemoteUpdateSupport, IpProfile_SBCRemoteReinviteSupport,
IpProfile_SBCRemoteDelayedOfferSupport,
IpProfile_SBCRemoteReferBehavior, IpProfile_SBCRemote3xxBehavior,
IpProfile_SBCRemoteMultiple18xSupport,
IpProfile_SBCRemoteEarlyMediaResponseType,
IpProfile_SBCRemoteEarlyMediaSupport, IpProfile_EnableSymmetricMKI,
IpProfile_MKISize, IpProfile_SBCEnforceMKISize,
IpProfile_SBCRemoteEarlyMediaRTP,
IpProfile_SBCRemoteSupportsRFC3960,
IpProfile_SBCRemoteCanPlayRingback, IpProfile_EnableEarly183,
IpProfile_EarlyAnswerTimeout, IpProfile_SBC2833DTMFPayloadType,
IpProfile_SBCUserRegistrationTime, IpProfile_ResetSRTPStateUponRekey,
IpProfile_AmdMode, IpProfile_SBCReliableHeldToneSource,
IpProfile_SBCPlayHeldTone, IpProfile_SBCRemoteHoldFormat;
[\IPProfile]
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Note: For a description of this table, see 'Configuring IP Profiles' on page
225.
Tel Profile Table
Web: Tel Profile This table parameter configures the Tel Profile table. Each Tel Profile ID
Settings includes a set of parameters (which are typically configured separately using
EMS: Protocol their individual, "global" parameters). You can later assign these Tel Profile
Definition > IDs to other elements such as in the <trunkgrouptableMP>Endpoint Phone
Telephony Profile Number table (TrunkGroup parameter). Therefore, Tel Profiles allow you to
[TelProfile] apply the same settings of a group of parameters to multiple channels, or
apply specific settings to different channels.
[TelProfile]
FORMAT TelProfile_Index = TelProfile_ProfileName,
TelProfile_TelPreference, TelProfile_CodersGroupID, TelProfile_IsFaxUsed,
TelProfile_JitterBufMinDelay, TelProfile_JitterBufOptFactor,
TelProfile_IPDiffServ, TelProfile_SigIPDiffServ, TelProfile_DtmfVolume,
TelProfile_InputGain, TelProfile_VoiceVolume,
TelProfile_EnableReversePolarity, TelProfile_EnableCurrentDisconnect,
TelProfile_EnableDigitDelivery, TelProfile_EnableEC, TelProfile_MWIAnalog,
TelProfile_MWIDisplay, TelProfile_FlashHookPeriod,
TelProfile_EnableEarlyMedia, TelProfile_ProgressIndicator2IP,
TelProfile_TimeForReorderTone, TelProfile_EnableDIDWink,
TelProfile_IsTwoStageDial, TelProfile_DisconnectOnBusyTone,
TelProfile_EnableVoiceMailDelay, TelProfile_DialPlanIndex,
TelProfile_Enable911PSAP, TelProfile_SwapTelToIpPhoneNumbers,
TelProfile_EnableAGC, TelProfile_ECNlpMode,
TelProfile_DigitalCutThrough, TelProfile_EnableFXODoubleAnswer,
TelProfile_CallPriorityMode;
[\TelProfile]
Notes:
 For a description of this parameter, see Configuring Tel Profiles on page
223.
 For a detailed description of each parameter, see its corresponding
"global" parameter.
TelProfile Field Web Name Global Parameter

TelProfile_ProfileName Profile -
Name
TelProfile_TelPreference Profile -
Preference
TelProfile_CodersGroupID Coder CodersGroup0
Group
TelProfile_IsFaxUsed Fax IsFaxUsed
Signaling
Method
TelProfile_JitterBufMinDelay Dynamic DJBufMinDelay
Jitter Buffer
Minimum
Delay
TelProfile_JitterBufOptFactor Dynamic DJBufOptFactor

Jitter Buffer
Optimization
Factor

TelProfile_IPDiffServ RTP IP PremiumServiceClassMediaDiffServ

DiffServ
TelProfile_SigIPDiffServ Signaling PremiumServiceClassControlDiffServ

DiffServ
TelProfile_DtmfVolume DTMF DTMFVolume
Volume
TelProfile_InputGain Input Gain InputGain

TelProfile_VoiceVolume Voice VoiceVolume
Volume
TelProfile_EnableReversePolarity Enable EnableReversalPolarity

Polarity
Reversal
TelProfile_EnableCurrentDisconnect Enable EnableCurrentDisconnect

Current
Disconnect
TelProfile_EnableDigitDelivery Enable Digit EnableDigitDelivery

Delivery
TelProfile_EnableEC Echo EnableEchoCanceller
Canceler
TelProfile_MWIAnalog MWI Analog MWIAnalogLamp
Lamp
TelProfile_MWIDisplay MWI MWIDisplay
Display
TelProfile_FlashHookPeriod Flash Hook FlashHookPeriod
Period
TelProfile_EnableEarlyMedia Enable EnableEarlyMedia

Early Media
TelProfile_ProgressIndicator2IP Progress ProgressIndicator2IP
Indicator to
IP
TelProfile_TimeForReorderTone Time For TimeForReorderTone
Reorder
Tone
TelProfile_EnableDIDWink Enable DID EnableDIDWink

Wink
TelProfile_IsTwoStageDial Dialing IsTwoStageDial
Mode
TelProfile_DisconnectOnBusyTone Disconnect DisconnectOnBusyTone

Call on
Detection of
Busy Tone
TelProfile_EnableVoiceMailDelay Enable -
Voice Mail
Delay
TelProfile_DialPlanIndex Dial Plan DialPlanIndex
Index
TelProfile_Enable911PSAP Enable 911 Enable911PSAP

PSAP
TelProfile_SwapTelToIpPhoneNumbers Swap Tel SwapTEl2IPCalled&CallingNumbers

To IP Phone
Numbers
TelProfile_EnableAGC Enable AGC EnableAGC
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TelProfile_ECNlpMode EC NLP ECNLPMode

Mode
TelProfile_DigitalCutThrough - DigitalCutThrough
TelProfile_EnableFXODoubleAnswer - EnableFXODoubleAnswer
TelProfile_CallPriorityMode - CallPriorityMode
The parameter IpPreference parameter EnableVo.
44.10 Channel Parameters

This subsection describes the device's channel parameters.
44.10.1 Voice Parameters

The voice parameters are described in the table below.
Table 44-34: Voice Parameters
Web/EMS: Input Gain Defines the pulse-code modulation (PCM) input gain control (in
[InputGain] decibels). This parameter sets the level for the received (Tel-to-
IP) signal.
The valid range is -32 to 31 dB. The default is 0 dB.
Note: This parameter can also be configured in an IP Profile
and/or a Tel Profile.
Web: Voice Volume Defines the voice gain control (in decibels). This parameter sets
EMS: Volume (dB) the level for the transmitted (IP-to-Tel) signal.
[VoiceVolume] The valid range is -32 to 31 dB. The default is 0 dB.
EMS: Payload Format Determines the bit ordering of the G.726/G.727 voice payload
[VoicePayloadFormat] format.
 [0] = (Default) Little Endian
 [1] = Big Endian
Notes:
 To ensure high voice quality when using G.726/G.727, both
communicating ends should use the same endianness format.
Therefore, when the device communicates with a third-party
entity that uses the G.726/G.727 voice coder and voice quality
is poor, change the settings of this parameter (between Big
Endian and Little Endian).
 The G.727 coder is currently not supported by MP-124 Rev. E.
Web: MF Transport Type Currently, not supported.
[MFTransportType]
Web: Enable Answer Detector Currently, not supported.
[EnableAnswerDetector]
Web: Answer Detector Activity Defines the time (in 100-msec resolution) between activating the
Delay Answer Detector and the time that the detector actually starts to

[AnswerDetectorActivityDelay] operate.
Note: AD is currently not supported by MP-124 Rev. E.
Web: Answer Detector Silence Currently, not supported.
Time
[AnswerDetectorSilenceTime]
Web: Answer Detector Currently, not supported.
Redirection
[AnswerDetectorRedirection]
Web: Answer Detector Sensitivity Defines the Answer Detector sensitivity.
EMS: Sensitivity The range is 0 (most sensitive) to 2 (least sensitive). The default
[AnswerDetectorSensitivity] is 0.
Note: AD is currently not supported by MP-124 Rev. E.
Web: Echo Canceler Enables echo cancellation (i.e., echo from voice calls is
EMS: Echo Canceller Enable removed).
[EnableEchoCanceller]  [0] Disable
EMS: Echo Canceller Hybrid Defines the four-wire to two-wire worst-case Hybrid loss, the ratio
Loss between the signal level sent to the hybrid and the echo level
[ECHybridLoss] returning from the hybrid.
 [0] = (Default) 6 dB
 [1] = N/A
 [2] = 0 dB
 [3] = 3 dB
EMS: ECN lp Mode Defines the echo cancellation Non-Linear Processing (NLP)
[ECNLPMode] mode.
 [0] = (Default) NLP adapts according to echo changes
 [1] = Disables NLP
 [2] = Silence output NLP
Note: This parameter can also be configured in a Tel Profile.
Enables the Aggressive NLP at the first 0.5 second of the call.
[EchoCancellerAggressiveNLP]  [0] = Disable
 [1] = (Default) Enable. The echo is removed only in the first
half of a second of the incoming IP signal.
Defines the number of spectral coefficients added to an SID
[RTPSIDCoeffNum] packet being sent according to RFC 3389.
The valid values are [0] (default), [4], [6], [8] and [10].
44.10.2 Coder Parameters

The coder parameters are described in the table below.
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MP-11x & MP-124
Table 44-35: Coder Parameters
EMS: AMR Coder Header Determines the payload format of the AMR header.
Format  [0] = Non-standard multiple frames packing in a single RTP
[AMRCoderHeaderFormat] frame. Each frame has a CMR and TOC header.
 [1] = AMR frame according to RFC 3267 bundling.
 [2] = AMR frame according to RFC 3267 interleaving.
 [3] = AMR is passed using the AMR IF2 format.
Note: Bandwidth Efficient mode is not supported; the mode is
always Octet-aligned.
Web: DSP Version Template Determines the DSP template used by the device. Each DSP
Number template supports specific coders, channel capacity, and features.
EMS: Version Template Number The default is DSP Template 0.
[DSPVersionTemplateNumber]
Notes:
 For a list of supported DSP templates, see DSP Templates on
page 649.
44.10.3 DTMF Parameters

The dual-tone multi-frequency (DTMF) parameters are described in the table below.
Table 44-36: DTMF Parameters
Web/EMS: DTMF Transport Determines the DTMF transport type.

Type  [0] Mute DTMF = DTMF digits are removed from the voice
[DTMFTransportType] stream and are not relayed to remote side.
 [2] Transparent DTMF = DTMF digits remain in the voice
stream.
 [3] RFC 2833 Relay DTMF = (Default) DTMF digits are removed
from the voice stream and are relayed to remote side according
to RFC 2833.
 [7] RFC 2833 Relay Decoder Mute = DTMF digits are sent
according to RFC 2833 and muted when received.
Note: This parameter is automatically updated if the parameters
TxDTMFOption or RxDTMFOption are configured.
Web: DTMF Volume (-31 to 0 Defines the DTMF gain control value (in decibels) to the or analog
dB) side.
EMS: DTMF Volume (dBm) The valid range is -31 to 0 dB. The default is -11 dB.
[DTMFVolume]
Web: DTMF Generation Twist Defines the range (in decibels) between the high and low frequency
EMS: DTMF Twist Control components in the DTMF signal. Positive decibel values cause the
[DTMFGenerationTwist] higher frequency component to be stronger than the lower one.
Negative values cause the opposite effect. For any parameter
value, both components change so that their average is constant.
The valid range is -10 to 10 dB. The default is 0 dB.
EMS: DTMF Inter Interval Defines the time (in msec) between generated DTMF digits to

(msec) PSTN side (if TxDTMFOption = 1, 2 or 3).
[DTMFInterDigitInterval] The default is 100 msec. The valid range is 0 to 32767.
EMS: DTMF Length (msec) Defines the time (in msec) for generating DTMF tones to the PSTN
[DTMFDigitLength] side (if TxDTMFOption = 1, 2 or 3). It also configures the duration
that is sent in INFO (Cisco) messages.
EMS: Rx DTMF Relay Hang Defines the Voice Silence time (in msec) after playing DTMF or MF
Over Time (msec) digits to the Tel/PSTN side that arrive as Relay from the IP side.
[RxDTMFHangOverTime] Valid range is 0 to 2,000 msec. The default is 1,000 msec.
EMS: Tx DTMF Relay Hang Defines the Voice Silence time (in msec) after detecting the end of
Over Time (msec) DTMF or MF digits at the Tel/PSTN side when the DTMF Transport
[TxDTMFHangOverTime] Type is either Relay or Mute.
Valid range is 0 to 2,000 msec. The default is 1,000 msec.
Web/EMS: NTE Max Duration Defines the maximum time for sending Named Telephony Events /
[NTEMaxDuration] NTEs (RFC 4733/2833 DTMF relay) to the IP side, regardless of
the DTMF signal duration on the TDM side.
The range is -1 to 200,000,000 msec. The default is -1 (i.e., NTE
stops only upon detection of an End event).
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44.10.4 RTP, RTCP and T.38 Parameters

The RTP, RTCP and T.38 parameters are described in the table below.
Table 44-37: RTP/RTCP and T.38 Parameters
Web: Dynamic Jitter Buffer Minimum Defines the minimum delay (in msec) for the Dynamic Jitter
Delay Buffer.
EMS: Minimal Delay (dB) The valid range is 0 to 150. The default delay is 10.
[DJBufMinDelay]
Notes:
 This parameter can also be configured in an IP Profile
 For more information on Jitter Buffer, see Dynamic Jitter
Buffer Operation on page 179.
Web: Dynamic Jitter Buffer Defines the Dynamic Jitter Buffer frame error/delay
Optimization Factor optimization factor.
EMS: Opt Factor The valid range is 0 to 12. The default factor is 10.
[DJBufOptFactor]
Notes:
 For data (fax and modem) calls, set this parameter to 12.
 This parameter can also be configured in an IP Profile
 For more information on Jitter Buffer, see Dynamic Jitter
Buffer Operation on page 179.
Web/EMS: Analog Signal Transport Determines the analog signal transport type.
Type  [0] Ignore Analog Signals = (Default) Ignore.
[AnalogSignalTransportType]  [1] RFC 2833 Analog Signal Relay = Transfer hookflash
using RFC 2833.
Web: RTP Redundancy Depth Enables the device to generate RFC 2198 redundant
EMS: Redundancy Depth packets. This can be used for packet loss where the missing
[RTPRedundancyDepth] information (audio) can be reconstructed at the receiver's
end from the redundant data that arrives in subsequent
packets. This is required, for example, in wireless networks
where a high percentage (up to 50%) of packet loss can be
experienced.
 [0] 0 = (Default) Disable.
 [1] 1 = Enable - previous voice payload packet is added
to current packet.
Notes:
 When enabled, you can configure the payload type,
using the RFC2198PayloadType parameter.
 The RTP redundancy dynamic payload type can be
included in the SDP, by using the
EnableRTPRedundancyNegotiation parameter.
 This parameter can also be configured in an IP Profile.

Web: Enable RTP Redundancy Enables the device to include the RTP redundancy dynamic
Negotiation payload type in the SDP, according to RFC 2198.
[EnableRTPRedundancyNegotiation]  [0] Disable (default)
 [1] Enable
When enabled, the device includes in the SDP message the
RTP payload type "RED" and the payload type configured
by the parameter RFC2198PayloadType.
a=rtpmap:<PT> RED/8000
Where <PT> is the payload type as defined by
RFC2198PayloadType. The device sends the INVITE
message with "a=rtpmap:<PT> RED/8000" and responds
with a 18x/200 OK and "a=rtpmap:<PT> RED/8000" in the
SDP.
Notes:
 For this feature to be functional, you must also set the
parameter RTPRedundancyDepth to 1 (i.e., enabled).
 Currently, the negotiation of “RED” payload type is not
supported and therefore, it should be configured to the
same PT value for both parties.
Web: RFC 2198 Payload Type Defines the RTP redundancy packet payload type according
EMS: Redundancy Payload Type to RFC 2198.
[RFC2198PayloadType] The range is 96 to 127. The default is 104.
RTPRedundancyDepth is set to 1.
Web: Packing Factor N/A. Controlled internally by the device according to the
EMS: Packetization Factor selected coder.
[RTPPackingFactor]
Web/EMS: Basic RTP Packet Interval N/A. Controlled internally by the device according to the
[BasicRTPPacketInterval] selected coder.
Web: RTP Directional Control N/A. Controlled internally by the device according to the
[RTPDirectionControl] selected coder.
Web/EMS: RFC 2833 TX Payload Defines the Tx RFC 2833 DTMF relay dynamic payload
Type type.
[RFC2833TxPayloadType] The valid range is 96 to 99, and 106 to 127. The default is
96. The 100, 102 to 105 range is allocated for proprietary
usage.
Notes:
 Certain vendors (e.g., Cisco) use payload type 101 for
RFC 2833.
 When RFC 2833 payload type negotiation is used (i.e.,
the parameter TxDTMFOption is set to 4), this payload
type is used for the received DTMF packets. If
negotiation isn't used, this payload type is used for
receive and for transmit.
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Web/EMS: RFC 2833 RX Payload Defines the Rx RFC 2833 DTMF relay dynamic payload
Type type.
[RFC2833RxPayloadType] The valid range is 96 to 99, and 106 to 127. The default is
96. The 100, 102 to 105 range is allocated for proprietary
usage.
Notes:
 Certain vendors (e.g., Cisco) use payload type 101 for
RFC 2833.
 When RFC 2833 payload type negotiation is used (i.e.,
the parameter TxDTMFOption is set to 4), this payload
type is used for the received DTMF packets. If
negotiation isn't used, this payload type is used for
receive and for transmit.
[EnableDetectRemoteMACChange] Determines whether the device changes the RTP packets
according to the MAC address of received RTP packets and
according to Gratuitous Address Resolution Protocol
(GARP) messages.
 [0] = Nothing is changed.
 [1] = If the device receives RTP packets with a different
source MAC address (than the MAC address of the
transmitted RTP packets), then it sends RTP packets to
this MAC address and removes this IP entry from the
device's ARP cache table.
 [2] = (Default) The device uses the received GARP
packets to change the MAC address of the transmitted
RTP packets.
 [3] = Options 1 and 2 are used.
Notes:
required.
 If the device is located in a network subnet which is
connected to other gateways using a router that uses
Virtual Router Redundancy Protocol (VRRP) for
redundancy, then set this parameter to 0 or 2.

Web: RTP Base UDP Port Defines the lower boundary of the UDP port used for RTP,
EMS: Base UDP Port RTCP (RTP port + 1) and T.38 (RTP port + 2). For example,
[BaseUDPport] if the Base UDP Port is set to 6000, then one channel may
use the ports RTP 6000, RTCP 6001, and T.38 6002, while
another channel may use RTP 6010, RTCP 6011, and T.38
6012, and so on.
The range of possible UDP ports is 6,000 to 64,000. The
default base UDP port is 6000.
Once this parameter is configured, the UDP port range
(lower to upper boundary) is calculated as follows:
 MP-112/MP-114: BaseUDPport to (BaseUDPport +
3*10)
 MP-118: BaseUDPport to (BaseUDPport + 7*10)
 MP-124: BaseUDPport to (BaseUDPport + 23*10)
Notes:
required.
 You can define a UDP port range per Media Realm (see
Configuring Media Realms on page 188).
 The UDP ports are allocated randomly to channels.
 If RTP Base UDP Port is not a factor of 10, the following
message is generated: 'invalid local RTP port'.
EMS: No Op Enable Enables the transmission of RTP or T.38 No-Op packets.
[NoOpEnable]  [0] = Disable (default)
 [1] = Enable
This mechanism ensures that the NAT binding remains
open during RTP or T.38 silence periods.
EMS: No Op Interval Defines the time interval in which RTP or T.38 No-Op
[NoOpInterval] packets are sent in the case of silence (no RTP/T.38 traffic)
when No-Op packet transmission is enabled.
The valid range is 20 to 65,000 msec. The default is 10,000.
Note: To enable No-Op packet transmission, use the
NoOpEnable parameter.
EMS: No Op Payload Type Defines the payload type of No-Op packets.
[RTPNoOpPayloadType] The valid range is 96 to 127 (for the range of Dynamic RTP
Payload Type for all types of non hard-coded RTP Payload
types, refer to RFC 3551). The default is 120.
Note: When defining this parameter, ensure that it doesn't
cause collision with other payload types.
Disables RTCP traffic when there is no RTP traffic. This
[RTCPActivationMode] feature is useful, for example, to stop RTCP traffic that is
typically sent when calls are put on hold (by an INVITE with
'a=inactive' in the SDP).
 [0] Active Always = (Default) RTCP is active even during
inactive RTP periods, i.e., when the media is in 'recvonly'
or 'inactive' mode.
 [1] Inactive Only If RTP Inactive = No RTCP is sent
when RTP is inactive.
RTP Control Protocol Extended Reports (RTCP XR) Parameters
Version 6.6 563 MP-11x & MP-124

MP-11x & MP-124
Web: Enable RTCP XR Enables voice quality monitoring and RTCP XR, according
EMS: RTCP XR Enable to Internet-Draft draft-ietf-sipping-rtcp-summary-13.
[VQMonEnable]  [0] Disable (default)
 [1] Enable
required.
Web: Minimum Gap Size Defines the voice quality monitoring - minimum gap size
EMS: GMin (number of frames).
[VQMonGMin] The default is 16.
Web/EMS: Burst Threshold Defines the voice quality monitoring - excessive burst alert
[VQMonBurstHR] threshold.
The default is -1 (i.e., no alerts are issued).
Web/EMS: Delay Threshold Defines the voice quality monitoring - excessive delay alert
[VQMonDelayTHR] threshold.
The default is -1 (i.e., no alerts are issued).
Web: R-Value Delay Threshold Defines the voice quality monitoring - end of call low quality
EMS: End of Call Rval Delay alert threshold.
Threshold The default is -1 (i.e., no alerts are issued).
[VQMonEOCRValTHR]
Web: RTCP XR Packet Interval Defines the time interval (in msec) between adjacent RTCP
EMS: Packet Interval reports.
[RTCPInterval] The valid value range is 0 to 65,535. The default is 5,000.
Web: Disable RTCP XR Interval Determines whether RTCP report intervals are randomized
Randomization or whether each report interval accords exactly to the
EMS: Disable Interval Randomization parameter RTCPInterval.
[DisableRTCPRandomize]  [0] Disable = (Default) Randomize
 [1] Enable = No Randomize
EMS: RTCP XR Collection Server Defines the transport layer used for outgoing SIP dialogs
Transport Type initiated by the device to the RTCP XR Collection Server.
[RTCPXRESCTransportType]  [-1] Not Configured (default)
 [0] UDP
 [1] TCP
 [2] TLS
Note: When set to [-1], the value of the SIPTransportType
parameter is used.
Web: RTCP XR Collection Server Defines the IP address of the Event State Compositor
EMS: Esc IP (ESC). The device sends RTCP XR reports to this server,
[RTCPXREscIP] using SIP PUBLISH messages. The address can be
configured as a numerical IP address or as a domain name.
Note: Instead of sending RTCP XR to an ESC server, you
can send RTCP XR to an IP Group (see the
PublicationIPGroupID parameter).

Web: RTCP XR Report Mode Enables the device to send RTCP XR in SIP PUBLISH
EMS: Report Mode messages to the Event State Compositor (ESC) server and
[RTCPXRReportMode] defines the interval at which they are sent.
 [0] Disable = (Default) RTCP XR is not sent.
 [1] End Call = RTCP XR is sent at the end of the call.
 [2] End Call & Periodic = RTCP XR is sent at the end of
the call and periodically according to the RTCPInterval
parameter.
 [3] End Call & End Segment = RTCP XR is sent at the
end of the call and at the end of each media segment of
the call. A media segment is a change in media, for
example, when the coder is changed or when the caller
toggles between two called parties (using call
hold/retrieve). The RTCP XR sent at the end of a media
segment contains information only of that segment. If the
segment does not contain RTP/RTCP content, the RTCP
XR is not sent. For call hold, the device sends an RTCP
XR each time the call is placed on hold and each time it
is retrieved. In addition, the Start timestamp in the RTCP
XR indicates the start of the media segment; the End
timestamp indicates the time of the last sent periodic
RTCP XR (typically, up to 5 seconds before reported
segment ends).
publication-ip-group-id Defines the IP Group to where the RTCP XR is sent. If the
[PublicationIPGroupID] value is -1 (default) or 0, the RTCP XR is sent to the ESC
server, as configured by the RTCPXREscIP parameter.
Note: The parameter is applicable only to the Gateway
application.
Version 6.6 565 MP-11x & MP-124

MP-11x & MP-124
44.11 Gateway and IP-to-IP Parameters

44.11.1 Fax and Modem Parameters
The fax and modem parameters are described in the table below.
Table 44-38: Fax and Modem Parameters
Web: Fax Transport Mode Determines the fax transport mode used by the device.
EMS: Transport Mode  [0] Disable = transparent mode
[FaxTransportMode]  [1] T.38 Relay (default)
 [2] Bypass
 [3] Events Only
Note: This parameter is overridden by the parameter
IsFaxUsed. If the parameter IsFaxUsed is set to 1 (T.38 Relay)
or 3 (Fax Fallback), then FaxTransportMode is always set to 1
(T.38 relay).
Web: V.21 Modem Transport Type Determines the V.21 modem transport type.
EMS: V21 Transport  [0] Disable = (Default) Disable (Transparent)
[V21ModemTransportType]  [2] Enable Bypass
 [3] Events Only = Transparent with Events
Note: This parameter can also be configured in an IP Profile.
EMS: V22 Transport  [0] Disable = Disable (Transparent)
[V22ModemTransportType]  [2] Enable Bypass (default)
Notes:
 This parameter applies only to V.32 and V.32bis modems.
Web: V.34 Modem Transport Type Determines the V.90/V.34 modem transport type.

EMS: Bell Transport Type Determines the Bell modem transport method.
[BellModemTransportType]  [0] = Transparent (default)
 [2] = Bypass
 [3] = Transparent with events
Web/EMS: Fax CNG Mode Determines the device's handling of fax relay upon detection of
[FaxCNGMode] a fax CNG tone from originating faxes.
 [0] Doesn't send T.38 Re-INVITE = (Default) SIP re-INVITE
is not sent.
 [1] Sends on CNG tone = Sends a SIP re-INVITE with T.38
parameters in SDP to the terminating fax upon detection of a
fax CNG tone, if the CNGDetectorMode parameter is set to
1.
Notes:
 This feature is applicable only if the IsFaxUsed parameter is
set to [1] or [3].
 The device also sends T.38 re-INVITE if the
CNGDetectorMode parameter is set to [2], regardless of the
FaxCNGMode parameter settings.
Web/EMS: CNG Detector Mode Determines whether the device detects the fax calling tone
[CNGDetectorMode] (CNG).
 [0] Disable = (Default) The originating device doesn’t detect
CNG; the CNG signal passes transparently to the remote
side.
 [1] Relay = CNG is detected on the originating side. CNG
packets are sent to the remote side according to T.38 (if
IsFaxUsed = 1) and the fax session is started. A SIP Re-
INVITE message isn’t sent and the fax session starts by the
terminating device. This option is useful, for example, when
the originating device is located behind a firewall that blocks
incoming T.38 packets on ports that have not yet received
T.38 packets from the internal network (i.e., originating
device). To also send a Re-INVITE message upon detection
of a fax CNG tone in this mode, set the parameter
FaxCNGMode to 1.
 [2] Events Only = CNG is detected on the originating side
and a fax session is started by the originating side using the
Re-INVITE message. Usually, T.38 fax session starts when
the ‘preamble’ signal is detected by the answering side.
Some SIP devices don’t support the detection of this fax
signal on the answering side and thus, in these cases it is
possible to configure the device to start the T.38 fax session
when the CNG tone is detected by the originating side.
However, this mode is not recommended.
Web: Fax Relay Enhanced Defines the number of times that control packets are
Redundancy Depth retransmitted when using the T.38 standard.
EMS: Enhanced Relay The valid range is 0 to 4. The default is 2.
Redundancy Depth
[FaxRelayEnhancedRedundancy
Depth]
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Web: Fax Relay Redundancy Defines the number of times that each fax relay payload is
Depth retransmitted to the network.
EMS: Relay Redundancy Depth  [0] = (Default) No redundancy
[FaxRelayRedundancyDepth]  [1] = One packet redundancy
 [2] = Two packet redundancy
Note: This parameter is applicable only to non-V.21 packets.
Web: Fax Relay Max Rate (bps) Defines the maximum rate (in bps) at which fax relay messages
EMS: Relay Max Rate are transmitted (outgoing calls).
[FaxRelayMaxRate]  [0] 2400 = 2.4 kbps
 [1] 4800 = 4.8 kbps
 [2] 7200 = 7.2 kbps
 [3] 9600 = 9.6 kbps
 [4] 12000 = 12.0 kbps
 [5] 14400 = 14.4 kbps (default)
Note: The rate is negotiated between both sides (i.e., the device
adapts to the capabilities of the remote side). Negotiation of the
T.38 maximum supported fax data rate is provided in SIP’s SDP
T38MaxBitRate parameter. The negotiated T38MaxBitRate is
the minimum rate supported between the local and remote
endpoints.
Web: Fax Relay ECM Enable Enables Error Correction Mode (ECM) mode during fax relay.
EMS: Relay ECM Enable  [0] Disable
[FaxRelayECMEnable]  [1] Enable (default)
Web: Fax/Modem Bypass Coder Determines the coder used by the device when performing
Type fax/modem bypass. Typically, high-bit-rate coders such as
EMS: Coder Type G.711 should be used.
[FaxModemBypassCoderType]  [0] G.711Alaw= (Default) G.711 A-law 64
 [1] G.711Mulaw = G.711 µ-law
Web: Fax/Modem Bypass Packing Defines the number (20 msec) of coder payloads used to
Factor generate a fax/modem bypass packet.
EMS: Packetization Period The valid range is 1, 2, or 3 coder payloads. The default is 1
[FaxModemBypassM] coder payload.
Determines whether the device sends RFC 2833 ANS/ANSam
[FaxModemNTEMode] events upon detection of fax and/or modem Answer tones (i.e.,
CED tone).
 [1] = Enabled
Note: This parameter is applicable only when the fax or modem
transport type is set to bypass or Transparent-with-Events.
Web/EMS: Fax Bypass Payload Defines the fax bypass RTP dynamic payload type.
Type The valid range is 0 to 127. The default is 102.
[FaxBypassPayloadType]
EMS: Modem Bypass Payload Defines the modem bypass dynamic payload type.
Type The range is 0 to 127. The default is 103.
[ModemBypassPayloadType]

EMS: Relay Volume (dBm) Defines the fax gain control.

[FaxModemRelayVolume] The range is -18 to -3, corresponding to -18 dBm to -3 dBm in 1-
dB steps. The default is -6 dBm fax gain control.
Web/EMS: Fax Bypass Output Defines the fax bypass output gain control.
Gain The range is -31 to +31 dB, in 1-dB steps. The default is 0 (i.e.,
[FaxBypassOutputGain] no gain).
Web/EMS: Modem Bypass Output Defines the modem bypass output gain control.
Gain The range is -31 dB to +31 dB, in 1-dB steps. The default is 0
[ModemBypassOutputGain] (i.e., no gain).
EMS: Basic Packet Interval Defines the basic frame size used during fax/modem bypass
[FaxModemBypassBasicRTPPac sessions.
ketInterval]  [0] = (Default) Determined internally
 [1] = 5 msec (not recommended)
 [2] = 10 msec
 [3] = 20 msec
Note: When set to 5 msec (1), the maximum number of
simultaneous channels supported is 120.
EMS: Dynamic Jitter Buffer Defines the Jitter Buffer delay (in milliseconds) during fax and
Minimal Delay (dB) modem bypass session.
[FaxModemBypasDJBufMinDela The range is 0 to 150 msec. The default is 40.
y]
EMS: Enable Inband Network Enables in-band network detection related to fax/modem.
Detection  [0] = (Default) Disable.
[EnableFaxModemInbandNetwor  [1] = Enable. When this parameter is enabled on Bypass
kDetection] and transparent with events mode (VxxTransportType is set
to 2 or 3), a detection of an Answer Tone from the network
triggers a switch to bypass mode in addition to the local
Fax/Modem tone detections. However, only a high bit-rate
coder voice session effectively detects the Answer Tone sent
by a remote endpoint. This can be useful when, for example,
the payload of voice and bypass is the same, allowing the
originator to switch to bypass mode as well.
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EMS: NSE Mode Enables Cisco compatible fax and modem bypass mode.
[NSEMode]  [0] = (Default) NSE disabled
 [1] = NSE enabled
In NSE bypass mode, the device starts using G.711 A-Law
(default) or G.711µ-Law according to the
FaxModemBypassCoderType parameter. The payload type
used with these G.711 coders is a standard one (8 for G.711 A-
Law and 0 for G.711 µ-Law). The parameters defining payload
type for the 'old' Bypass mode FaxBypassPayloadType and
ModemBypassPayloadType are not used with NSE Bypass.
The bypass packet interval is selected according to the
FaxModemBypassBasicRtpPacketInterval parameter.
Notes:
 This feature can be used only if the
VxxModemTransportType parameter is set to 2 (Bypass).
 If NSE mode is enabled, the SDP contains the following line:
'a=rtpmap:100 X-NSE/8000'.
 To use this feature:
 The Cisco gateway must include the following definition:
'modem passthrough nse payload-type 100 codec
g711alaw'.
 Set the Modem transport type to Bypass mode
(VxxModemTransportType is set to 2) for all modems.
 Configure the gateway parameter NSEPayloadType =
100.
EMS: NSE Payload Type Defines the NSE payload type for Cisco Bypass compatible
[NSEPayloadType] mode.
The valid range is 96-127. The default is 105.
Note: Cisco gateways usually use NSE payload type of 100.
EMS: T38 Use RTP Port Defines the port (with relation to RTP port) for sending and
[T38UseRTPPort] receiving T.38 packets.
 [0] = (Default) Use the RTP port +2 to send/receive T.38
packets.
 [1] = Use the same port as the RTP port to send/receive
T.38 packets.
Notes:
 For this parameter to take effect, you must reset the device.
 When the device is configured to use V.152 to negotiate
audio and T.38 coders, the UDP port published in SDP for
RTP and for T38 must be different. Therefore, set the
T38UseRTPPort parameter to 0.
Web/EMS: T.38 Max Datagram Defines the maximum size of a T.38 datagram that the device
Size can receive. This value is included in the outgoing SDP when
[T38MaxDatagramSize] T.38 is used.
Web/EMS: T38 Fax Max Buffer Defines the maximum size (in bytes) of the device's T.38 buffer.
[T38FaxMaxBufferSize] This value is included in the outgoing SDP when T.38 is used
for fax relay over IP.

Web: Detect Fax on Answer Tone Determines when the device initiates a T.38 session for fax
EMS: Enables Detection of FAX on transmission.
Answer Tone  [0] Initiate T.38 on Preamble = (Default) The device to which
[DetFaxOnAnswerTone] the called fax is connected initiates a T.38 session on
receiving HDLC Preamble signal from the fax.
 [1] Initiate T.38 on CED = The device to which the called fax
is connected initiates a T.38 session on receiving a CED
answer tone from the fax. This option can only be used to
relay fax signals, as the device sends T.38 Re-INVITE on
detection of any fax/modem Answer tone (2100 Hz,
amplitude modulated 2100 Hz, or 2100 Hz with phase
reversals). The modem signal fails when using T.38 for fax
relay.
Note: This parameters is applicable only if the parameter
IsFaxUsed is set to 1 (T.38 Relay) or 3 (Fax Fallback).
Web: T38 Fax Session Immediate Enables fax transmission of T.38 "no-signal" packets to the
Start terminating fax machine.
[T38FaxSessionImmediateStart]  [0] Disable (default)
 [1] Immediate Start on Fax = Device activates T.38 fax relay
upon receipt of a re-INVITE with T.38 only in the SDP.
 [2] Immediate Start on Fax & Voice = Device activates T.38
fax relay upon receipt of a re-INVITE with T.38 and audio
media in the SDP.
This parameter is used for transmission from fax machines
connected to the device and located inside a NAT. Generally,
the firewall blocks T.38 (and other) packets received from the
WAN, unless the device behind NAT sends at least one IP
packet from the LAN to the WAN through the firewall. If the
firewall blocks T.38 packets sent from the termination IP fax, the
fax fails.
To overcome this, the device sends No-Op (“no-signal”) packets
to open a pinhole in the NAT for the answering fax machine.
The originating fax does not wait for an answer, but immediately
starts sending T.38 packets to the terminating fax machine.
Note: To enable No-Op packet transmission, use the
NoOpEnable and NoOpInterval parameters.
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44.11.2 DTMF and Hook-Flash Parameters

The DTMF and hook-flash parameters are described in the table below.
Table 44-39: DTMF and Hook-Flash Parameters
Hook-Flash Parameters
Web/EMS: Hook-Flash Code Defines the digit pattern that when received from the Tel side,
[HookFlashCode] indicates a Hook Flash event.
The valid range is a 25-character string. The default is a null
string.
Web/EMS: Hook-Flash Option Determines the hook-flash transport type (i.e., method by which
[HookFlashOption] hook-flash is sent and received).
 [0] Not Supported = (Default) Hook-Flash indication is not
sent.
 [1] INFO = Sends proprietary INFO message with Hook-Flash
indication.
 [4] RFC 2833
 [5] INFO (Lucent) = Sends proprietary SIP INFO message
with Hook-Flash indication.
 [6] INFO (NetCentrex) = Sends proprietary SIP INFO
message with Hook-Flash indication. The device sends the
INFO message as follows:
Content-Type: application/dtmf-relay
Signal=16
Where 16 is the DTMF code for hook flash.
 [7] INFO (HUAWEI) = Sends a SIP INFO message with Hook-
Flash indication. The device sends the INFO message as
follows:
Content-Length: 17
Content-Type: application/sscc
event=flashhook
Notes:
 FXO interfaces support only the receipt of RFC 2833 Hook-
Flash signals and INFO [1] type.
 FXS interfaces send Hook-Flash signals only if the
EnableHold parameter is set to 0.
Web: Min. Flash-Hook Detection Defines the minimum time (in msec) for detection of a hook-flash
Period [msec] event. Detection is guaranteed for hook-flash periods of at least
EMS: Min Flash Hook Time 60 msec (when setting the minimum time to 25). Hook-flash
[MinFlashHookTime] signals that last a shorter period of time are ignored.
Notes:
 It's recommended to reduce the detection time by 50 msec
from the desired value. For example, if you want to set the
value to 200 msec, then enter 150 msec (i.e., 200 minus 50).
Web: Max. Flash-Hook Detection Defines the hook-flash period (in msec) for both Tel and IP sides

Period [msec] (per device). For the IP side, it defines the hook-flash period that
EMS: Flash Hook Period is reported to the IP.
[FlashHookPeriod] For the analog side, it defines the following:
 FXS interfaces:
 Maximum hook-flash detection period. A longer signal is
considered an off-hook or on-hook event.
 Hook-flash generation period upon detection of a SIP
INFO message containing a hook-flash signal.
 FXO interfaces: Hook-flash generation period.
The valid range is 25 to 3,000. The default is 700.
Notes:
 For this parameter to take effect, you need to reset the device.
 For FXO interfaces, a constant of 100 msec must be added to
the required hook-flash period. For example, to generate a
450 msec hook-flash, set this parameter to 550.
 This parameter can also be configured in a Tel Profile.
DTMF Parameters
EMS: Use End of DTMF Determines when the detection of DTMF events is notified.
[MGCPDTMFDetectionPoint]  [0] = DTMF event is reported at the end of a detected DTMF
digit.
 [1] = (Default) DTMF event is reported at the start of a
detected DTMF digit.
Web: Declare RFC 2833 in SDP Defines the supported receive DTMF negotiation method.
EMS: Rx DTMF Option  [0] No = Don't declare RFC 2833 telephony-event parameter
[RxDTMFOption] in SDP.
 [3] Yes = (Default) Declare RFC 2833 telephony-event
parameter in SDP.
The device is always receptive to RFC 2833 DTMF relay packets.
Therefore, it is always correct to include the 'telephony-event'
parameter as default in the SDP. However, some devices use the
absence of the 'telephony-event' in the SDP to decide to send
DTMF digits in-band using G.711 coder. If this is the case, you
can set this parameter to 0.
Tx DTMF Option Table
Web/EMS: Tx DTMF Option This table parameter configures up to two preferred transmit
[TxDTMFOption] DTMF negotiation methods. The format of this parameter is as
follows:
[TxDTMFOption]
FORMAT TxDTMFOption_Index = TxDTMFOption_Type;
[\TxDTMFOption]
Where Type is:
 [0] Not Supported = (Default) No negotiation - DTMF digits are
sent according to the parameters DTMFTransportType and
RFC2833PayloadType.
 [1] INFO (Nortel) = Sends DTMF digits according to IETF
Internet-Draft draft-choudhuri-sip-info-digit-00.
 [2] NOTIFY = Sends DTMF digits according to IETF Internet-
Draft draft-mahy-sipping-signaled-digits-01.
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 [3] INFO (Cisco) = Sends DTMF digits according to Cisco
format.
 [4] RFC 2833.
 [5] INFO (Korea) = Sends DTMF digits according to Korea
Telecom format.
For example:
TxDTMFOption 0 = 1;
TxDTMFOption 1 = 3;
Notes:
 DTMF negotiation methods are prioritized according to the
order of their appearance.
 When out-of-band DTMF transfer is used ([1], [2], [3], or [5]),
the parameter DTMFTransportType is automatically set to 0
(DTMF digits are erased from the RTP stream).
 When RFC 2833 (4) is selected, the device:
a. Negotiates RFC 2833 payload type using local and
remote SDPs.
b. Sends DTMF packets using RFC 2833 payload type
according to the payload type in the received SDP.
c. Expects to receive RFC 2833 packets with the same
payload type as configured by the parameter
RFC2833PayloadType.
d. Removes DTMF digits in transparent mode (as part of the
voice stream).
 When TxDTMFOption is set to 0, the RFC 2833 payload type
is set according to the parameter RFC2833PayloadType for
both transmit and receive.
 The table ini file parameter TxDTMFOption can be repeated
twice for configuring the DTMF transmit methods.
[DisableAutoDTMFMute] Enables the automatic muting of DTMF digits when out-of-band
DTMF transmission is used.
 [0] = (Default) Automatic mute is used.
 [1] = No automatic mute of in-band DTMF.
When this parameter is set to 1, the DTMF transport type is set
according to the parameter DTMFTransportType and the DTMF
digits aren't muted if out-of-band DTMF mode is selected
(TxDTMFOption set to 1, 2 or 3). This enables the sending of
DTMF digits in-band (transparent of RFC 2833) in addition to out-
of-band DTMF messages.
Note: Usually this mode is not recommended.
Web/EMS: Enable Digit Delivery Enables the Digit Delivery feature whereby DTMF digits are sent
to IP to the destination IP address after the Tel-to-IP call is answered.
[EnableDigitDelivery2IP]  [0] Disable (default).
 [1] Enable = Enable digit delivery to IP.
To enable this feature, modify the called number to include at
least one 'p' character. The device uses the digits before the 'p'
character in the initial INVITE message. After the call is
answered, the device waits for the required time (number of 'p'
multiplied by 1.5 seconds), and then sends the rest of the DTMF
digits using the method chosen (in-band or out-of-band).
Notes:

 The called number can include several 'p' characters (1.5
seconds pause), for example, 1001pp699, 8888p9p300.
Web: Enable Digit Delivery to Tel Enables the Digit Delivery feature, which sends DTMF digits of
EMS: Enable Digit Delivery the called number to the device's port (phone line) after the call is
[EnableDigitDelivery] answered (i.e., line is off-hooked for FXS, or seized for FXO) for
IP-to-Tel calls.
 [1] Enable = Enable Digit Delivery feature for the FXO/FXS
device.
Notes:
 The called number can include characters 'p' (1.5 seconds
pause) and 'd' (detection of dial tone). If character 'd' is used, it
must be the first 'digit' in the called number. The character 'p'
can be used several times.
For example (for FXS/FXO interfaces), the called number can
be as follows: d1005, dpp699, p9p300. To add the 'd' and 'p'
digits, use the usual number manipulation rules.
 To use this feature with FXO interfaces, configure the device
to operate in one-stage dialing mode.
 If this parameter is enabled, it is possible to configure the
FXS/FXO interface to wait for dial tone per destination phone
number (before or during dialing of destination phone
number). Therefore, the parameter IsWaitForDialTone
(configurable for the entire device) is ignored.
 The FXS interface send SIP 200 OK responses only after the
DTMF dialing is complete.
[ReplaceNumberSignWithEsca Determines whether to replace the number sign (#) with the
peChar] escape character (%23) in outgoing SIP messages for Tel-to-IP
calls.
 [1] Enable = All number signs #, received in the dialed DTMF
digits are replaced in the outgoing SIP Request-URI and To
headers with the escape sign %23.
IsSpecialDigits is set 1.
Web: Special Digit Defines the representation for ‘special’ digits (‘*’ and ‘#’) that are
Representation used for out-of-band DTMF signaling (using SIP INFO/NOTIFY).
EMS: Use Digit For Special  [0] Special = (Default) Uses the strings ‘*’ and ‘#’.
DTMF  [1] Numeric = Uses the numerical values 10 and 11.
[UseDigitForSpecialDTMF]
[AdditionalOutOfBandDtmfFormat Enables the device to simultaneously send DTMF tones (signals)
] in SIP messages, e.g., INFO (out-of-band) and in RTP media
streams (in-band) with a special payload type (as defined in RFC
2833), when the FirstTxDTMFOption parameter is configured to
4.
 [0] unknown = (Default) DTMF is sent according to
FirstTxDTMFOption.
 [1] Nortel
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 [2] cisco
 [3] threecom
 [4] korea
44.11.3 Digit Collection and Dial Plan Parameters

The digit collection and dial plan parameters are described in the table below.
Table 44-40: Digit Collection and Dial Plan Parameters
Web/EMS: Dial Plan Index Defines the Dial Plan index to use in the external Dial Plan file. The
[DialPlanIndex] Dial Plan file is loaded to the device as a .dat file (converted using
the DConvert utility). The Dial Plan index can be defined globally or
per Tel Profile.
The valid value range is 0 to 7, where 0 denotes PLAN1, 1 denotes
PLAN2, and so on. The default is -1, indicating that no Dial Plan file
is used.
Notes:
 If this parameter is configured to select a Dial Plan index, the
settings of the parameter DigitMapping are ignored.
 If this parameter is configured to select a Dial Plan index from an
external Dial Plan file, the device first attempts to locate a
matching digit pattern in the Dial Plan file, and if not found, then
attempts to locate a matching digit pattern in the Digit Map rules
configured by the DigitMapping parameter.
 For more information on the Dial Plan file, see 'Dialing Plans for
Digit Collection' on page 376.
Defines the Dial Plan index in the external Dial Plan file for the Tel-
[Tel2IPSourceNumberMappi to-IP Source Number Mapping feature.
ngDialPlanIndex] The valid value range is 0 to 7, defining the Dial Plan index [Plan x]
in the Dial Plan file. The default is -1 (disabled).
For more information on this feature, see Modifying ISDN-to-IP
Calling Party Number.
Web: Digit Mapping Rules Defines the digit map pattern. If the digit string (i.e., dialed number)
EMS: Digit Map Pat terns matches one of the patterns in the digit map, the device stops
[DigitMapping] collecting digits and establishes a call with the collected number.
The digit map pattern can contain up to 52 options (rules), each
separated by a vertical bar (|). The maximum length of the entire digit
pattern is 152 characters. The available notations include the
following:
 [n-m]: Range of numbers (not letters).
 . (single dot): Repeat digits until next notation (e.g., T).
 x: Any single digit.
 T: Dial timeout (configured by the TimeBetweenDigits parameter).
 S: Short timer (configured by the TimeBetweenDigits parameter;
default is two seconds) that can be used when a specific rule is
defined after a more general rule. For example, if the digit map is
99|998, then the digit collection is terminated after the first two 9
digits are received. Therefore, the second rule of 998 can never

be matched. But when the digit map is 99s|998, then after dialing
the first two 9 digits, the device waits another two seconds within
which the caller can enter the digit 8.
An example of a digit map is shown below:
11xS|00T|[1-7]xxx|8xxxxxxx|#xxxxxxx|*xx|91xxxxxxxxxx|9011x.T
In the example above, the last rule can apply to International
numbers: 9 for dialing tone, 011 Country Code, and then any number
of digits for the local number ('x.').
Notes:
 If the DialPlanIndex parameter is configured (to select a Dial Plan
index), then the device first attempts to locate a matching digit
pattern in the Dial Plan file, and if not found, then attempts to
locate a matching digit pattern in the Digit Map rules configured
by the DigitMapping parameter.
 For more information on digit mapping, see 'Digit Mapping' on
page 275.
Web: Max Digits in Phone Defines the maximum number of collected destination number digits
Num that can be received (i.e., dialed) from the Tel side. When the
EMS: Max Digits in Phone number of collected digits reaches this maximum, the device uses
Number these digits for the called destination number.
[MaxDigits] The valid range is 1 to 49. The default is 5.
Notes:
 Instead of using this parameter, Digit Mapping rules can be
configured.
 Dialing ends when any of the following scenarios occur:
 Maximum number of digits is dialed
 Interdigit Timeout (TimeBetweenDigits) expires
 Pound (#) key is pressed
 Digit map pattern is matched
Web: Inter Digit Timeout for Defines the time (in seconds) that the device waits between digits
Overlap Dialing [sec] that are dialed by the user.
EMS: Interdigit Timeout (Sec) When this inter-digit timeout expires, the device uses the collected
[TimeBetweenDigits] digits to dial the called destination number.
Web: Enable Special Digits Determines whether the asterisk (*) and pound (#) digits can be used
EMS: Use '#' For Dial in DTMF.
Termination  [0] Disable = Use '*' or '#' to terminate number collection (refer to
[IsSpecialDigits] the parameter UseDigitForSpecialDTMF). (Default.)
 [1] Enable = Allows '*' and '#' for telephone numbers dialed by a
user or for the endpoint telephone number.
Note: These symbols can always be used as the first digit of a dialed
number even if you disable this parameter.
Version 6.6 577 MP-11x & MP-124

MP-11x & MP-124
44.11.4 Voice Mail Parameters

The voice mail parameters are described in the table below. For more information on the
Voice Mail application, refer to the CPE Configuration Guide for Voice Mail.
Note: Voice Mail is applicable only to FXO interfaces.
Table 44-41: Voice Mail Parameters
Web/EMS: Voice Mail Interface Enables the device's Voice Mail application and determines the
[VoiceMailInterface] communication method used between the PBX and the device.
 [1] DTMF
 [2] SMDI
Note: To disable voice mail per Hunt Group, you can use a Tel
Profile with the EnableVoiceMailDelay parameter set to disabled
(0). This eliminates the phenomenon of call delay on lines not
implementing voice mail when voice mail is enabled using this
global parameter.
Web: Enable VoiceMail URI Enables the interworking of target and cause for redirection from
EMS: Enable VMURI Tel to IP and vice versa, according to RFC 4468.
[EnableVMURI]  [0] Disable (default)
 [1] Enable
[WaitForBusyTime] Defines the time (in msec) that the device waits to detect busy
and/or reorder tones. This feature is used for semi-supervised
PBX call transfers (i.e., the LineTransferMode parameter is set
to 2).
The valid value range is 0 to 20000 (i.e., 20 sec). The default is
2000 (i.e., 2 sec).
Web/EMS: Line Transfer Mode Defines the call transfer method used by the device. This
[LineTransferMode] parameter is applicable to FXO call transfer.
 [0] None = (Default) IP.
 [1] Blind = PBX blind transfer:
 After receiving a SIP REFER message from the IP side,
the device (FXO) sends a hook-flash to the PBX, dials
the digits (that are received in the Refer-To header), and
then immediately releases the line (i.e., on-hook). The
PBX performs the transfer internally.
 [2] Semi Supervised = PBX semi-supervised transfer:
the device sends a hook-flash to the PBX, and then dials
the digits (that are received in the Refer-To header). If
no busy or reorder tones are detected (within the device
completes the call transfer by releasing the line. If these
tones are detected, the transfer is cancelled, the device
sends a SIP NOTIFY message with a failure reason in
the NOTIFY body (such as 486 if busy tone detected),
and generates an additional hook-flash toward the FXO

line to restore connection to the original call.
 [3] Supervised = PBX Supervised transfer:
the device sends a hook-flash to the PBX, and then dials
the digits (that are received in the Refer-To header). The
device waits for connection of the transferred call and
then completes the call transfer by releasing the line. If
speech is not detected, the transfer is cancelled, the
device sends a SIP NOTIFY message with a failure
reason in the NOTIFY body (such as 486 if busy tone
detected) and generates an additional hook-flash toward
the FXO line to restore connection to the original call.
SMDI Parameters
Web/EMS: Enable SMDI Enables Simplified Message Desk Interface (SMDI) interface on
[SMDI] the device.
 [0] Disable = (Default) Normal serial
 [1] Enable (Bellcore)
 [2] Ericsson MD-110
 [3] NEC (ICS)
Notes:
 When the RS-232 connection is used for SMDI messages
(Serial SMDI), it cannot be used for other applications, for
example, to access the Command Line Interface (CLI).
Web/EMS: SMDI Timeout Defines the time (in msec) that the device waits for an SMDI
[SMDITimeOut] Call Status message before or after a Setup message is
received. This parameter synchronizes the SMDI and analog
CAS interfaces.
If the timeout expires and only an SMDI message is received,
the SMDI message is dropped. If the timeout expires and only a
Setup message is received, the call is established.
The valid range is 0 to 10000 (i.e., 10 seconds). The default is
2000.
Message Waiting Indication (MWI) Parameters
Web: MWI Off Digit Pattern Defines the digit code used by the device to notify the PBX that
EMS: MWI Off Code there are no messages waiting for a specific extension. This
[MWIOffCode] code is added as prefix to the dialed number.
Web: MWI On Digit Pattern Defines the digit code used by the device to notify the PBX of
EMS: MWI On Code messages waiting for a specific extension. This code is added
[MWIOnCode] as prefix to the dialed number.
Web: MWI Suffix Pattern Defines the digit code used by the device as a suffix for 'MWI
EMS: MWI Suffix Code On Digit Pattern' and 'MWI Off Digit Pattern'. This suffix is
[MWISuffixCode] added to the generated DTMF string after the extension
number.
Version 6.6 579 MP-11x & MP-124

MP-11x & MP-124
Web: MWI Source Number Defines the calling party's phone number used in the Q.931
EMS: MWI Source Name MWI Setup message to PSTN. If not configured, the channel's
[MWISourceNumber] phone number is used as the calling number.
Defines the IP Group ID used when subscribing to an MWI
[MWISubscribeIPGroupID] server. The 'The SIP Group Name' field value of the IP Group
table is used as the Request-URI host name in the outgoing
MWI SIP SUBSCRIBE message. The request is sent to the IP
address defined for the Proxy Set that is associated with the IP
Group. The Proxy Set's capabilities such as proxy redundancy
and load balancing are also applied to the message.
For example, if the 'SIP Group Name' field of the IP Group is set
to "company.com", the device sends the following SUBSCRIBE
message:
SUBSCRIBE sip:company.com...
Instead of:
SUBSCRIBE sip:10.33.10.10...
Note: If this parameter is not configured, the MWI SUBSCRIBE
message is sent to the MWI server as defined by the
MWIServerIP parameter.
Digit Patterns The following digit pattern parameters apply only to voice mail applications that use
the DTMF communication method. For the available pattern syntaxes, refer to the CPE Configuration
Guide for Voice Mail.
Web: Forward on Busy Digit Defines the digit pattern used by the PBX to indicate 'call
Pattern (Internal) forward on busy' when the original call is received from an
EMS: Digit Pattern Forward On internal extension.
Busy The valid range is a 120-character string.
[DigitPatternForwardOnBusy]
Web: Forward on No Answer Digit Defines the digit pattern used by the PBX to indicate 'call
Pattern (Internal) forward on no answer' when the original call is received from an
EMS: Digit Pattern Forward On No internal extension.
Answer The valid range is a 120-character string.
[DigitPatternForwardOnNoAnsw
er]
Web: Forward on Do Not Disturb Defines the digit pattern used by the PBX to indicate 'call
Digit Pattern (Internal) forward on do not disturb' when the original call is received from
EMS: Digit Pattern Forward On an internal extension.
DND The valid range is a 120-character string.
[DigitPatternForwardOnDND]
Web: Forward on No Reason Digit Defines the digit pattern used by the PBX to indicate 'call
Pattern (Internal) forward with no reason' when the original call is received from
EMS: Digit Pattern Forward No an internal extension.
Reason The valid range is a 120-character string.
[DigitPatternForwardNoReason]
Web: Forward on Busy Digit Defines the digit pattern used by the PBX to indicate 'call
Pattern (External) forward on busy' when the original call is received from an
EMS: VM Digit Pattern On Busy external line (not an internal extension).
External The valid range is a 120-character string.
[DigitPatternForwardOnBusyExt]

Web: Forward on No Answer Digit Defines the digit pattern used by the PBX to indicate 'call
Pattern (External) forward on no answer' when the original call is received from an
EMS: VM Digit Pattern On No external line (not an internal extension).
Answer Ext The valid range is a 120-character string.
[DigitPatternForwardOnNoAnsw
erExt]
Web: Forward on Do Not Disturb Defines the digit pattern used by the PBX to indicate 'call
Digit Pattern (External) forward on do not disturb' when the original call is received from
EMS: VM Digit Pattern On DND an external line (not an internal extension).
[DigitPatternForwardOnDNDExt]
Web: Forward on No Reason Digit Defines the digit pattern used by the PBX to indicate 'call
Pattern (External) forward with no reason' when the original call is received from
EMS: VM Digit Pattern No Reason an external line (not an internal extension).
[DigitPatternForwardNoReasonE
xt]
Web: Internal Call Digit Pattern Defines the digit pattern used by the PBX to indicate an internal
EMS: Digit Pattern Internal Call call.
[DigitPatternInternalCall] The valid range is a 120-character string.
Web: External Call Digit Pattern Defines the digit pattern used by the PBX to indicate an external
EMS: Digit Pattern External Call call.
[DigitPatternExternalCall] The valid range is a 120-character string.
Web: Disconnect Call Digit Pattern Defines a digit pattern that when received from the Tel side,
EMS: Tel Disconnect Code indicates the device to disconnect the call.
[TelDisconnectCode] The valid range is a 25-character string.
Web: Digit To Ignore Digit Pattern Defines a digit pattern that if received as Src (S) or Redirect (R)
EMS: Digit To Ignore numbers is ignored and not added to that number.
[DigitPatternDigitToIgnore] The valid range is a 25-character string.
Version 6.6 581 MP-11x & MP-124

MP-11x & MP-124
44.11.5 Supplementary Services Parameters

This subsection describes the device's supplementary telephony services parameters.
44.11.5.1 Caller ID Parameters

The caller ID parameters are described in the table below.
Table 44-42: Caller ID Parameters
Caller ID Permissions Table

Web: Caller ID Permissions Table This table parameter enables (per port) Caller ID generation
EMS: SIP Endpoints > Caller ID (for FXS interfaces) and detection (for FXO interfaces).
[EnableCallerID] The format of this parameter is as follows:
[EnableCallerID]
FORMAT EnableCallerID_Index = EnableCallerID_IsEnabled;
[\EnableCallerID]
Where,
 Index = Port number (where 0 denotes Port 1).
For example:
EnableCallerID 0 = 1; (caller ID enabled on Port 1)
EnableCallerID 1 = 0; (caller ID disabled on Port 2)
Note: For a detailed description of this table, see Configuring
Caller ID Permissions on page 310.
Caller Display Information Table
Web: Caller Display Information This table parameter enables the device to send Caller ID
Table information to the IP side when a call is made. The called party
EMS: SIP Endpoints > Caller ID can use this information for caller identification. The information
[CallerDisplayInfo] configured in this table is sent in the SIP INVITE message's
From header.
[CallerDisplayInfo]
FORMAT CallerDisplayInfo_Index =
CallerDisplayInfo_DisplayString,
CallerDisplayInfo_IsCidRestricted;
[\CallerDisplayInfo]
Where,
 Index = Port number, where 0 denotes Port 1.
For example:
CallerDisplayInfo 0 = Susan C.,0; ("Susan C." is sent as
the Caller ID for Port 1)
CallerDisplayInfo 1 = Mark M.,0; ("Mark M." is sent as
Caller ID for Port 2)
Caller Display Information on page 307.

Web/EMS: Enable Caller ID Enables Caller ID.

[EnableCallerID]  [0] Disable (default)
 [1] Enable
If the Caller ID service is enabled, then for FXS interfaces,
calling number and Display text (from IP) are sent to the
device's port.
For FXO interfaces, the Caller ID signal is detected and sent to
IP in the SIP INVITE message (as 'Display' element).
For information on the Caller ID table, see Configuring Caller
Display Information on page 307.
To disable/enable caller ID generation per port, see
Configuring Call Forward on page 309.
Web: Caller ID Type Determines the standard used for detection (FXO) and
EMS: Caller id Types generation (FXS) of Caller ID, and detection (FXO) / generation
[CallerIDType] (FXS) of MWI (when specified) signals:
 [0] Standard Bellcore = (Default) Caller ID and MWI
 [1] Standard ETSI = Caller ID and MWI
 [2] Standard NTT
 [4] Standard BT = Britain
 [16] Standard DTMF Based ETSI
 [17] Standard Denmark = Caller ID and MWI
 [18] Standard India
 [19] Standard Brazil
Notes:
 Typically, the Caller ID signals are generated / detected
between the first and second rings. However, sometimes
the Caller ID is detected before the first ring signal. In such
a scenario, set the RingsBeforeCallerID parameter to 0.
 Caller ID detection for Britain [4] is not supported on the
device’s FXO ports. Only FXS ports can generate the Britain
[4] Caller ID.
 To select the Bellcore Caller ID sub standard, use the
BellcoreCallerIDTypeOneSubStandard parameter. To select
the ETSI Caller ID substandard, use the
ETSICallerIDTypeOneSubStandard parameter.
 To select the Bellcore MWI sub standard, use the
BellcoreVMWITypeOneStandard parameter. To select the
ETSI MWI sub standard, use the
ETSIVMWITypeOneStandard parameter.
 If you define Caller ID Type as NTT [2], you need to define
the NTT DID signaling form (FSK or DTMF) using the
NTTDIDSignallingForm parameter.
Version 6.6 583 MP-11x & MP-124

MP-11x & MP-124
Web: Enable FXS Caller ID Enables the interworking of Calling Party Category (cpc) code
Category Digit For Brazil Telecom from SIP INVITE messages to FXS Caller ID first digit.
[AddCPCPrefix2BrazilCallerID]  [0] Disable (default)
 [1] Enable
When this parameter is enabled, the device sends the Caller ID
number (calling number) with the cpc code (received in the SIP
INVITE message) to the device's FXS port. The cpc code is
added as a prefix to the caller ID (after IP-to-Tel calling number
manipulation). For example, assuming that the incoming
INVITE contains the following From (or P-Asserted-Id) header:
From:<sip:+551137077801;[email protected]>;
tag=53700
The calling number manipulation removes "+55" (leaving 10
digits), and then adds the prefix 7, the cpc code for payphone
user. Therefore, the Caller ID number that is sent to the FXS
port, in this example is 71137077801.
If the incoming INVITE message doesn't contain the 'cpc'
parameter, nothing is added to the Caller ID number.
CPC Value in CPC Code Description
Received INVITE Prefixed to Caller
ID (Sent to FXS
Endpoint)
cpc=unknown 1 Unknown user
cpc=subscribe 1 -
cpc=ordinary 1 Ordinary user
cpc=priority 2 Pre-paid user
cpc=test 3 Test user
cpc=operator 5 Operator
cpc=data 6 Data call
cpc=payphone 7 Payphone user
Notes:
 For this parameter to be enabled, you must also set the
parameter EnableCallingPartyCategory to 1.
[EnableCallerIDTypeTwo] Disables the generation of Caller ID type 2 when the phone is
off-hooked. Caller ID type 2 (also known as off-hook Caller ID)
is sent to a currently busy telephone to display the caller ID of
the waiting call.
 [0] = Caller ID type 2 isn't played.
 [1] = (Default) Caller ID type 2 is played.

EMS: Caller ID Timing Mode Determines when Caller ID is generated.

[AnalogCallerIDTimingMode]  [0] = (Default) Caller ID is generated between the first two
rings.
 [1] = The device attempts to find an optimized timing to
generate the Caller ID according to the selected Caller ID
type.
Notes:
 If this parameter is set to 1 and used with distinctive ringing,
the Caller ID signal doesn't change the distinctive ringing
timing.
EMS: Bellcore Caller ID Type One Determines the Bellcore Caller ID sub-standard.
Sub Standard  [0] = (Default) Between rings.
[BellcoreCallerIDTypeOneSubSta  [1] = Not ring related.
ndard]
required.
EMS: ETSI Caller ID Type One Sub Determines the ETSI FSK Caller ID Type 1 sub-standard (FXS
Standard only).
[ETSICallerIDTypeOneSubStanda  [0] = (Default) ETSI between rings.
rd]  [1] = ETSI before ring DT_AS.
 [2] = ETSI before ring RP_AS.
 [3] = ETSI before ring LR_DT_AS.
 [4] = ETSI not ring related DT_AS.
 [5] = ETSI not ring related RP_AS.
 [6] = ETSI not ring related LR_DT_AS.
required.
Web: Asserted Identity Mode Determines whether the SIP header P-Asserted-Identity or P-
EMS: Asserted ID Mode Preferred-Identity is used in the generated SIP INVITE, 200
[AssertedIdMode] OK, or UPDATE request for Caller ID (or privacy). These
headers are used to present the originating party's Caller ID.
The Caller ID is composed of a Calling Number and a Calling
Name (optional).
 [0] Disabled = (Default) P-Asserted-Identity nor P-Preferred-
Identity headers are not added.
 [1] Add P-Asserted-Identity
 [2] Add P-Preferred-Identity
The header used also depends on the calling Privacy (allowed
or restricted). These headers are used together with the
Privacy header. If Caller ID is restricted (i.e., P-Asserted-
Identity is not sent), the Privacy header includes the value 'id'
('Privacy: id'). Otherwise, for allowed Caller ID, 'Privacy: none'
is used. If Caller ID is restricted (received from PSTN / Tel or
configured in the device), the From header is set to
<[email protected]>.
Version 6.6 585 MP-11x & MP-124

MP-11x & MP-124
Web/EMS: Use Destination As Enables the device to include the Called Party Number, from
Connected Number outgoing Tel calls (after number manipulation), in the SIP P-
[UseDestinationAsConnectedNu Asserted-Identity header. The device includes the SIP P-
mber] Asserted-Identity header in 180 Ringing and 200 OK responses
 [1] Enable
Notes:
 For this feature, you must also enable the device to include
the P-Asserted-Identity header in 180/200 OK responses,
by setting the parameter AssertedIDMode to Add P-
Asserted-Identity.
 This parameter is applicable to FXO interfaces.
Web: Caller ID Transport Type Determines the device's behavior for Caller ID detection.
EMS: Transport Type  [0] Disable = The caller ID signal is not detected - DTMF
[CallerIDTransportType] digits remain in the voice stream.
 [1] Relay = (Currently not applicable.)
 [3] Mute = (Default) The caller ID signal is detected from the
Tel side and then erased from the voice stream.
Note: Caller ID detection is applicable only to FXO interfaces.
Reject Anonymous Calls Per Port Table
This table parameter determines whether the device rejects
[RejectAnonymousCallPerPort] incoming anonymous calls. If enabled, when a device's FXS
interface receives an anonymous call, it rejects the call and
responds with a SIP 433 (Anonymity Disallowed) response.
[RejectAnonymousCallPerPort]
FORMAT RejectAnonymousCallPerPort_Index =
RejectAnonymousCallPerPort_Enable;
[\RejectAnonymousCallPerPort]
Where,
 Enable = accept [0] (default) or reject [1] incoming
anonymous calls.
For example:
RejectAnonymousCallPerPort 0 = 0;
RejectAnonymousCallPerPort 1 = 1;

44.11.5.2 Call Waiting Parameters

The call waiting parameters are described in the table below.
Table 44-43: Call Waiting Parameters
Web/EMS: Enable Call Waiting Enables the Call Waiting feature.

[EnableCallWaiting]  [0] Disable
If enabled, when an FXS interface receives a call on a busy
endpoint, it responds with a 182 response (and not with a 486
busy). The device plays a call waiting indication signal. When
hook-flash is detected, the device switches to the waiting call.
The device that initiated the waiting call plays a call waiting
ringback tone to the calling party after a 182 response is
received.
Notes:
 The device's Call Progress Tones (CPT) file must include a
Call Waiting ringback tone (caller side) and a call waiting
tone (called side, FXS only).
 The EnableHold parameter must be enabled on both the
calling and the called side.
 You can use the table parameter CallWaitingPerPort to
enable Call Waiting per port.
 For information on the Call Waiting feature, see Call Waiting
on page 287.
EMS: Send 180 For Call Waiting Determines the SIP response code for indicating Call Waiting.
[Send180ForCallWaiting]  [0] = (Default) Use 182 Queued response to indicate call
waiting.
 [1] = Use 180 Ringing response to indicate call waiting.
Call Waiting Table
Web: Call Waiting Table This table parameter configures call waiting per FXS port. The
EMS: SIP Endpoints > Call Waiting format of this parameter is as follows:
[CallWaitingPerPort] [CallWaitingPerPort]
FORMAT CallWaitingPerPort_Index =
CallWaitingPerPort_IsEnabled;
[\CallWaitingPerPort]
For example:
CallWaitingPerPort 0 = 0; (call waiting disabled for Port 1)
CallWaitingPerPort 1 = 1; (call waiting enabled for Port 2)
Notes:
 Index denotes port number, where 0 denotes Port 1.
 This parameter is applicable only to FXS ports.
 For a detailed description of this table, see Configuring Call
Waiting on page 311.
Version 6.6 587 MP-11x & MP-124

MP-11x & MP-124
Web: Number of Call Waiting Defines the number of call waiting indications that are played to
Indications the called telephone that is connected to the device for Call
EMS: Call Waiting Number of Waiting.
Indications The valid range is 1 to 100 indications. The default is 2.
[NumberOfWaitingIndications]
Note: This parameter is applicable only to FXS ports.
Web: Time Between Call Waiting Defines the time (in seconds) between consecutive call waiting
Indications indications for call waiting.
EMS: Call Waiting Time Between The valid range is 1 to 100. The default is 10.
Indications
[TimeBetweenWaitingIndications]
Web/EMS: Time Before Waiting Defines the interval (in seconds) before a call waiting indication
Indications is played to the port that is currently in a call.
[TimeBeforeWaitingIndications] The valid range is 0 to 100. The default time is 0 seconds.
Web/EMS: Waiting Beep Duration Defines the duration (in msec) of call waiting indications that
[WaitingBeepDuration] are played to the port that is receiving the call.
EMS: First Call Waiting Tone ID Defines the index of the first Call Waiting Tone in the CPT file.
[FirstCallWaitingToneID] This feature enables the called party to distinguish between
different call origins (e.g., external versus internal calls).
There are three ways to use the distinctive call waiting tones:
 Playing the call waiting tone according to the SIP Alert-Info
header in the received 180 Ringing SIP response. The value
of the Alert-Info header is added to the value of the
FirstCallWaitingToneID parameter.
 Playing the call waiting tone according to PriorityIndex in the
ToneIndex table parameter.
 Playing the call waiting tone according to the parameter
“CallWaitingTone#' of a SIP INFO message.
The device plays the tone received in the 'play tone
CallWaitingTone#' parameter of an INFO message plus the
value of this parameter minus 1.
The valid range is -1 to 1,000. The default is -1 (i.e., not used).
Notes:
 It is assumed that all Call Waiting Tones are defined in
sequence in the CPT file.
 SIP Alert-Info header examples:
 Alert-Info:<Bellcore-dr2>
 Alert-Info:<http://…/Bellcore-dr2> (where "dr2" defines
call waiting tone #2)
 The SIP INFO message is according to Broadsoft's
application server definition. Below is an example of such an
INFO message:
INFO sip:[email protected]:5060 SIP/2.0
Via:SIP/2.0/UDP
192.168.13.40:5060;branch=z9hG4bK040066422630
From:
<sip:[email protected]:5060>;tag=1455352915
To: <sip:[email protected]:5060>

Call-ID:[email protected]
CSeq:342168303 INFO
Content-Length:28
Content-Type:application/broadsoft
play tone CallWaitingTone1
44.11.5.3 Call Forwarding Parameters

The call forwarding parameters are described in the table below.
Table 44-44: Call Forwarding Parameters
Web: Enable Call Forward Enables the Call Forwarding feature.

[EnableForward]  [0] Disable
For FXS interfaces, the Call Forward table (FwdInfo parameter) must
be defined to use the Call Forward service. The device uses SIP
REFER messages for call forwarding.
Notes:
 To use this service, the devices at both ends must support this
option.
 For the device to respond to SIP 3xx responses with a new SIP
request (forwarding the original request), set this parameter to
Enable.
Call Forwarding Table
Web: Call Forwarding Table This table parameter configures call forwarding of IP-to-Tel calls (using
EMS: Analog Gateway SIP 302 response) to other device ports or an IP destination, based on
Provisioning > Tab: Call the device's port to which the call was originally routed.
Forward The format of this parameter is as follows:
[FwdInfo] [FwdInfo]
FORMAT FwdInfo_Index = FwdInfo_Type, FwdInfo_Destination,
FwdInfo_NoReplyTime;
[\FwdInfo]
Where,
 Index = Port number, where 0 denotes Port 1.
For example:
 Below configuration forwards calls originally destined to Port 1 to
"1001" upon On Busy:
FwdInfo 0 = 1,1001,30;
 Below configuration forwards calls originally destined to Port 2 to an
IP address upon On Busy:
FwdInfo 1 = 1,[email protected],30;
Note: For a detailed description of this table, see Configuring Call
Forward on page 309.
Call Forward Reminder Ring Parameters
Notes:
 These parameters are applicable only to FXS interfaces.
 For a description of this feature, see Call Forward Reminder Ring on page 285.
Version 6.6 589 MP-11x & MP-124

MP-11x & MP-124
Web/EMS: Enable NRT Enables endpoint subscription for Ring reminder event notification
Subscription feature.
[EnableNRTSubscription]  [0] Disable (default)
 [1] Enable
Web: AS Subscribe Defines the IP Group ID that contains the Application server for
IPGroupID Subscription.
[ASSubscribeIPGroupID] The valid value range is 1 to 8. The default is -1 (i.e., not configured).
Web: NRT Retry Defines the Retry period (in seconds) for Dialog subscription if a
Subscription Time previous request failed.
EMS: NRT Subscription The valid value range is 10 to 7200. The default is 120.
Retry Time
[NRTSubscribeRetryTime]
Web/EMS: Call Forward Defines the ringing tone type played when call forward notification is
Ring Tone ID accepted.
[CallForwardRingToneID] The valid value range is 1 to 5. The default is 1.
44.11.5.4 Message Waiting Indication Parameters

The message waiting indication (MWI) parameters are described in the table below.
Table 44-45: MWI Parameters
Web: Enable MWI Enables Message Waiting Indication (MWI).

EMS: MWI Enable  [0] Disable (default).
[EnableMWI]  [1] Enable
Notes:
 The device supports only the receipt of SIP MWI NOTIFY
messages (the device doesn't generate these messages).
 For more information on MWI, see 'Message Waiting
Indication' on page 287.
Web/EMS: MWI Analog Lamp Enables the visual display of MWI.
[MWIAnalogLamp]  [0] Disable (default).
 [1] Enable = Enables visual MWI by supplying line voltage
of approximately 100 VDC to activate the phone's lamp.
Notes:
 This parameter is applicable only for FXS interfaces.
Web/EMS: MWI Display Enables sending MWI information to the phone display.
[MWIDisplay]  [0] Disable = (Default) MWI information isn't sent to display.
 [1] Enable = The device generates an MWI message
(determined by the parameter CallerIDType), which is
displayed on the MWI display.
Note:

Web: Subscribe to MWI Enables subscription to an MWI server.

EMS: Enable MWI Subscription  [0] No (default)
[EnableMWISubscription]  [1] Yes
Notes:
 To configure the MWI server address, use the MWIServerIP
parameter.
 To configure whether the device subscribes per endpoint or
per the entire device, use the parameter SubscriptionMode.
Web: MWI Server IP Address Defines the MWI server's IP address. If provided, the device
EMS: MWI Server IP subscribes to this IP address. The MWI server address can be
[MWIServerIP] configured as a numerical IP address or as a domain name. If
not configured, the Proxy IP address is used instead.
Web/EMS: MWI Server Transport Determines the transport layer used for outgoing SIP dialogs
Type initiated by the device to the MWI server.
[MWIServerTransportType]  [-1] Not Configured (default)
 [0] UDP
 [1] TCP
 [2] TLS
Note: When set to ‘Not Configured’, the value of the parameter
Web: MWI Subscribe Expiration Defines the MWI subscription expiration time in seconds.
Time The default is 7200 seconds. The range is 10 to 2,000,000.
EMS: MWI Expiration Time
[MWIExpirationTime]
Web: MWI Subscribe Retry Time Defines the subscription retry time (in seconds) after last
EMS: Subscribe Retry Time subscription failure.
[SubscribeRetryTime] The default is 120 seconds. The range is 10 to 2,000,000.
Web: Subscription Mode Determines the method the device uses to subscribe to an MWI
[SubscriptionMode] server.
 [0] Per Endpoint = (Default) Each endpoint subscribes
separately - typically used for FXS interfaces.
 [1] Per Gateway = Single subscription for the entire device -
typically used for FXO interfaces.
EMS: ETSI VMWI Type One Determines the ETSI Visual Message Waiting Indication
Standard (VMWI) Type 1 sub-standard.
[ETSIVMWITypeOneStandard]  [0] = (Default) ETSI VMWI between rings
 [1] = ETSI VMWI before ring DT_AS
 [2] = ETSI VMWI before ring RP_AS
 [3] = ETSI VMWI before ring LR_DT_AS
 [4] = ETSI VMWI not ring related DT_AS
 [5] = ETSI VMWI not ring related RP_AS
 [6] = ETSI VMWI not ring related LR_DT_AS
required.
EMS: Bellcore VMWI Type One Determines the Bellcore VMWI sub-standard.
Standard  [0] = (Default) Between rings.
[BellcoreVMWITypeOneStandard]  [1] = Not ring related.
Version 6.6 591 MP-11x & MP-124

MP-11x & MP-124
required.
[EnableLowVoltageMwiGeneration] Defines the Message Waiting Indication (MWI) voltage level
mode (low or high) that the FXS port generates to light a lamp
on an FXS phone to indicate a message in waiting.
 [0] = (Default) The FXS port generates a high DC voltage
(90 VDC) for the MWI signal:
 Constant MWI Light Mode: The FXS generates a
constant high DC voltage to light up the MWI indicator
on the phone. For this setup, make sure that the
parameters NeonMwiOnDurationTime and
NeonMwiOffDurationTime are configured to 1234
(default). (The parameters LedMwiOnDurationTime and
LedMwiOffDurationTime are not applicable.)
 Blinking MWI Light Mode: The FXS generates a high
DC voltage to light up the MWI indicator and a normal
on-hook voltage to turn it off. To configure the on-off
light duration (blinking duty cycle), use the parameters
NeonMwiOnDurationTime and
NeonMwiOffDurationTime. This blinking mode is less
recommended since the voltage transitions might "ding"
the phone's ringer. (The parameters
LedMwiOnDurationTime and LedMwiOffDurationTime
are not applicable.)
 [1] = The FXS port generates low on-hook voltage
transitions (at 50 Hz) for the MWI signal:
 Constant MWI Light Mode: The FXS constantly
generates on-hook voltage transitions to constantly light
up the MWI indicator on the phone. For this setup, make
sure that the parameters LedMwiOnDurationTime and
LedMwiOffDurationTime are configured to 1234
(default). (The parameters NeonMwiOnDurationTime
and NeonMwiOffDurationTime are not applicable.)
 Blinking MWI Light Mode: The FXS generates on-
hook voltage transitions to light up the MWI indicator
and stops transitioning the on-hook voltage to turn it off.
To configure the on-off light duration (blinking duty
cycle), use the parameters LedMwiOnDurationTime and
LedMwiOffDurationTime. (The parameters
NeonMwiOnDurationTime and
NeonMwiOffDurationTime are not applicable.) .
Note:
 The parameter is applicable only to FXS interfaces.
 The feature can be configured per port using the ini file
parameter syntax: EnableLowVoltageMwiGeneration_x,
where x is the port number.
[LedMwiOnDurationTime] Defines the duration (in msec) that the visual message waiting
indicator (lamp) on the phone is lit when using the on-hook low-
voltage transitions mode (i.e.,
EnableLowVoltageMwiGeneration is configured to 1).
If you want the MWI light indication to be lit constantly (instead
of blinking), configure this parameter and the
LedMwiOffDurationTime parameter to 1234.

If you want the MWI light indication to blink, configure this
parameter to any desired on-duration and the
LedMwiOffDurationTime parameter to any off-duration.
Note:
 The parameter applies to all FXS ports.
[LedMwiOffDurationTime] Defines the duration (in msec) that the visual message waiting
indicator (lamp) on the phone is off when using the on-hook
low-voltage transitions mode (i.e.,
EnableLowVoltageMwiGeneration is configured to 1).
LedMwiOnDurationTime parameter to 1234.
parameter to any desired off-duration and the
LedMwiOnDurationTime parameter to any on-duration.
Note:
[NeonMwiOnDurationTime] Defines the duration (in msec) that the visual message waiting
indicator (lamp) on the phone is lit when using the high-voltage
mode (i.e., EnableLowVoltageMwiGeneration is configured to
0).
NeonMwiOffDurationTime parameter to 1234 (default).
parameter to any desired on-duration and the
NeonMwiOffDurationTime parameter to any off-duration.
Note:
[NeonMwiOffDurationTime] Defines the duration (in msec) that the visual message waiting
indicator (lamp) on the phone is off when using the high-voltage
mode (i.e., EnableLowVoltageMwiGeneration is configured to
0).
NeonMwiOnDurationTime parameter to 1234 (default).
parameter to any desired off-duration and the
NeonMwiOnDurationTime parameter to any on-duration.
Version 6.6 593 MP-11x & MP-124

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Note:
44.11.5.5 Call Hold Parameters

The call hold parameters are described in the table below.
Table 44-46: Call Hold Parameters
Web/EMS: Enable Hold Enables the Call Hold feature that allows users, connected to the device,
[EnableHold] to place a call on hold (or remove from hold). This is done using the
phone's Hook Flash button. On receiving a hold request, the remote party
is placed on hold and hears the hold tone.
 [0] Disable
Notes:
 To use this service, the devices at both ends must support this option.
Web/EMS: Hold Format Determines the format of the SDP in the Re-INVITE hold request.
[HoldFormat]  [0] 0.0.0.0 = (Default) The SDP "c=" field contains the IP address
"0.0.0.0" and the "a=inactive" attribute.
 [1] Send Only = The SDP "c=" field contains the device's IP address
and the "a=sendonly" attribute.
 [2] x.y.z.t = The SDP "c=" field contains the device's IP address and
the "a=inactive" attribute.
Note: The device does not send any RTP packets when it is in hold
state.
Web/EMS:Held Timeout Defines the time interval that the device allows for a call to remain on
[HeldTimeout] hold. If a Resume (un-hold Re-INVITE) message is received before the
timer expires, the call is renewed. If this timer expires, the call is released
(terminated).
 [-1] = (Default) The call is placed on hold indefinitely until the initiator
of the on hold retrieves the call again.
 [0 - 2400] = Time to wait (in seconds) after which the call is released.
Web: Call Hold Reminder Defines the duration (in seconds) that the Call Hold Reminder Ring is
Ring Timeout played. If a user hangs up while a call is still on hold or there is a call
EMS: CHRR Timeout waiting, then the FXS interface immediately rings the extension for the
[CHRRTimeout] duration specified by this parameter. If the user off-hooks the phone, the
call becomes active.
Notes:
 This Reminder Ring feature can be disabled using the
DisableReminderRing parameter.

Disables the reminder ring, which notifies the FXS user of a call on hold
[DisableReminderRing] or a waiting call when the phone is returned to on-hook position.
 [0] = (Default) The reminder ring feature is active. In other words, if a
call is on hold or there is a call waiting and the phone is changed from
offhook to onhook, the phone rings (for a duration defined by the
CHRRTimeout parameter) to "remind" you of the call hold or call
waiting.
 [1] = Disables the reminder ring. If a call is on hold or there is a call
waiting and the phone is changed from offhook to onhook, the call is
released (and the device sends a SIP BYE to the IP).
Notes:
 This parameter is typically used for MLPP, allowing preemption to
clear held calls.
Determines whether the device sends DTMF signals (or DTMF SIP INFO
[PlayDTMFduringHold] message) when a call is on hold.
 [0] = (Default) Disable.
 [1] = Enable - If the call is on hold, the device stops playing the Held
tone (if it is played) and sends DTMF:
 To Tel side: plays DTMF digits according to the received SIP
INFO message(s). (The stopped held tone is not played again.)
 To IP side: sends DTMF SIP INFO messages to an IP destination
if it detects DTMF digits from the Tel side.
44.11.5.6 Call Transfer Parameters

The call transfer parameters are described in the table below.
Table 44-47: Call Transfer Parameters
Web/EMS: Enable Transfer Enables the Call Transfer feature.

[EnableTransfer]  [0] Disable
 [1] Enable = (Default) Enable the call transfer service
(using REFER). If the transfer service is enabled, the user
can activate Transfer using hook-flash signaling. If this
service is enabled, the remote party performs the call
transfer.
Notes:
 To use call transfer, the devices at both ends must support
this option.
 To use call transfer, set the parameter EnableHold to 1.
Version 6.6 595 MP-11x & MP-124

MP-11x & MP-124
Web: Transfer Prefix Defines the string that is added as a prefix to the
EMS: Logical Prefix For Transferred transferred/forwarded called number when the REFER/3xx
Call message is received.
[xferPrefix] Notes:
 The number manipulation rules apply to the user part of the
Refer-To and/or Contact URI before it is sent in the INVITE
message.
 This parameter can be used to apply different manipulation
rules to differentiate transferred/forwarded number from the
originally dialed number.
Web: Transfer Prefix IP 2 Tel Defines the prefix that is added to the destination number
[XferPrefixIP2Tel] received in the SIP Refer-To header (for IP-to-Tel calls). This
parameter is applicable to FXO blind transfer modes, i.e.,
LineTransferMode = 1, 2 or 3,.
The valid range is a string of up to 9 characters. The default is
an empty string.
Web/EMS: Enable Semi-Attended Determines the device behavior when Transfer is initiated
Transfer while in Alerting state.
[EnableSemiAttendedTransfer]  [0] Disable = (Default) Send REFER with the Replaces
header.
 [1] Enable = Send CANCEL, and after a 487 response is
received, send REFER without the Replaces header.
Web: Blind Defines the keypad sequence to activate blind transfer for
EMS: Blind Transfer established Tel-to-IP calls. The Tel user can perform blind
[KeyBlindTransfer] transfer by dialing the KeyBlindTransfer digits, followed by a
transferee destination number.
After the KeyBlindTransfer DTMF digits sequence is dialed,
the current call is put on hold (using a Re-INVITE message), a
dial tone is played to the channel, and then the phone number
collection starts.
After the destination phone number is collected, it is sent to
the transferee in a SIP REFER request in a Refer-To header.
The call is then terminated and a confirmation tone is played
to the channel. If the phone number collection fails due to a
mismatch, a reorder tone is played to the channel.
Note: For FXS/FXO interfaces, it is possible to configure
whether the KeyBlindTransfer code is added as a prefix to the
dialed destination number, by using the parameter
KeyBlindTransferAddPrefix.
EMS: Blind Transfer Add Prefix Determines whether the device adds the Blind Transfer code
[KeyBlindTransferAddPrefix] (defined by the KeyBlindTransfer parameter) to the dialed
destination number.
 [1] Enable
EMS: Blind Transfer Disconnect Defines the duration (in milliseconds) for which the device
Timeout waits for a disconnection from the Tel side after the Blind
[BlindTransferDisconnectTimeout] Transfer Code (KeyBlindTransfer) has been identified. When
this timer expires, a SIP REFER message is sent toward the
IP side. If this parameter is set to 0, the REFER message is
immediately sent.
The valid value range is 0 to 1,000,000. The default is 0.

44.11.5.7 Three-Way Conferencing Parameters

The three-way conferencing parameters are described in the table below.
Table 44-48: Three-Way Conferencing Parameters
Web: Enable 3-Way Conference Enables the 3-Way Conference feature.

EMS: Enable 3 Way  [0] Disable (default)
[Enable3WayConference]  [1] Enable
required.
Web: Three Way Conference Mode Determines the mode of operation when the 3-Way
EMS: 3 Way Mode Conference feature is used.
[3WayConferenceMode]  [0] AudioCodes Media Server = (Default) The Conference-
initiating INVITE (sent by the device) uses the
ConferenceID concatenated with a unique identifier as the
Request-URI. This same Request-URI is set as the Refer-
To header value in the REFER messages that are sent to
the two remote parties. This conference mode is used when
operating with AudioCodes IPMedia conferencing server.
 [1] Non-AudioCodes Media Server = The Conference-
initiating INVITE (sent by the device) uses only the
ConferenceID as the Request-URI. The conference server
sets the Contact header of the 200 OK response to the
actual unique identifier (Conference URI) to be used by the
participants. This Conference URI is then included (by the
device) in the Refer-To header value in the REFER
messages sent by the device to the remote parties. The
remote parties join the conference by sending INVITE
messages to the conference using this conference URI.
 [2] On Board = On-board, three-way conference. The
conference is established on the device without the need of
an external Conference server. You can limit the number of
simultaneous, on-board 3-way conference calls, by using
the MaxInBoardConferenceCalls parameter. The device
utilizes resources from idle ports to establish the conference
call. You can designate ports that you do not want to use as
resources for on-board, conference calls initiated by other
ports. This is configured using the
3WayConfNoneAllocateablePorts parameter.
 [3] Huawei Media Server = The conference is managed by
an external, third-party Conferencing server. The
conference-initiating INVITE sent by the device, uses only
the ConferenceID as the Request-URI. The Conferencing
server sets the Contact header of the 200 OK response to
the actual unique identifier (Conference URI) to be used by
the participants. The Conference URI is included in the URI
of the REFER with a Replaces header sent by the device to
the Conferencing server. The Conferencing server then
sends an INVITE with a Replaces header to the remote
participants.
Notes:
Version 6.6 597 MP-11x & MP-124

MP-11x & MP-124
 When using an external conference server (options [0] or
[1]), a conference call with up to six participants can be
established.
 For local, on-board three-way conferencing (option [2]) on
MP-112, the following additional parameter settings must be
made:
EnableIPMediaChannels = 1
[ IPMediaChannels ]
FORMAT IPMediaChannels_Index =
IPMediaChannels_ModuleID,
IPMediaChannels_DSPChannelsReserved;
IPMediaChannels 0 = 1, 2;
[ \IPMediaChannels ]
Web: Max 3 Way Conference Defines the maximum number of simultaneous, on-board
EMS: Max In Board Calls three-way conference calls.
[MaxInBoardConferenceCalls] The valid range is 0 to 2. The default is 2.
Notes:
 For enabling on-board, three-way conferencing, use the
3WayConferenceMode parameter.
Web: Three Way Conference Non Defines the ports that are not allocated as resources for on-
Allocatable Ports board three-way conference calls that are initiated by other
EMS: Non Allocateable Port ports. Ports that are not configured with this parameter (and
Number that are idle) are used by the device as a resource for
[3WayConfNoneAllocateablePort establishing these type of conference calls.
s] The valid range is up to 8 ports. To add a range of ports, use
the comma separator. For example, for not allowing the use of
ports 2, 4 and 8 as resources, enter the following value: 2,4,8.
The order of the entered values is not relevant (i.e., the
example above can be entered as 8,2,4). The default is 0.
Notes:
 To enable on-board, three-way conferencing, use the
3WayConferenceMode and MaxInBoardConferenceCalls
parameters.
Web: Establish Conference Code Defines the DTMF digit pattern, which upon detection
EMS: Establish Code generates the conference call when three-way conferencing is
[ConferenceCode] enabled (Enable3WayConference is set to 1).
The valid range is a 25-character string. The default is “!”
(Hook-Flash).
Note: If the FlashKeysSequenceStyle parameter is set to 1 or
2, the setting of the ConferenceCode parameter is overridden.
Web/EMS: Conference ID Defines the Conference Identification string.
[ConferenceID] The valid value is a string of up to 16 characters. The default is
"conf".
The device uses this identifier in the conference-initiating
INVITE that is sent to the media server when the
Enable3WayConference parameter is set to 1.

Web: Use Different RTP port After Enables the use of different RTP ports for the two calls
Hold involved in a three-way conference call made by the FXS
[UseDifferentRTPportAfterHold] endpoint in the initial outgoing INVITE requests.
 [0] Disable = First and second calls use the same RTP port
in the initial outgoing INVITE request. If a three-way
conference is then made, the device sends a re-INVITE to
the held call to retrieve it and to change the RTP port to a
different port number.
For example: The first call is made on port 6000 and
placed on hold. The second call is made, also on port
6000. The device sends a re-INVITE to the held call to
retrieve it and changes the port to 6010.
 [1] Enable = First and second calls use different RTP ports
in the initial outgoing INVITE request. If a three-way
conference is then made, the device sends a re-INVITE to
the held call to retrieve it, without changing the port of the
held call.
Notes:
 When this feature is enabled and only one RTP port is
available, only one call can be made by the FXS endpoint,
as there is no free RTP port for a second call.
 When this feature is enabled and you are using the Call
Forking feature, every forked call is sent with a different
RTP port. As the device can fork a call to up to 10
destinations, the device requires at least 10 free RTP ports.
44.11.5.8 MLPP and Emergency Call Parameters

The Multilevel Precedence and Preemption (MLPP) and emergency E911 call parameters
are described in the table below.
Table 44-49: MLPP and Emergency E911 Call Parameters
Web/EMS: Call Priority Mode Enables priority call handling for all calls.
[CallPriorityMode]  [0] Disable (default).
 [1] MLPP = MLPP Priority Call handling is enabled. MLPP
prioritizes call handling whereby the relative importance of
various kinds of communications is strictly defined, allowing
higher precedence communication at the expense of lower
precedence communications. Higher priority calls override
less priority calls when, for example, congestion occurs in a
network.
 [2] Emergency = Preemption of IP-to-Tel E911 emergency
calls. If the device receives an E911 call and there are
unavailable channels to receive the call, the device
terminates one of the channel calls and sends the E911 call
to that channel. The preemption is done only on a channel
pertaining to the same Hunt Group for which the E911 call
was initially destined and if the channel select mode
(configured by the ChannelSelectMode parameter) is set to
Version 6.6 599 MP-11x & MP-124

MP-11x & MP-124
other than “By Dest Number” (0). The preemption is done
only if the incoming IP-to-Tel call is identified as an
emergency call. The device identifies emergency calls by one
of the following:
 The destination number of the IP call matches one of the
numbers defined by the EmergencyNumbers parameter.
(For E911, you must define this parameter with the value
"911".)
 The incoming SIP INVITE message contains the
“emergency” value in the Priority header.
Notes:
 This parameter is applicable to FXS/FXO.
 For FXO interfaces, the preemption is done only on existing
IP-to-Tel calls. In other words, if all the current FXO channels
are busy with calls that were initiated by the FXO (i.e., Tel-to-
IP calls), new incoming emergency IP-to-Tel calls are
dropped.
 MLPP and Emergency services can also be configured in a
Tel Profile.
 For more information, see 'Pre-empting Existing Call for E911
IP-to-Tel Call' on page 293.
Emergency E911 Parameters
Web/EMS: Emergency Numbers Defines a list of “emergency” numbers.

[EmergencyNumbers] For FXS: When one of these numbers is dialed, the outgoing
INVITE message includes the SIP Priority and Resource-Priority
headers. If the user places the phone on-hook, the call is not
disconnected. Instead, a Hold Re-INVITE request is sent to the
remote party. Only if the remote party disconnects the call (i.e., a
BYE is received) or a timer expires (set by the
EmergencyRegretTimeout parameter) is the call terminated.
For FXO: These emergency numbers are used for the
preemption of E911 IP-to-Tel calls when there are unavailable or
busy channels. In this scenario, the device terminates one of the
busy channels and sends the emergency call to this channel.
This feature is enabled by setting the CallPriorityMode
parameter to 2 (“Emergency”). For a description of this feature,
see 'Pre-empting Existing Call for E911 IP-to-Tel Call' on page
293.
The list can include up to four different numbers, where each
number can be up to four digits long.
Example: EmergencyNumbers = ‘100’,’911’,’112’
Web: Emergency Calls Regret Defines the time (in minutes) that the device waits before
Timeout tearing-down an emergency call (defined by the parameter
EMS: Emergency Regret Timeout EmergencyNumbers). Until this time expires, an emergency call
[EmergencyRegretTimeout] can only be disconnected by the remote party, typically, by a
Public Safety Answering Point (PSAP).
Multilevel Precedence and Preemption (MLPP) Parameters

Web: MLPP DiffServ Defines the DiffServ value (differentiated services code
EMS: Diff Serv point/DSCP) used in IP packets containing SIP messages that
[MLPPDiffserv] are related to MLPP calls. This parameter defines DiffServ for
incoming MLPP calls with the Resource-Priority header.
Notes:
 The same value must be configured for this parameter and
the parameter PremiumServiceClassControlDiffServ.
 Outgoing calls are tagged according to the parameter
PremiumServiceClassControlDiffServ.
Web/EMS: Precedence Ringing Defines the index of the Precedence Ringing tone in the Call
Type Progress Tones (CPT) file. This tone is used when the
[PrecedenceRingingType] parameter CallPriorityMode is set to 1 and a Precedence call is
received from the IP side.
The valid range is -1 to 16. The default is -1 (i.e., plays standard
ringing tone).
EMS: E911 MLPP Behavior Defines the E911 (or Emergency Telecommunication
[E911MLPPBehavior] Services/ETS) MLPP Preemption mode:
 [0] = (Default) Standard Mode - ETS calls have the highest
priority and preempt any MLPP call.
 [1] = Treat as routine mode - ETS calls are handled as
routine calls.
[RPRequired] Determines whether the SIP resource-priority tag is added in the
SIP Require header of the INVITE message for Tel-to-IP calls.
 [0] Disable = Excludes the SIP resource-priority tag from the
SIP Require header.
 [1] Enable = (Default) Adds the SIP resource-priority tag in
the SIP Require header.
Note: This parameter is applicable only to MLPP priority call
handling (i.e., only when the CallPriorityMode parameter is set to
1).
Multiple Differentiated Services Code Points (DSCP) per MLPP Call Priority Level (Precedence)
Parameters
The MLPP service allows placement of priority calls, where properly validated users can preempt
(terminate) lower-priority phone calls with higher-priority calls. For each MLPP call priority level, the
DSCP can be set to a value from 0 to 63. The Resource Priority value in the Resource-Priority SIP
header can be one of the following:
MLPP Precedence Level Precedence Level in Resource-Priority SIP Header
0 (lowest) routine
2 priority
4 immediate
6 flash
8 flash-override
9 (highest) flash-override-override
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MP-11x & MP-124
Web/EMS: RTP DSCP for MLPP Defines the RTP DSCP for MLPP Routine precedence call level.
Routine The valid range is -1 to 63. The default is -1.
[MLPPRoutineRTPDSCP]
Note: If set to -1, the DiffServ value is taken from the global
parameter PremiumServiceClassMediaDiffServ or as defined in
IP Profiles per call.
Web/EMS: RTP DSCP for MLPP Defines the RTP DSCP for MLPP Priority precedence call level.
Priority The valid range is -1 to 63. The default is -1.
[MLPPPriorityRTPDSCP]
Web/EMS: RTP DSCP for MLPP Defines the RTP DSCP for MLPP Immediate precedence call
Immediate level.
[MLPPImmediateRTPDSCP] The valid range is -1 to 63. The default is -1.
Web/EMS: RTP DSCP for MLPP Defines the RTP DSCP for MLPP Flash precedence call level.
Flash The valid range is -1 to 63. The default is -1.
[MLPPFlashRTPDSCP]
Web/EMS: RTP DSCP for MLPP Defines the RTP DSCP for MLPP Flash-Override precedence
Flash Override call level.
[MLPPFlashOverRTPDSCP] The valid range is -1 to 63. The default is -1.
Web/EMS: RTP DSCP for MLPP Defines the RTP DSCP for MLPP Flash-Override-Override
Flash-Override-Override precedence call level.
[MLPPFlashOverOverRTPDSCP] The valid range is -1 to 63. The default is -1.
44.11.5.9 Call Cut-Through Parameters

The call cut-through parameters are described in the table below.
Table 44-50: Call Cut-Through Parameters

Web: Enable Calls Cut Enables FXS endpoints to receive incoming IP calls while the port is in off-
Through hook state.
EMS: Cut Through  [0] Disable (default)
[CutThrough]  [1] Enable
If enabled, the FXS interface answers the call and 'cuts through' the voice
channel if there is no other active call on the port, even if the port is in off-
hook state.
When the call is terminated (by the remote IP party), the device plays a
reorder tone for a user-defined time (configured by the
CutThroughTimeForReorderTone parameter) and is then ready to answer
the next incoming call without on-hooking the phone.
The waiting call is automatically answered by the device when the current
call is terminated (configured by setting the parameter EnableCallWaiting
to 1).
Note:
 This feature is applicable only to FXS interfaces.
 You can also configure the feature using an IP Profile
(TelProfile_IP2TelCutThroughCallBehavior):
 [0] NO cut through, no paging = Disabled
 [1] cutThrough = Channel Cut-Through enabled. When the IP side
ends the call, the device can play a reorder tone to the Tel side for
a user-defined duration (configured by the
CutThroughTimeForReorderTone parameter). Once the tone stops
playing, the FXS phone is ready to automatically answer another
incoming IP call, while in off-hook state.
 [2] cutThrough + paging = Channel Cut-Through enabled and no
tones are played.
44.11.5.10 Automatic Dialing Parameters

The automatic dialing upon off-hook parameters are described in the table below.
Table 44-51: Automatic Dialing Parameters
Automatic Dialing Table

Web: Automatic Dialing This table parameter defines telephone numbers that are automatically
Table dialed when a specific FXS or FXO port is off-hooked. The format of this
EMS: Analog Gateway parameter is as follows:
Provisioning > [TargetOfChannel]
Automatic dialing FORMAT TargetOfChannel_Index = TargetOfChannel_Destination,
[TargetOfChannel] TargetOfChannel_Type, TargetOfChannel_HotLineToneDuration;
[\TargetOfChannel]
For example, the below configuration defines automatic dialing of phone
number 911 when the phone connected to Port 1 is off-hooked for over 10
seconds:
TargetOfChannel 0 = 911, 1 ,10;
Notes:
 The first index of this table ini file parameter is 0.
 TargetOfChannel_Index is the port number, where 0 denotes Port 1.
 This is parameter is applicable only to FXS and FXO interfaces.
Version 6.6 603 MP-11x & MP-124

MP-11x & MP-124
 For a detailed description of this table, see 'Configuring Automatic
Dialing' on page 305.
44.11.5.11 Direct Inward Dialing Parameters

The Direct Inward Dialing (DID) parameters are described in the table below.
Table 44-52: DID Parameters
Web/EMS: DID Wink Enables Direct Inward Dialing (DID) using Wink-Start signaling, typically
[EnableDIDWink] used for signaling between an E-911 switch and the PSAP.
 [1] Single = The device can be used for connection to EIA/TIA-464B
DID Loop Start lines. Both FXO (detection) and FXS (generation) are
supported:
 The FXO interface dials DTMF (or MF) digits upon detection of a
Wink signal, instead of a dial tone.
 The FXS interface generates a Wink signal upon detection of an
off-hook state, instead of playing a dial tone.
Example: (Wink) KP I(I) xxx-xxxx ST (Off Hook)
Where:
 I = one or two information digits
 x = ANI
Note: The FXO interface generates such MF digits when the
Enable911PSAP parameter is set to 1.
 [2] Double Wink = Double-wink signaling. The FXS interface
generates the first wink upon detection of an off-hook state in the line.
The second wink is generated after a user-defined interval (configured
by the TimeBetweenDIDWinks parameter), after which the DTMF/MF
digits are collected by the device. Digits that arrive between the first
and second wink are ignored as they contain the same number.
Example: (Wink) KP 911 ST (Wink) KP I(I) xxx-xxxx ST (Off Hook)
 [3] Wink & Polarity= The FXS interface generates the first wink after it
detects an off-hook state. A polarity change from normal to reversed is
generated after a user-defined time (configured by the
TimeBetweenDIDWinks parameter). DTMF/MF digits are collected
only after this polarity change. Digits that arrive between the first wink
and the polarity change are ignored as they always contain the same
number. In this mode, the FXS interface does not generate a polarity
change to normal if the Tel-to-IP call is answered by an IP party.
Polarity reverts to normal when the call is released.
Example: (Wink) KP 911 ST (Polarity) KP I(I) xxx-xxxx ST (Off Hook)
Notes:
 Options [2] and [3] are applicable only to FXS interfaces.
 The EnableReversalPolarity and PolarityReversalType parameters
must be set to [1] for FXS interfaces.
 See also the Enable911PSAP parameter.

[TimeBetweenDIDWinks] Defines the interval (in msec) for wink signaling:

 Double-wink signaling [2]: interval between the first and second wink.
 Wink and Polarity signaling [3]: interval between wink and polarity
change.
Note: See the EnableDIDWink parameter for configuring the wink
signaling type.
Web/EMS: Delay Before Defines the time interval (in msec) between the detection of the off-hook
DID Wink and the generation of the DID Wink.
[DelayBeforeDIDWink] The valid range is 0 to 1,000. The default is 0.
EMS: NTT DID Signalling Determines the type of DID signaling support for NTT (Japan) modem:
Form DTMF- or Frequency Shift Keying (FSK)-based signaling. The devices
[NTTDIDSignallingForm] can be connected to Japan's NTT PBX using 'Modem' DID lines. These
DID lines are used to deliver a called number to the PBX.
 [0] = (Default) FSK-based signaling
 [1] = DTMF-based signaling
EMS: Enable DID This table parameter enables support for Japan NTT 'Modem' DID. FXS
[EnableDID] interfaces can be connected to Japan's NTT PBX using 'Modem' DID
lines. These DID lines are used to deliver a called number to the PBX.
The DID signal can be sent alone or combined with an NTT Caller ID
signal.
[EnableDID]
FORMAT EnableDID_Index = EnableDID_IsEnable;
[\EnableDID]
Where,
 Index = Port number (where 0 denotes Port 1).
 IsEnable = Enables [1] or disables [0] (default) Japan NTT Modem
DID support.
For example:
EnableDID 0 = 1; (DID is enabled on Port 1)
[WinkTime] Defines the time (in msec) elapsed between two consecutive polarity
reversals. This parameter can be used for DID signaling.
The valid range is 0 to 4,294,967,295. The default is 200.
Notes:
 This parameter is applicable to FXS and FXO interfaces.
44.11.6 Answer and Disconnect Supervision Parameters

The answer and disconnect supervision parameters are described in the table below.
Version 6.6 605 MP-11x & MP-124

MP-11x & MP-124
Table 44-53: Answer and Disconnect Parameters
Web: Answer Supervision Enables the sending of SIP 200 OK upon detection of
EMS: Enable Voice Detection speech, fax, or modem.
[EnableVoiceDetection]  [1] Yes = The device sends a SIP 200 OK (in response
to an INVITE message) when speech, fax, or modem is
detected.
 [0] No = (Default) The device sends a SIP 200 OK only
after it completes dialing.
Typically, this feature is used only when early media
(enabled using the EnableEarlyMedia parameter) is used
to establish the voice path before the call is answered.
Note: This feature is applicable only to one-stage dialing
(FXO).
Web/EMS: Max Call Duration (min) Defines the maximum duration (in minutes) of a call. If this
[MaxCallDuration] duration is reached, the device terminates the call. This
feature is useful for ensuring available resources for new
calls, by ensuring calls are properly terminated.
The valid range is 0 to 35,791. The default is 0 (i.e., no
limitation).
Web/EMS: Disconnect on Dial Tone Determines whether the device disconnects a call when a
[DisconnectOnDialTone] dial tone is detected from the PBX.
 [0] Disable = (Default) Call is not released.
 [1] Enable = Call is released if a dial tone is detected on
the device's FXO port.
Notes:
 This option is in addition to the mechanism that
disconnects a call when either busy or reorder tones are
detected.
Web: Send Digit Pattern on Connect Defines a digit pattern to send to the Tel side after a SIP
EMS: Connect Code 200 OK is received from the IP side. The digit pattern is a
[TelConnectCode] user-defined DTMF sequence that is used to indicate an
answer signal (e.g., for billing).
The valid range is 1 to 8 characters.
Note: This parameter is applicable to FXO.
Web: Disconnect on Broken Connection Determines whether the device releases the call if RTP
EMS: Disconnect Calls on Broken packets are not received within a user-defined timeout.
Connection  [0] No
[DisconnectOnBrokenConnection]  [1] Yes (default)
Notes:
 The timeout is configured by the
BrokenConnectionEventTimeout parameter.
 This feature is applicable only if the RTP session is
used without Silence Compression. If Silence
Compression is enabled, the device doesn't detect a
broken RTP connection.
 During a call, if the source IP address (from where the
RTP packets are received) is changed without notifying
the device, the device filters these RTP packets. To
overcome this, set the DisconnectOnBrokenConnection

parameter to 0; the device doesn't detect RTP packets
arriving from the original source IP address and
switches (after 300 msec) to the RTP packets arriving
from the new source IP address.
Web: Broken Connection Timeout Defines the time period (in 100-msec units) after which a
EMS: Broken Connection Event call is disconnected if an RTP packet is not received.
Timeout The valid range is from 3 (i.e., 300 msec) to an unlimited
[BrokenConnectionEventTimeout] value (e.g., 20 hours). The default is 100 (i.e., 10000 msec
or 10 seconds).
Notes:
 This parameter is applicable only if the parameter
DisconnectOnBrokenConnection is set to 1.
 Currently, this feature functions only if Silence
Suppression is disabled.
Web: Disconnect Call on Silence Determines whether calls are disconnected after detection
Detection of silence.
EMS: Disconnect On Detection Of  [1] Yes = The device disconnects calls in which silence
Silence occurs (in both call directions) for more than a user-
[EnableSilenceDisconnect] defined time.
 [0] No = (Default) Call is not disconnected when silence
is detected.
The silence duration can be configured by the
FarEndDisconnectSilencePeriod parameter (default 120).
Note: To activate this feature, enable Silence
Supppression for the used coder and configure the
FarEndDisconnectSilenceMethod parameter to 1.
Web: Silence Detection Period [sec] Defines the duration of the silence period (in seconds) after
EMS: Silence Detection Time Out which the call is disconnected.
[FarEndDisconnectSilencePeriod] The range is 10 to 28,800 (i.e., 8 hours). The default is 120
seconds.
required.
Web: Silence Detection Method Determines the silence detection method.
[FarEndDisconnectSilenceMethod]  [0] None = Silence detection option is disabled.
 [1] Packets Count = According to packet count.
 [2] Voice/Energy Detectors = (Default) According to
energy and voice detectors.
 [3] All = According to packet count, and energy and
voice detectors.
required.
Version 6.6 607 MP-11x & MP-124

MP-11x & MP-124
[FarEndDisconnectSilenceThreshold] Defines the threshold of the packet count (in percentages)

below which is considered silence by the device.
The valid range is 1 to 100%. The default is 8%.
Notes:
 This parameter is applicable only if silence is detected
according to packet count
(FarEndDisconnectSilenceMethod is set to 1).
required.
[BrokenConnectionDuringSilence] Enables the generation of the BrokenConnection event
during a silence period if the channel’s NoOp feature is
enabled (using the parameter NoOpEnable) and if the
channel stops receiving NoOp RTP packets.
 [1] Enable
Web: Disconnect Call on Busy Tone Determines whether a call is disconnected upon detection
Detection of a busy tone.
EMS: Disconnect On Detection End  [0] Disable = Call is not disconnected upon detection of
Tones a busy tone.
[DisconnectOnBusyTone]  [1] Enable = (Default) Call is released upon detection of
busy or reorder (fast busy) tone.
Notes:
Polarity (Current) Reversal for Call Release (Analog Interfaces) Parameters
[SetDefaultLinePolarityState] Defines the FXO line polarity, required for DID signaling.
 [0] = Positive line polarity
 [1] = Negative line polarity
 [2] = (Default) Auto - The device detects the polarity
upon power-up or upon insertion of the RJ-11 cable,
and uses it as a reference polarity.
Typically, if the RJ-11 cabling is connected correctly
(without crossing, Tip to Tip, Ring to Ring), the Tip line is
positive compared to the Ring line. In this case, set this
parameter to 0. With this configuration, the device
assumes that the idle line polarity is Tip line positive.
When the device receives a SIP INVITE, it checks the FXO
line polarity. If the polarity is "Reversed", it skips this FXO
line and goes to the next line.
Notes:
 A device reset is required for the settings of the
parameter to take effect.
 To take advantage of this new feature, configure all
FXO lines as a single Trunk Group with ascending or
descending channel select mode, and configure routing
rules to route incoming INVITE messages to this Trunk
Group.

Web: Enable Polarity Reversal Enables the polarity reversal feature for call release.
EMS: Enable Reversal Polarity  [0] Disable = (Default) Disable the polarity reversal
[EnableReversalPolarity] service.
 [1] Enable = Enable the polarity reversal service.
If the polarity reversal service is enabled, the FXS interface
changes the line polarity on call answer and then changes
it back on call release.
The FXO interface sends a 200 OK response when polarity
reversal signal is detected (applicable only to one-stage
dialing) and releases a call when a second polarity reversal
signal is detected.
Note: This parameter can also be configured in a Tel
Profile.
Web/EMS: Enable Current Disconnect Enables call release upon detection of a Current
[EnableCurrentDisconnect] Disconnect signal.
 [0] Disable = (Default) Disable the current disconnect
service.
 [1] Enable = Enable the current disconnect service.
If the current disconnect service is enabled:
 The FXO releases a call when a current disconnect
signal is detected on its port.
 The FXS interface generates a 'Current Disconnect
Pulse' after a call is released from IP.
The current disconnect duration is configured by the
CurrentDisconnectDuration parameter. The current
disconnect threshold (FXO only) is configured by the
CurrentDisconnectDefaultThreshold parameter. The
frequency at which the analog line voltage is sampled is
configured by the TimeToSampleAnalogLineVoltage
parameter.
Note: This parameter can also be configured in a Tel
Profile.
EMS: Polarity Reversal Type Defines the voltage change slope during polarity reversal
[PolarityReversalType] or wink.
 [0] = (Default) Soft reverse polarity.
 [1] = Hard reverse polarity.
Notes:
 Some Caller ID signals use reversal polarity and/or
Wink signals. In these cases, it is recommended to set
the parameter PolarityReversalType to 1 (Hard).
required.
Version 6.6 609 MP-11x & MP-124

MP-11x & MP-124
EMS: Current Disconnect Duration Defines the duration (in msec) of the current disconnect
[CurrentDisconnectDuration] pulse.
The range is 200 to 1500. The default is 900.
Notes:
 This parameter is applicable for FXS and FXO
interfaces.
 The FXO interface detection window is 100 msec below
the parameter's value and 350 msec above the
parameter's value. For example, if this parameter is set
to 400 msec, then the detection window is 300 to 750
msec.
required.
[CurrentDisconnectDefaultThreshold] Defines the line voltage threshold at which a current
disconnect detection is considered.
The valid range is 0 to 20 Volts. The default is 4 Volts.
Notes:
required.
Defines the frequency at which the analog line voltage is
[TimeToSampleAnalogLineVoltage] sampled (after offhook), for detection of the current
disconnect threshold.
The valid range is 100 to 2500 msec. The default is 1000
msec.
Notes:
required.

44.11.7 Tone Parameters

This subsection describes the device's tone parameters.
44.11.7.1 Telephony Tone Parameters

The telephony tone parameters are described in the table below.
Table 44-54: Tone Parameters
Web: SIP Hold Behavior Enables the device to handle incoming re-INVITE messages
[SIPHoldBehavior] with the "a=sendonly" attribute in the SDP, in the same way
as if an "a=inactive" is received in the SDP. When enabled,
the device plays a held tone to the Tel phone and responds
with a SIP 200 OK containing the "a=recvonly" attribute in the
SDP.
 [1] Enable
Web/EMS: Dial Tone Duration [sec] Defines the duration (in seconds) that the dial tone is played.
[TimeForDialTone] FXS interfaces play the dial tone after the phone is picked up
(off-hook). FXO interfaces play the dial tone after the port is
seized in response to ringing (from PBX/PSTN).
The valid range is 0 to 60. The default time is 16.
Notes:
 During play of dial tone, the device waits for DTMF digits.
 This parameter is not applicable when Automatic Dialing is
enabled.
Web/EMS: Stutter Tone Duration Defines the duration (in msec) of the confirmation tone. A
[StutterToneDuration] stutter tone is played (instead of a regular dial tone) when a
Message Waiting Indication (MWI) is received. The stutter
tone is composed of a confirmation tone (Tone Type #8),
which is played for the defined duration (StutterToneDuration)
followed by a stutter dial tone (Tone Type #15). Both these
tones are defined in the CPT file.
The range is 1,000 to 60,000. The default is 2,000 (i.e., 2
seconds).
Notes:
 If you want to configure the duration of the confirmation
tone to longer than 16 seconds, you must increase the
value of the parameter TimeForDialTone accordingly.
 The MWI tone takes precedence over the call forwarding
reminder tone. For more information on MWI, see
Message Waiting Indication on page 287.
Version 6.6 611 MP-11x & MP-124

MP-11x & MP-124
Web: FXO AutoDial Play BusyTone Determines whether the device plays a busy / reorder tone to
EMS: Auto Dial Play Busy Tone the PSTN side if a Tel-to-IP call is rejected by a SIP error
[FXOAutoDialPlayBusyTone] response (4xx, 5xx or 6xx). If a SIP error response is
received, the device seizes the line (off-hook), and then plays
a busy / reorder tone to the PSTN side (for the duration
defined by the parameter TimeForReorderTone). After
playing the tone, the line is released (on-hook).
 [1] = Enable
Note: This parameter is applicable only to FXO interfaces.
Web: Hotline Dial Tone Duration Defines the duration (in seconds) of the hotline dial tone. If no
EMS: Hot Line Tone Duration digits are received during this duration, the device initiates a
[HotLineToneDuration] call to a user-defined number (configured in the Automatic
Dialing table - TargetOfChannel - see Configuring Automatic
Dialing on page 305).
Notes:
 This parameter is applicable to FXS and FXO interfaces.
 You can define the Hotline duration per FXS/FXO port
using the Automatic Dialing table.
Web/EMS: Reorder Tone Duration Defines the duration (in seconds) that the device plays a busy
[sec] or reorder tone before releasing the line. Typically, after
[TimeForReorderTone] playing the busy or reorder tone for this duration, the device
starts playing an offhook warning tone.
The valid range is 0 to 254. The default is 0 seconds. Note
that the Web interface denotes the default value as a string
value of "255".
Notes:
 The selected busy or reorder tone is according to the SIP
release cause code received from IP.
Web: Time Before Reorder Tone Defines the delay interval (in seconds) from when the device
[sec] receives a SIP BYE message (i.e., remote party terminates
EMS: Time For Reorder Tone call) until the device starts playing a reorder tone to the FXS
[TimeBeforeReorderTone] phone.
Web: Cut Through Reorder Tone Defines the duration (in seconds) of the reorder tone played
Duration [sec] to the Tel side after the IP call party releases the call, for the
[CutThroughTimeForReOrderTone] Cut-Through feature. After the tone stops playing, an
incoming call is immediately answered if the FXS is off-
hooked.
The valid values are 0 to 30. The default is 0 (i.e., no reorder
tone is played).
Note: To enable the Cut-Through feature, use the
CutThrough (for FXS channels) parameter.

Web/EMS: Enable Comfort Tone Determines whether the device plays a comfort tone (Tone
[EnableComfortTone] Type #18) to the FXS/FXO endpoint after a SIP INVITE is
sent and before a SIP 18x response is received.
 [1] Enable
Note: This parameter is applicable to FXS and FXO
interfaces.
[WarningToneDuration] Defines the duration (in seconds) for which the offhook
warning tone is played to the user.
The valid range is -1 to 2,147,483,647. The default is 600.
Note: A negative value indicates that the tone is played
infinitely.
Web: Play Ringback Tone to Tel Determines the playing method of the ringback tone to the Tel
EMS: Play Ring Back Tone To Tel (for analog interfaces) side.
[PlayRBTone2Tel]  [0] Don't Play =
 Ringback tone is not played.
 [1] Play on Local =
 Plays a ringback tone to the Tel side of the call when
a SIP 180/183 response is received.
 [2] Prefer IP = (Default):
 Plays a ringback tone to the Tel side only if a 180/183
response without SDP is received. If 180/183 with
SDP message is received, the device cuts through the
voice channel and doesn't play the ringback tone.
 [3] Play Local Until Remote Media Arrive = Plays a
ringback tone according to received media. The behaviour
is similar to [2]. If a SIP 180 response is received and the
voice channel is already open (due to a previous 183 early
media response or due to an SDP in the current 180
response), the device plays a local ringback tone if there
are no prior received RTP packets. The device stops
playing the local ringback tone as soon as it starts
receiving RTP packets. At this stage, if the device receives
additional 18x responses, it does not resume playing the
local ringback tone.
Note: This parameter is applicable to the Gateway and IP-to-
IP applications.
Web: Play Ringback Tone to IP Determines whether the device plays a ringback tone to the
EMS: Play Ring Back Tone To IP IP side for IP-to-Tel calls.
[PlayRBTone2IP]  [0] Don't Play = (Default) Ringback tone isn't played.
 [1] Play = Ringback tone is played after SIP 183 session
progress response is sent.
Notes:
 To enable the device to send a 183/180+SDP responses,
set the EnableEarlyMedia parameter to 1.
 If the EnableDigitDelivery parameter is set to 1, the device
doesn't play a ringback tone to IP and doesn't send 183 or
180+SDP responses.
Version 6.6 613 MP-11x & MP-124

MP-11x & MP-124
Tone Index Table

Web: Tone Index Table This table parameter configures the Tone Index table, which
EMS: Analog Gateway Provisioning > allows you to define distinctive ringing and call waiting tones
Tone Index per FXS endpoint (or for a range of FXS endpoints).
[ToneIndex] The format of this parameter is as follows:
[ToneIndex]
FORMAT ToneIndex_Index = ToneIndex_FXSPort_First,
ToneIndex_FXSPort_Last, ToneIndex_SourcePrefix,
ToneIndex_DestinationPrefix, ToneIndex_PriorityIndex;
[\ToneIndex]
For example, the configuration below plays the tone Index #3
to FXS ports 1 and 2 if the source number prefix of the
received call is 20.
ToneIndex 1 = 1, 2, 20*, , 3;
FXS Distinctive Ringing and Call Waiting Tones per
Source/Destination Number.
44.11.7.2 Tone Detection Parameters

The signal tone detection parameters are described in the table below.
Table 44-55: Tone Detection Parameters
EMS: DTMF Enable Enables the detection of DTMF signaling.

[DTMFDetectorEnable]  [0] = Disable
 [1] = Enable (default)
EMS: MF R1 Enable Enables the detection of MF-R1 signaling.
[MFR1DetectorEnable]  [0] = Disable (default)
 [1] = Enable
EMS: User Defined Tone Enable Enables the detection of User Defined Tones signaling,
[UserDefinedToneDetectorEnable] applicable for Special Information Tone (SIT) detection.
 [1] = Enable
EMS: SIT Enable Enables SIT detection according to the ITU-T recommendation
[SITDetectorEnable] E.180/Q.35.
 [1] = Enable
(applicable to FXO interfaces):
 SITDetectorEnable = 1
 UserDefinedToneDetectorEnable = 1
 DisconnectOnBusyTone = 1 (applicable for busy, reorder,
and SIT tones)
required.

EMS: UDT Detector Frequency Defines the deviation (in Hz) allowed for the detection of each
Deviation signal frequency.
[UDTDetectorFrequencyDeviation] The valid range is 1 to 50. The default is 50.
required.
EMS: CPT Detector Frequency Defines the deviation (in Hz) allowed for the detection of each
Deviation CPT signal frequency.
[CPTDetectorFrequencyDeviation] The valid range is 1 to 30. The default is 10.
required.
44.11.7.3 Metering Tone Parameters

The metering tone parameters are described in the table below.
Table 44-56: Metering Tone Parameters
Web: Generate Metering Tones Determines the method used to configure the metering tones
EMS: Metering Mode that are generated to the Tel side.
[PayPhoneMeteringMode]  [0] Disable = (Default) Metering tones aren't generated.
 [1] Internal Table = Metering tones are generated
according to the device's Charge Code table (using the
ChargeCode parameter).
Notes:
 If you select 'Internal Table', you must configure the
Charge Codes table, using the ChargeCode parameter
(see Configuring Charge Codes Table on page 302).
Web: Analog Metering Type Determines the metering method for generating pulses
EMS: Metering Type (sinusoidal metering burst frequency) by the FXS port.
[MeteringType]  [0] 12 KHz = (Default) 12 kHz sinusoidal bursts.
 [1] 16 KHz = 16 kHz sinusoidal bursts.
 [2] = Polarity Reversal pulses.
Notes:
Web: Analog TTX Voltage Level Determines the metering signal/pulse voltage level (TTX).
EMS: TTX Voltage Level  [0] 0V = 0 Vrms sinusoidal bursts.
[AnalogTTXVoltageLevel ]  [1] 0.5V = (Default) 0.5 Vrms sinusoidal bursts.
 [2] 1V = 1 Vrms sinusoidal bursts
Notes:
Version 6.6 615 MP-11x & MP-124

MP-11x & MP-124
Charge Codes Table

Web: Charge Codes Table This table parameter configures metering tones and their time
EMS: Charge Codes intervals that the FXS interface generates to the Tel side.
[ChargeCode] The format of this parameter is as follows:
[ChargeCode]
FORMAT ChargeCode_Index = ChargeCode_EndTime1,
ChargeCode_PulseInterval1,
ChargeCode_PulsesOnAnswer1, ChargeCode_EndTime2,
ChargeCode_PulsesOnAnswer4;
[\ChargeCode]
Where,
 EndTime = Period (1 - 4) end time.
 PulseInterval = Period (1 - 4) pulse interval.
 PulsesOnAnswer = Period (1 - 4) pulses on answer.
For example:
ChargeCode 1 = 7,30,1,14,20,2,20,15,1,0,60,1;
ChargeCode 2 = 5,60,1,14,20,1,0,60,1;
ChargeCode 3 = 0,60,1;
ChargeCode 0 = 6, 3, 1, 12, 2, 1, 18, 5, 2, 0, 2, 1;
Notes:
 To associate a configured Charge Code to an outgoing Tel-
to-IP call, use the Tel to IP Routing.
 To configure the Charge Codes table using the Web
interface, see Configuring Charge Codes Table on page
302.
44.11.8 Telephone Keypad Sequence Parameters

The telephony keypad sequence parameters are described in the table below.
Table 44-57: Telephone Keypad Sequence Parameters
Web/EMS: Call Pickup Key Defines the keying sequence for performing a call pick-up. Call
[KeyCallPickup] pick-up allows the FXS endpoint to answer another telephone's
incoming call by pressing this user-defined sequence of digits.
When the user dials these digits (e.g., #77), the incoming call
from another phone is forwarded to the user's phone.
The valid value is a string of up to 15 characters (0-9, #, and *).
The default is undefined.
Notes:
 Call pick-up is configured only for FXS endpoints pertaining
to the same Hunt Group.

Prefix for External Line

[Prefix2ExtLine] Defines a string prefix (e.g., '9' dialed for an external line) that
when dialed, the device plays a secondary dial tone (i.e., stutter
tone) to the FXS line and then starts collecting the subsequently
dialed digits from the FXS line.
The valid range is a one-character string. The default is an
empty string.
Notes:
 You can enable the device to add this string as the prefix to
the collected (and sent) digits, using the parameter
AddPrefix2ExtLine.
Determines whether the prefix string for accessing an external
[AddPrefix2ExtLine] line (defined by the parameter Prefix2ExtLine) is added to the
dialed number as the prefix and together sent to the IP
destination (Tel-to-IP calls).
 [1] = Enable
For example, if this parameter is enabled and the prefix string
for the external line is defined as "9" (using the parameter
Prefix2ExtLine) and the FXS user wants to make a call to
destination "123", the device collects and sends all the dialed
digits, including the prefix string, as "9123" to the IP destination
number.
Hook Flash Parameters
Web: Flash Keys Sequence Style Determines the hook-flash key sequence for FXS interfaces.
[FlashKeysSequenceStyle]  [0] Flash hook = (Default) Only the phone's flash button is
used, according to the following scenarios:
 During an existing call, if the user presses the flash
button, the call is put on hold; a dial tone is heard and
the user is able to initiate a second call. Once the
second call is established, on-hooking transfers the first
(held) call to the second call.
 During an existing call, if a call comes in (call waiting),
pressing the flash button places the active call on hold
and answers the waiting call; pressing flash again
toggles between these two calls.
 [1] Sequence 1 = Sequence of flash and digit:
 Flash + 1: holds a call or toggles between two existing
calls
 Flash + 2: makes a call transfer.
 Flash + 3: makes a three-way conference call (if the
Three-Way Conference feature is enabled, i.e., the
parameter Enable3WayConference is set to 1 and the
parameter 3WayConferenceMode is set to 2).
 [2] Sequence 2 = Sequence of flash and digit:
 Flash only: Places a call on hold.
 Flash + 1:
1) When the device handles two calls (an active and a
held call) and this key sequence is dialed, it sends a SIP
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BYE message to the active call and the previously held
call becomes the active call.
2) When there is an active call and an incoming waiting
call, if this key sequence is dialed, the device
disconnects the active call and the waiting call becomes
an active call.
 Flash + 2: Places a call on hold and answers a call-
waiting call, or toggles between active and on-hold calls.
 Flash + 3: Makes a three-way conference call. This is
applicable only if the Enable3WayConference parameter
is set to 1 and the 3WayConferenceMode parameter is
set to 2. Note that the settings of the ConferenceCode
parameter is ignored.
 Flash + 4: Makes a call transfer.
Web: Flash Keys Sequence Defines the Flash keys sequence timeout - the time (in msec)
Timeout that the device waits for digits after the user presses the Flash
[FlashKeysSequenceTimeout] button (Flash Hook + Digit mode - when the parameter
FlashKeysSequenceStyle is set to 1 or 2).
The valid range is 100 to 5,000. The default is 2,000.
Keypad Feature - Call Forward Parameters
Web: Forward Unconditional Defines the keypad sequence to activate the immediate call
EMS: Call Forward Unconditional forward option.
[KeyCFUnCond]
Web: Forward No Answer Defines the keypad sequence to activate the forward on no
EMS: Call Forward No Answer answer option.
[KeyCFNoAnswer]
Web: Forward On Busy Defines the keypad sequence to activate the forward on busy
EMS: Call Forward Busy option.
[KeyCFBusy]
Web: Forward On Busy or No Defines the keypad sequence to activate the forward on 'busy
Answer or no answer' option.
EMS: CF Busy Or No Answer
[KeyCFBusyOrNoAnswer]
Web: Do Not Disturb
Defines the keypad sequence to activate the Do Not Disturb
EMS: CF Do Not Disturb
option (immediately reject incoming calls).
[KeyCFDoNotDisturb]
To activate the required forward method from the telephone:
1 Dial the user-defined sequence number on the keypad; a dial tone is heard.
2 Dial the telephone number to which the call is forwarded (terminate the number with #); a
confirmation tone is heard.
Web: Forward Deactivate Defines the keypad sequence to deactivate any of the call
EMS: Call Forward Deactivation forward options. After the sequence is pressed, a confirmation
[KeyCFDeact] tone is heard.
Keypad Feature - Caller ID Restriction Parameters

Web: Restricted Caller ID Activate Defines the keypad sequence to activate the restricted Caller ID
EMS: CLIR option. After the sequence is pressed, a confirmation tone is
[KeyCLIR] heard.
Web: Restricted Caller ID Defines the keypad sequence to deactivate the restricted Caller
Deactivate ID option. After the sequence is pressed, a confirmation tone is
EMS: CLIR Deactivation heard.
[KeyCLIRDeact]
Keypad Feature - Hotline Parameters
Web: Hot-line Activate Defines the keypad sequence to activate the delayed hotline
EMS: Hot Line option.
[KeyHotLine] To activate the delayed hotline option from the telephone,
perform the following:
1 Dial the user-defined sequence number on the keypad; a
dial tone is heard.
2 Dial the telephone number to which the phone automatically
dials after a configurable delay (terminate the number with
#); a confirmation tone is heard.
Web: Hot-line Deactivate Defines the keypad sequence to deactivate the delayed hotline
EMS: Hot Line Deactivation option. After the sequence is pressed, a confirmation tone is
[KeyHotLineDeact] heard.
Keypad Feature - Transfer Parameters
Note: See the description of the KeyBlindTransfer parameter for this feature.
Keypad Feature - Call Waiting Parameters
Web: Call Waiting Activate Defines the keypad sequence to activate the Call Waiting
EMS: Keypad Features CW option. After the sequence is pressed, a confirmation tone is
[KeyCallWaiting] heard.
Web: Call Waiting Deactivate Defines the keypad sequence to deactivate the Call Waiting
EMS: Keypad Features CW Deact option. After the sequence is pressed, a confirmation tone is
[KeyCallWaitingDeact] heard.
Keypad Feature - Reject Anonymous Call Parameters
Web: Reject Anonymous Call Defines the keypad sequence to activate the reject anonymous
Activate call option, whereby the device rejects incoming anonymous
EMS: Reject Anonymous Call calls. After the sequence is pressed, a confirmation tone is
[KeyRejectAnonymousCall] heard.
Web: Reject Anonymous Call Defines the keypad sequence that de-activates the reject
Deactivate anonymous call option. After the sequence is pressed, a
EMS: Reject Anonymous Call confirmation tone is heard.
Deact
[KeyRejectAnonymousCallDeact]
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44.11.9 General FXO Parameters

The general FXO and FXS parameters are described in the table below.
Table 44-58: General FXO and FXS Parameters
FXS Parameters
Web: FXS Coefficient Type Determines the FXS line characteristics (AC and DC)
EMS: Country Coefficients according to USA or Europe (TBR21) standards.
[FXSCountryCoefficients]  [66] Europe = TBR21
 [70] USA = (Default) United States
required.
[EnhancedFXSLineCurrent] Defines the FXS off-hook current, which is the current that the
device supplies to the analog line when it is in off-hook state.
 [0] 20 mA (Default)
 [1] 25 mA
 [2] 32 mA
Notes:
 The parameter is applicable only to the first four FXS
ports; the other ports have a fixed current of 20 mA.
 For the parameter to take effect, a device reset is
required.
FXO Parameters
Web: FXO Coefficient Type Determines the FXO line characteristics (AC and DC)
EMS: Country Coefficients according to USA or TBR21 standard.
[CountryCoefficients]  [66] Europe = TBR21
 [70] USA = (Default) United States
required.
Defines the FXO line DC termination (i.e., resistance).
[FXODCTermination]  [0] = (Default) DC termination is set to 50 Ohms.
 [1] = DC termination set to 800 Ohms. The termination
changes from 50 to 800 Ohms only when moving from
onhook to offhook.
required.
Enables limiting the FXO loop current to a maximum of 60
[EnableFXOCurrentLimit] mA (according to the TBR21 standard).
 [0] = (Default) FXO line current limit is disabled.
 [1] = FXO loop current is limited to a maximum of 60 mA.
required.
[FXONumberOfRings] Defines the number of rings before the device's FXO interface
answers a call by seizing the line.
When set to 0, the FXO seizes the line after one ring. When
set to 1, the FXO seizes the line after two rings.

Notes:
 This parameter is applicable only if automatic dialing is not
used.
 If caller ID is enabled and if the number of rings defined by
the parameter RingsBeforeCallerID is greater than the
number of rings defined by this parameter, the greater
value is used.
Web/EMS: Dialing Mode Determines the dialing mode for IP-to-Tel (FXO) calls.
[IsTwoStageDial]  [0] One Stage = One-stage dialing. In this mode, the
device seizes one of the available lines (according to the
ChannelSelectMode parameter), and then dials the
destination phone number received in the INVITE
message. To specify whether the dialing must start after
detection of the dial tone or immediately after seizing the
line, use the IsWaitForDialTone parameter.
 [1] Two Stages = (Default) Two-stage dialing. In this
mode, the device seizes one of the PSTN/PBX lines
without performing any dialing, connects the remote IP
user to the PSTN/PBX, and all further signaling (dialing
and Call Progress Tones) is performed directly with the
PBX without the device's intervention.
Web/EMS: Waiting For Dial Tone Determines whether or not the device waits for a dial tone
[IsWaitForDialTone] before dialing the phone number for IP-to-Tel (FXO) calls.
 [0] No
 [1] Yes (default)
When one-stage dialing and this parameter are enabled, the
device dials the phone number (to the PSTN/PBX line) only
after it detects a dial tone.
If this parameter is disabled, the device immediately dials the
phone number after seizing the PSTN/PBX line without
'listening' for a dial tone.
Notes:
 The correct dial tone parameters must be configured in the
CPT file.
 The device may take 1 to 3 seconds to detect a dial tone
(according to the dial tone configuration in the CPT file). If
the dial tone is not detected within 6 seconds, the device
releases the call and sends a SIP 500 "Server Internal
Error” response.
Web: Time to Wait before Dialing
[msec] Defines the delay before the device starts dialing on the FXO
EMS: Time Before Dial line in the following scenarios:
[WaitForDialTime]
 The delay between the time the line is seized and dialing
begins during the establishment of an IP-to-Tel call.
Note: Applicable only for one-stage dialing when the
parameter IsWaitForDialTone is disabled.
 The delay between detection of a Wink and the start of
dialing during the establishment of an IP-to-Tel call (for
DID lines, EnableDIDWink is set to 1).
 For call transfer - the delay after hook-flash is generated
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and dialing begins.
The valid range (in milliseconds) is 0 to 20,000 (i.e., 20
seconds). The default is 1,000 (i.e., 1 second).
Web: Ring Detection Timeout [sec] Defines the timeout (in seconds) for detecting the second ring
EMS: Timeout Between Rings after the first detected ring.
[FXOBetweenRingTime] If automatic dialing is not used and Caller ID is enabled, the
device seizes the line after detection of the second ring signal
(allowing detection of caller ID sent between the first and the
second rings). If the second ring signal is not received within
this timeout, the device doesn't initiate a call to IP.
If automatic dialing is used, the device initiates a call to IP
when the ringing signal is detected. The FXO line is seized
only if the remote IP party answers the call. If the remote
party doesn't answer the call and the second ring signal is not
received within this timeout, the device releases the IP call.
This parameter is typically set to between 5 and 8. The
default is 8.
Notes:
 This parameter is applicable only for Tel-to-IP calls.
 This timeout is calculated from the end of the ring until the
start of the next ring. For example, if the ring cycle is two
seconds on and four seconds off, the timeout value should
be configured to five seconds (i.e., greater than the off
time, e.g., four).
Web: Rings before Detecting Caller Determines the number of rings before the device starts
ID detecting Caller ID.
EMS: Rings Before Caller ID  [0] 0 = Before first ring.
[RingsBeforeCallerID]  [1] 1 = (Default) After first ring.
 [2] 2 = After second ring.
Web/EMS: Guard Time Between Defines the time interval (in seconds) after a call has ended
Calls and a new call can be accepted for IP-to-Tel (FXO) calls.
[GuardTimeBetweenCalls] The valid range is 0 to 10. The default is 1.
Note: Occasionally, after a call ends and on-hook is applied,
a delay is required before placing a new call (and performing
off-hook). This is necessary to prevent incorrect hook-flash
detection or other glare phenomena.
Web: FXO Double Answer Enables the FXO Double Answer feature, which rejects
[EnableFXODoubleAnswer] (disconnects) incoming Tel (FXO)-to-IP collect calls and
signals (informs) this call denial to the PSTN.
 [1] Enable
Note: This feature can also be configured in a Tel Profile.
FXO Ring Timeout Defines the delay (in msec) before the device generates a
fxo-ring-timeout SIP INVITE (call) to the IP side upon detection of a
RING_START event from the Tel (FXO) side. This occurs
[FXORingTimeout]
instead of waiting for a RING_END event.
This feature is useful for telephony services that employ
constant ringing (i.e., no RING_END is sent). For example,
Ringdown circuit is a service that sends a constant ringing
current over the line, instead of cadence-based 2 second on,

4 second off. For example, when a telephone goes off-hook,
a phone at the other end instantly rings.
If a RING_END event is received before the timeout expires,
the device does not initiate a call and ignores the detected
ring. The device ignores RING_END events detected after
the timeout expires.
The valid value range is 0 to 50 (msec), in steps of 100-msec.
For example, a value of 50 represents 5 sec. The default
value is 0 (i.e., standard ring operation - the FXO interface
sends an INVITE upon receipt of the RING_END event).
Note: The parameter can be configured for a Tel Profile.
[EnablePulseDialDetection] Enables the device to detect pulse (rotary) dialing from
analog equipment (e.g., telephones) connected to the
device's FXS port interfaces.
 [1] Enable
Note: For the parameter to take effect, a device reset is
required.
[EnablePulseDialGeneration] Enables pulse dialing generation to the analog side when
dialing is received from the FXO side.
 [0] Disable = (Default) Device generates DTMF signals.
 [1] Enable = Generates pulse dialing.
required.
[PulseDialGenerationBreakTime] Defines the duration of the Break connection (off-hook) for
FXO pulse dial generation.
The valid value range is 20 to 120 (in msec). The default is
60.
required.
[PulseDialGenerationMakeTime] Defines the duration of the Make connection (on-hook) for
FXO pulse dial generation.
40.
required.
[PulseDialGenerationInterDigitTime] Defines the inter-digit duration (time between consecutively
dialed digits) for FXO pulse dial generation.
700.
required.
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44.11.10 Hunt Groups and Routing Parameters

The routing parameters are described in the table below.
Table 44-59: Routing Parameters
Endpoint Phone Number Table

Web: Endpoint Phone Number Table This table parameter configures and activates the device's
EMS: SIP Endpoints > Phones endpoints. This is done by defining telephone numbers and
[TrunkGroup] assigning them to Hunt Groups. The format of this
parameter is shown below:
[TrunkGroup]
FORMAT TrunkGroup_Index =
TrunkGroup_TrunkGroupNum, TrunkGroup_FirstTrunkId,
TrunkGroup_FirstBChannel, TrunkGroup_LastBChannel,
TrunkGroup_FirstPhoneNumber, TrunkGroup_ProfileId,
TrunkGroup_LastTrunkId, TrunkGroup_Module;
[\TrunkGroup]
For example:
The configuration below assigns channels 1 through 4 to
Hunt Group 1 and assigns phone numbers 101 to Channel
1, 102 to Channel 2, and so on:
TrunkGroup 0 = 1, 255, 1, 4, 101, 0, 255, 255;
Note: For a description of this table, see Configuring
Endpoint Phone Numbers on page 235.
Hunt Group Settings
Web: Hunt Group Settings This table parameter configures the rules for channel
EMS: SIP Routing > Hunt Group allocation per Hunt Group. The format of this parameter is as
[TrunkGroupSettings] follows:
[TrunkGroupSettings]
FORMAT TrunkGroupSettings_Index =
TrunkGroupSettings_TrunkGroupId,
TrunkGroupSettings_ChannelSelectMode,
TrunkGroupSettings_RegistrationMode,
TrunkGroupSettings_GatewayName,
TrunkGroupSettings_ContactUser,
TrunkGroupSettings_ServingIPGroup,
TrunkGroupSettings_MWIInterrogationType,
TrunkGroupSettings_TrunkGroupName;
[\TrunkGroupSettings]
For example:
TrunkGroupSettings 0 = 1, 0, 5, branch-hq, user, 1, 255, ;
TrunkGroupSettings 1 = 2, 1, 0, localname, user1, 2, 255, ;
Note: For a description of this table, see 'Configuring Hunt
Group Settings' on page 237.

Web: Channel Select Mode Defines the method for allocating incoming IP-to-Tel calls to
EMS: Channel Selection Mode a channel (port) for all Hunt Groups.
[ChannelSelectMode]  [0] By Dest Phone Number (default)
 [1] Cyclic Ascending
 [2] Ascending
 [3] Cyclic Descending
 [4] Descending
 [5] Dest Number + Cyclic Ascending.
 [6] By Source Phone Number
 [9] Ring to Hunt Group
 [11] Dest Number + Ascending
Notes:
 For a detailed description of the parameter's options, see
'Configuring Hunt Group Settings' on page 237.
 Channel select mode per Hunt Group can be configured
in the Hunt Group Settings (see 'Configuring Hunt Group
Settings' on page 237).
Web: Default Destination Number Defines the default destination phone number, which is used
[DefaultNumber] if the received message doesn't contain a called party
number and no phone number is configured in the <Endpoint
Phone Number Table' (see to Configuring Endpoint Phone
Numbers on page 235). This parameter is used as a starting
number for the list of channels comprising all the device's
Hunt Groups.
The default is 1000.
Web: Source IP Address Input Determines which IP address the device uses to determine
[SourceIPAddressInput] the source of incoming INVITE messages for IP-to-Tel
routing.
 [-1] = (Default) Not configured.
 [0] SIP Contact Header = The IP address in the Contact
header of the incoming INVITE message is used.
 [1] Layer 3 Source IP = The actual IP address (Layer 3)
from where the SIP packet was received is used.
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Web: Use Source Number As Display Determines the use of Tel Source Number and Display
Name Name for Tel-to-IP calls.
EMS: Display Name  [0] No = (Default) If a Tel Display Name is received, the
[UseSourceNumberAsDisplayName] Tel Source Number is used as the IP Source Number
and the Tel Display Name is used as the IP Display
Name. If no Display Name is received from the Tel side,
the IP Display Name remains empty.
 [1] Yes = If a Tel Display Name is received, the Tel
Source Number is used as the IP Source Number and
the Tel Display Name is used as the IP Display Name. If
no Display Name is received from the Tel side, the Tel
Source Number is used as the IP Source Number and
also as the IP Display Name.
 [2] Overwrite = The Tel Source Number is used as the IP
Source Number and also as the IP Display Name (even if
the received Tel Display Name is not empty).
 [3] Original = Similar to option [2], except that the
operation is done before regular calling number
manipulation.
Web/EMS: Use Display Name as Defines how the display name (caller ID) received from the
Source Number IP side (in the SIP From header) effects the source number
[UseDisplayNameAsSourceNumber] sent to the Tel side, for IP-to-Tel calls.
 [0] No = (Default) If a display name is received from the
IP side, the source number of the IP side is used as the
Tel source number.
 [1] Yes = If a display name is received from the IP side,
the display name of the IP side is used as the Tel source
number and Presentation is set to Allowed (0). If no
display name is received from the IP side, the source
number of the IP side is used as the Tel source number
and Presentation is set to Restricted (1). For example:
 If 'From: 100 <sip:[email protected]>' is
received from the IP side, the outgoing source
number (and display name) are set to "100" and
Presentation is set to Allowed (0).
 If 'From: <sip:[email protected]>' is received
from the IP side, the outgoing source number is set
to "400" and Presentation is set to Restricted (1).
 [2] Preferred = If a display name is received from the IP
side, the display name of the IP side is used as the Tel
source number. If no display name is received from the
IP side, this setting does not affect the Tel source
number.
Web: Use Routing Table for Host Determines whether to use the device's routing table to
Names and Profiles obtain the URI host name and optionally, an IP profile (per
EMS: Use Routing Table For Host call) even if a Proxy server is used.
Names  [0] Disable = (Default) Don't use internal routing table.
[AlwaysUseRouteTable]  [1] Enable = Use the Tel to IP Routing.
Notes:
 This parameter appears only if the 'Use Default Proxy'
parameter is enabled.
 The domain name is used instead of a Proxy name or IP
address in the INVITE SIP URI.

Web/EMS: Tel to IP Routing Mode For a description of this parameter, see 'Configuring Tel to
[RouteModeTel2IP] IP Routing' on page 256.
Tel to IP Routing
Web: Tel to IP Routing This table parameter configures the Tel to IP Routing for
EMS: SIP Routing > Tel to IP routing Tel-to-IP calls. The format of this parameter is as
[Prefix] follows:
[PREFIX]
FORMAT PREFIX_Index = PREFIX_DestinationPrefix,
PREFIX_DestAddress, PREFIX_SourcePrefix,
PREFIX_ProfileId, PREFIX_MeteringCode,
PREFIX_DestPort, PREFIX_SrcIPGroupID,
PREFIX_DestHostPrefix, PREFIX_DestIPGroupID,
PREFIX_SrcHostPrefix, PREFIX_TransportType,
PREFIX_SrcTrunkGroupID, PREFIX_DestSRD,
PREFIX_CostGroup, PREFIX_ForkingGroup;
[\PREFIX]
For example:
PREFIX 0 = *, domain.com, *, 0, 255, $$, -1, , 1, , -1, -1, -1,,;
PREFIX 1 = 20, 10.33.37.77, *, 0, 255, $$, -1, , 2, , 0, -1,,;
Note: For a detailed description of this table, see
'Configuring Tel to IP Routing' on page 256.
IP to Hunt Group Routing Table
Web: IP to Hunt Group Routing Table This table parameter configures the routing of IP-to-Hunt
EMS: SIP Routing > IP to Hunt Groups. The format of this parameter is as follows:
[PSTNPrefix] [PSTNPrefix]
ORMAT PstnPrefix_Index = PstnPrefix_DestPrefix,
PstnPrefix_TrunkGroupId, PstnPrefix_SourcePrefix,
PstnPrefix_SourceAddress, PstnPrefix_ProfileId,
PstnPrefix_SrcIPGroupID, PstnPrefix_DestHostPrefix,
PstnPrefix_SrcHostPrefix, PstnPrefix_SrcSRDID,
PstnPrefix_TrunkId;
[\PSTNPrefix]
For example:
PstnPrefix 0 = 100, 1, 200, *, 0, 2, , , ,;
PstnPrefix 1 = *, 2, *, , 1, 3, acl, joe, , ,;
Note: For a detailed description of this table, see
'Configuring IP to Hunt Group Routing Table' on page 263.
Web/EMS: IP to Tel Routing Mode Determines whether to route IP calls to the Hunt Group
[RouteModeIP2Tel] before or after manipulation of the destination number
(configured in 'Configuring Source/Destination Number
Manipulation Rules' on page 241).
 [0] Route calls before manipulation = (Default) Calls are
routed before the number manipulation rules are applied.
 [1] Route calls after manipulation = Calls are routed after
the number manipulation rules are applied.
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Web: IP Security Determines the device's policy on accepting or blocking SIP

EMS: Secure Call From IP calls (IP-to-Tel calls). This is useful in preventing unwanted
[SecureCallsFromIP] SIP calls, SIP messages, and/or VoIP spam.
 [0] Disable = (Default) The device accepts all SIP calls.
 [1] Secure Incoming calls = The device accepts SIP calls
(i.e., calls from the IP side) only from IP addresses that
are defined in the Tel to IP Routing or Proxy Set table, or
IP addresses resolved from DNS servers from FQDN
values defined in the Proxy Set table. All other incoming
calls are rejected.
 [2] Secure All calls = The device accepts SIP calls only
from IP addresses (in dotted-decimal notation format)
that are defined in the Tel to IP Routing table or Proxy
Set table, and rejects all other incoming calls. In addition,
if an FQDN is defined in the routing table or Proxy Set
table, the call is allowed to be sent only if the resolved
DNS IP address appears in one of these tables;
otherwise, the call is rejected. Therefore, the difference
between this option and option [1] is that this option is
concerned only about numerical IP addresses that are
defined in the tables.
Note: If this parameter is set to [0] or [1], when using
Proxies or Proxy Sets, it is unnecessary to configure the
Proxy IP addresses in the routing table. The device allows
SIP calls received from the Proxy IP addresses even if these
addresses are not configured in the routing table.
Web/EMS: Filter Calls to IP Enables filtering of Tel-to-IP calls when a Proxy is used (i.e.,
[FilterCalls2IP] IsProxyUsed parameter is set to 1 - see 'Configuring Proxy
and Registration Parameters' on page 216).
 [0] Don't Filter = (Default) The device doesn't filter calls
when using a Proxy.
 [1] Filter = Filtering is enabled.
When this parameter is enabled and a Proxy is used, the
device first checks the Tel to IP Routingbefore making a call
through the Proxy. If the number is not allowed (i.e., number
isn't listed in the table or a call restriction routing rule of IP
address 0.0.0.0 is applied), the call is released.
Note: When no Proxy is used, this parameter must be
disabled and filtering is according to the Tel to IP Routing.

Web: Add CIC Determines whether to add the Carrier Identification Code
[AddCicAsPrefix] (CIC) as a prefix to the destination phone number for IP-to-
Tel calls. When this parameter is enabled, the 'cic'
parameter in the incoming SIP INVITE can be used for IP-to-
Tel routing decisions. It routes the call to the appropriate
Hunt Group based on this parameter's value.
 [1] Yes
For example, as a result of receiving the below INVITE, the
destination number after number manipulation is
cic+167895550001:
INVITE
sip:5550001;[email protected]:5060;user=phone
SIP/2.0
Note: After the cic prefix is added, the IP to Hunt Group
Routing Table can be used to route this call to a specific
Hunt Group. The Destination Number IP to Tel Manipulation
table must be used to remove this prefix before placing the
call to the Tel.
Web: ENUM Resolution Defines the ENUM service for translating telephone numbers
CLI: enum-service-domain to IP addresses or domain names (FQDN). For example,
[EnumService] e164.arpa, e164.customer.net, or NRENum.net.
The valid value is a string of up to 50 characters. The default
is "e164.arpa".
Note: ENUM-based routing is configured in the Outbound IP
Routing table using the "ENUM" string value as the
destination address to denote this parameter's value.
44.11.11 IP Connectivity Parameters

The IP connectivity parameters are described in the table below.
Table 44-60: IP Connectivity Parameters
Web: Enable Alt Routing Tel to IP Enables the Alternative Routing feature for Tel-to-IP calls.
EMS: Enable Alternative Routing  [0] Disable = (Default) Disables the Alternative Routing
[AltRoutingTel2IPEnable] feature.
 [1] Enable = Enables the Alternative Routing feature.
 [2] Status Only = The Alternative Routing feature is
disabled, but read-only information on the QoS of the
destination IP addresses is provided.
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Web: Alt Routing Tel to IP Mode Determines the IP Connectivity event(s) reason for triggering
EMS: Alternative Routing Mode Alternative Routing.
[AltRoutingTel2IPMode]  [0] None = Alternative routing is not used.
 [1] Connectivity = Alternative routing is performed if a ping
or SIP OPTIONS message to the initial destination fails
(determined according to the AltRoutingTel2IPConnMethod
parameter).
 [2] QoS = Alternative routing is performed if poor QoS is
detected.
 [3] Both = (Default) Alternative routing is performed if either
ping or SIP OPTIONS to initial destination fails, poor QoS is
detected, or the DNS host name is not resolved.
Notes:
 QoS is quantified according to delay and packet loss
calculated according to previous calls. QoS statistics are
reset if no new data is received within two minutes.
 To receive quality information (displayed in the 'Quality
Status' and 'Quality Info.' fields in 'Viewing IP Connectivity'
on page 421) per destination, this parameter must be set to
2 or 3.
Web: Alt Routing Tel to IP Determines the method used by the device for periodically
Connectivity Method querying the connectivity status of a destination IP address.
EMS: Alternative Routing  [0] ICMP Ping = (Default) Internet Control Message Protocol
Telephone to IP Connection (ICMP) ping messages.
Method  [1] SIP OPTIONS = The remote destination is considered
[AltRoutingTel2IPConnMethod] offline if the latest OPTIONS transaction timed out. Any
response to an OPTIONS request, even if indicating an
error, brings the connectivity status to online.
Web: Alt Routing Tel to IP Keep Defines the time interval (in seconds) between SIP OPTIONS
Alive Time Keep-Alive messages used for the IP Connectivity application.
EMS: Alternative Routing Keep The valid range is 5 to 2,000,000. The default is 60.
Alive Time
[AltRoutingTel2IPKeepAliveTime]
Web: Max Allowed Packet Loss for Defines the packet loss (in percentage) at which the IP
Alt Routing [%] connection is considered a failure and Alternative Routing
[IPConnQoSMaxAllowedPL] mechanism is activated.
The default is 20%.
Web: Max Allowed Delay for Alt Defines the transmission delay (in msec) at which the IP
Routing [msec] connection is considered a failure and the Alternative Routing
[IPConnQoSMaxAllowedDelay] mechanism is activated.
The range is 100 to 10,000. The default is 250.

44.11.12 Alternative Routing Parameters

The alternative routing parameters are described in the table below.
Table 44-61: Alternative Routing Parameters
Web/EMS: Redundant Determines the type of redundant routing mechanism when a call
Routing Mode can’t be completed using the main route.
[RedundantRoutingMode]  [0] Disable = No redundant routing is used. If the call can’t be
completed using the main route (using the active Proxy or the first
matching rule in the Routing table), the call is disconnected.
 [1] Routing Table = (Default) Internal routing table is used to
locate a redundant route.
 [2] Proxy = Proxy list is used to locate a redundant route.
Note: To implement the Redundant Routing Mode mechanism, you
first need to configure the parameter AltRouteCauseTEL2IP
(Reasons for Alternative Routing table).
[EnableAltMapTel2IP] Enables different Tel-to-IP destination number manipulation rules per
routing rule when several (up to three) Tel-to-IP routing rules are
defined and if alternative routing using release causes is used. For
example, if an INVITE message for a Tel-to-IP call is returned with a
SIP 404 Not Found response, the call can be re-sent to a different
destination number (as defined using the parameter
NumberMapTel2IP).
 [1] = Enable
Web/EMS: Alternative Defines the duration (in milliseconds) for which the device plays a
Routing Tone Duration [ms] tone to the endpoint on each attempt for Tel-to-IP alternative routing.
[AltRoutingToneDuration] When the device finishes playing the tone, a new SIP INVITE
message is sent to the new IP destination. The tone played is the call
forward tone (Tone Type #25 in the CPT file).
The valid range is 0 to 20,000. The default is 0 (i.e., no tone is
played).
Note: This parameter is applicable only to Tel-to-IP alternative
routing.
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Reasons for Alternative Tel-to-IP Routing Table

Web: Reasons for Alternative This table parameter configures SIP call failure reason values
Routing received from the IP side. If an IP call is released as a result of one of
EMS: Alt Route Cause Tel to these reasons, the device attempts to locate an alternative IP route
IP for the call in the Tel to IP Routing (if a Proxy is not used) or used as
[AltRouteCauseTel2IP] a redundant Proxy (you need to set the parameter
RedundantRoutingMode to 2). The release reason for Tel-to-IP calls
is provided in SIP 4xx, 5xx, and 6xx response codes.
[AltRouteCauseTel2IP]
FORMAT AltRouteCauseTel2IP_Index =
AltRouteCauseTel2IP_ReleaseCause;
[\AltRouteCauseTel2IP]
For example:
AltRouteCauseTel2IP 0 = 486; (Busy Here)
AltRouteCauseTel2IP 1 = 480; (Temporarily Unavailable)
AltRouteCauseTel2IP 2 = 408; (No Response)
Note: For a detailed description of this table, see 'Alternative Routing
Based on SIP Responses' on page 269.
Reasons for Alternative IP-to-Tel Routing Table
Web: Reasons for Alternative This table parameter configures call failure reason values received
IP-to-Tel Routing from the Tel side . If a call is released as a result of one of these
EMS: Alt Route Cause IP to reasons, the device attempts to locate an alternative Hunt Group for
Tel the call in the IP to Hunt Group Routing Table.
[AltRouteCauseIP2Tel] The format of this parameter is as follows:
[AltRouteCauseIP2Tel]
FORMAT AltRouteCauseIP2Tel_Index =
AltRouteCauseIP2Tel_ReleaseCause;
[\AltRouteCauseIP2Tel]
For example:
AltRouteCauseIP2Tel 0 = 3 (No Route to Destination)
AltRouteCauseIP2Tel 1 = 1 (Unallocated Number)
AltRouteCauseIP2Tel 2 = 17 (Busy Here)
AltRouteCauseIP2Tel 2 = 27 (Destination Out of Order)
to Trunk upon Q.931 Call Release Cause Code' on page 271.

Forward On Busy Trunk Destination Table

Web/EMS: Forward On Busy This table parameter configures the Forward On Busy Trunk
Trunk Destination Destination table. This table allows you to define an alternative IP
[ForwardOnBusyTrunkDest] destination if a trunk is busy for IP-to-Tel calls.
[ForwardOnBusyTrunkDest]
FORMAT ForwardOnBusyTrunkDest_Index =
ForwardOnBusyTrunkDest_TrunkGroupId,
ForwardOnBusyTrunkDest_ForwardDestination;
[\ForwardOnBusyTrunkDest]
For example, the below configuration forwards IP-to-Tel calls to
destination user “112” at host IP address 10.13.4.12, port 5060, using
transport protocol TCP, if Trunk Group ID 2 is unavailable:
ForwardOnBusyTrunkDest 1 = 2,
[email protected]:5060;transport=tcp;
to IP Destination upon Busy Trunk' on page 272.
44.11.13 Number Manipulation Parameters

The number manipulation parameters are described in the table below.
Table 44-62: Number Manipulation Parameters
Web: Copy Destination Number Determines whether the device copies the called number to the
to Redirect Number outgoing SIP Diversion header for Tel-to-IP calls. Therefore, the
EMS: Copy Dest to Redirect called number is used as a redirect number. Call redirection
Number information is typically used for Unified Messaging and voice mail
[CopyDest2RedirectNumber] services to identify the recipient of a message.
 [0] Don't copy = (Default) Disable.
 [1] Copy after phone number manipulation = Copies the called
number after manipulation. The device first performs Tel-to-IP
destination phone number manipulation (i.e., on the SIP To
header), and only then copies the manipulated called number to
the SIP Diversion header for the Tel-to-IP call. Therefore, with
this option, the called and redirect numbers are identical.
 [2] Copy before phone number manipulation = Copies the called
number before manipulation. The device first copies the original
called number to the SIP Diversion header, and then performs
Tel-to-IP destination phone number manipulation. Therefore,
this allows you to have different numbers for the called (i.e., SIP
To header) and redirect (i.e., SIP Diversion header) numbers.
Web/EMS: Add Hunt Group ID Determines whether the Hunt Group ID is added as a prefix to the
as Prefix destination phone number (i.e., called number) for Tel-to-IP calls.
[AddTrunkGroupAsPrefix]  [0] No = (Default) Don't add Hunt Group ID as prefix.
 [1] Yes = Add Hunt Group ID as prefix to called number.
Notes:
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 This option can be used to define various routing rules.
 To use this feature, you must configure the Hunt Group IDs (see
Configuring Endpoint Phone Numbers on page 235).
Web: Add Trunk ID as Prefix Determines whether or not the port number is added as a prefix to
EMS: Add Port ID As Prefix the called (destination) number for Tel-to-IP calls.
[AddPortAsPrefix]  [0] No (Default)
 [1] Yes
If enabled, the device adds the following prefix to the called phone
number: port number (single digit in the range 1 to 8for 8-port
devices, two digits in the range 01 to 24 for MP-124).
This option can be used to define various routing rules.
Web/EMS: Add Trunk Group ID Determines whether the device adds the Hunt Group ID (from
as Prefix to Source where the call originated) as the prefix to the calling number (i.e.
[AddTrunkGroupAsPrefixToS source number).
ource]  [0] No (default)
 [1] Yes
Web: IP to Tel Remove Routing Determines whether or not the device removes the prefix (as
Table Prefix configured in the IP to Hunt Group Routing Table - see 'Configuring
EMS: Remove Prefix IP to Hunt Group Routing Table' on page 263) from the destination
[RemovePrefix] number for IP-to-Tel calls, before sending it to the Tel.
 [1] Yes
For example: To route an incoming IP-to-Tel call with destination
number "21100", the IP to Hunt Group Routing Table is scanned
for a matching prefix. If such a prefix is found (e.g., "21"), then
before the call is routed to the corresponding Hunt Group, the
prefix "21" is removed from the original number, and therefore, only
"100" remains.
Notes:
 This parameter is applicable only if number manipulation is
performed after call routing for IP-to-Tel calls (i.e.,
RouteModeIP2Tel parameter is set to 0).
 Similar operation (of removing the prefix) is also achieved by
using the usual number manipulation rules.
[SwapTel2IPCalled&CallingNu Determines whether the device swaps the calling and called
mbers] numbers received from the Tel side (for Tel-to-IP calls). The SIP
INVITE message contains the swapped numbers.
 [0] = (Default) Disabled
 [1] = Swap calling and called numbers
Web: Add Number Plan and Determines whether the TON/PLAN parameters are included in the
Type to RPI Header Remote-Party-ID (RPID) header.
EMS: Add Ton 2 RPI  [0] No
[AddTON2RPI]  [1] Yes (default)
If the Remote-Party-ID header is enabled (EnableRPIHeader = 1)
and AddTON2RPI = 1, it's possible to configure the calling and
called number type and number plan using the Number
Manipulation tables for Tel-to-IP calls.
Web/EMS: Source Manipulation Determines the SIP headers containing the source number after
Mode

[SourceManipulationMode] manipulation:
 [0] = (Default) The SIP From and P-Asserted-Identity headers
contain the source number after manipulation.
 [1] = Only SIP From header contains the source number after
manipulation, while the P-Asserted-Identity header contains the
source number before manipulation.
Calling Name Manipulations IP-to-Tel Table
Configures rules for manipulating the calling name (caller ID) in the
[CallingNameMapIp2Tel] received SIP message for IP-to-Tel calls. This can include
modifying or removing the calling name. The format of this table ini
file parameter is as follows:
[ CallingNameMapIp2Tel ]
FORMAT CallingNameMapIp2Tel_Index =
CallingNameMapIp2Tel_DestinationPrefix,
CallingNameMapIp2Tel_SourcePrefix,
CallingNameMapIp2Tel_CallingNamePrefix,
CallingNameMapIp2Tel_SourceAddress,
CallingNameMapIp2Tel_RemoveFromLeft,
CallingNameMapIp2Tel_RemoveFromRight,
CallingNameMapIp2Tel_LeaveFromRight,
CallingNameMapIp2Tel_Prefix2Add,
CallingNameMapIp2Tel_Suffix2Add;
[ \CallingNameMapIp2Tel ]
Note: For a detailed description of this table, see 'Configuring SIP
Calling Name Manipulation' on page 248.
Calling Name Manipulations Tel-to-IP Table
This table parameter configures rules for manipulating the calling
[CallingNameMapTel2Ip] name (caller ID) for Tel-to-IP calls. This can include modifying or
removing the calling name.
[ CallingNameMapTel2Ip ]
FORMAT CallingNameMapTel2Ip_Index =
CallingNameMapTel2Ip_DestinationPrefix,
CallingNameMapTel2Ip_SourcePrefix,
CallingNameMapTel2Ip_CallingNamePrefix,
CallingNameMapTel2Ip_SrcTrunkGroupID,
CallingNameMapTel2Ip_SrcIPGroupID,
CallingNameMapTel2Ip_RemoveFromLeft,
CallingNameMapTel2Ip_RemoveFromRight,
CallingNameMapTel2Ip_LeaveFromRight,
CallingNameMapTel2Ip_Prefix2Add,
CallingNameMapTel2Ip_Suffix2Add;
[ \CallingNameMapTel2Ip ]
Note: For a detailed description of this table, see 'Configuring SIP
Calling Name Manipulation' on page 248.
Destination Phone Number Manipulation for IP-to-Tel Calls Table
Web: Destination Phone This table parameter manipulates the destination number of IP-to-
Number Manipulation Table for Tel calls. The format of this parameter is as follows:
IP > Tel Calls [NumberMapIp2Tel]
EMS: SIP Manipulations > FORMAT NumberMapIp2Tel_Index =
Destination IP to Telcom NumberMapIp2Tel_DestinationPrefix,
[NumberMapIP2Tel] NumberMapIp2Tel_SourcePrefix,
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NumberMapIp2Tel_SourceAddress,
NumberMapIp2Tel_NumberType, NumberMapIp2Tel_NumberPlan,
NumberMapIp2Tel_RemoveFromLeft,
NumberMapIp2Tel_RemoveFromRight,
NumberMapIp2Tel_LeaveFromRight,
NumberMapIp2Tel_Prefix2Add, NumberMapIp2Tel_Suffix2Add,
NumberMapIp2Tel_IsPresentationRestricted;
[\NumberMapIp2Tel]
For example:
NumberMapIp2Tel 0 = 03,22,$$,$$,$$,2,667,$$,$$;
EMS: Perform Additional Enables additional destination number manipulation for IP-to-Tel
IP2TEL Destination calls. The additional manipulation is done on the initially
Manipulation manipulated destination number, and this additional rule is also
[PerformAdditionalIP2TELDes configured in the manipulation table (NumberMapIP2Tel
tinationManipulation] parameter). This enables you to configure only a few manipulation
rules for complex number manipulation requirements (that
generally require many rules).
 [1] = Enable
Destination Phone Number Manipulation for Tel-to-IP Calls Table
Web: Destination Phone This table parameter manipulates the destination number of Tel-to-
Number Manipulation Table for IP calls. The format of this parameter is as follows:
Tel > IP Calls [NumberMapTel2Ip]
EMS: SIP Manipulations > FORMAT NumberMapTel2Ip_Index =
Destination Telcom to IPs NumberMapTel2Ip_DestinationPrefix,
[NumberMapTel2IP] NumberMapTel2Ip_SourcePrefix,
NumberMapTel2Ip_SourceAddress,
NumberMapTel2Ip_NumberType, NumberMapTel2Ip_NumberPlan,
NumberMapTel2Ip_RemoveFromLeft,
NumberMapTel2Ip_RemoveFromRight,
NumberMapTel2Ip_LeaveFromRight,
NumberMapTel2Ip_Prefix2Add, NumberMapTel2Ip_Suffix2Add,
NumberMapTel2Ip_IsPresentationRestricted,
NumberMapTel2Ip_SrcTrunkGroupID, NumberMapTel2Ip_
SrcIPGroupID;
[\NumberMapTel2Ip]
For example:
NumberMapTel2Ip 0 = 01,$$,*,0,0,2,$$,$$,971,$$,$$,$$,$$;
NumberMapTel2Ip 1 = 10,10,*,255,255,3,0,5,100,$$,255,$$,$$;
Source Phone Number Manipulation for IP-to-Tel Calls Table
Web: Source Phone Number This parameter table manipulates the source number for IP-to-Tel
Manipulation Table for IP > Tel calls. The format of this parameter is as follows:
Calls [SourceNumberMapIp2Tel]
EMS: SIP Manipulations > FORMAT SourceNumberMapIp2Tel_Index =
Source IP to Telcom SourceNumberMapIp2Tel_DestinationPrefix,
[SourceNumberMapIP2Tel] SourceNumberMapIp2Tel_SourcePrefix,
SourceNumberMapIp2Tel_SourceAddress,
SourceNumberMapIp2Tel_NumberType,
SourceNumberMapIp2Tel_NumberPlan,

SourceNumberMapIp2Tel_RemoveFromLeft,
SourceNumberMapIp2Tel_RemoveFromRight,
SourceNumberMapIp2Tel_LeaveFromRight,
SourceNumberMapIp2Tel_Prefix2Add,
SourceNumberMapIp2Tel_Suffix2Add,
SourceNumberMapIp2Tel_IsPresentationRestricted;
[\SourceNumberMapIp2Tel]
For example:
SourceNumberMapIp2Tel 0 = 22,03,$$,$$,$$,$$,2,667,$$,$$;
SourceNumberMapIp2Tel 1 =
034,01,1.1.1.1,$$,0,2,$$,$$,972,$$,10;
EMS: Perform Additional Enables additional source number manipulation for IP-to-Tel calls.
IP2TEL Source Manipulation The additional manipulation is done on the initially manipulated
[PerformAdditionalIP2TELSo source number, and this additional rule is also configured in the
urceManipulation] manipulation table (SourceNumberMapIP2Tel parameter). This
enables you to configure only a few manipulation rules for complex
number manipulation requirements (that generally require many
rules).
 [1] = Enable
Source Phone Number Manipulation for Tel-to-IP Calls Table
Web: Source Phone Number This table parameter manipulates the source phone number for
Manipulation Table for Tel > IP Tel-to-IP calls. The format of this parameter is as follows:
Calls [SourceNumberMapTel2Ip]
EMS: SIP Manipulations > FORMAT SourceNumberMapTel2Ip_Index =
Source Telcom to IP SourceNumberMapTel2Ip_DestinationPrefix,
[SourceNumberMapTel2IP] SourceNumberMapTel2Ip_SourcePrefix,
SourceNumberMapTel2Ip_SourceAddress,
SourceNumberMapTel2Ip_NumberType,
SourceNumberMapTel2Ip_NumberPlan,
SourceNumberMapTel2Ip_RemoveFromLeft,
SourceNumberMapTel2Ip_RemoveFromRight,
SourceNumberMapTel2Ip_LeaveFromRight,
SourceNumberMapTel2Ip_Prefix2Add,
SourceNumberMapTel2Ip_Suffix2Add,
SourceNumberMapTel2Ip_IsPresentationRestricted,
NumberMapTel2Ip_SrcTrunkGroupID,
NumberMapTel2Ip_SrcIPGroupID;
[\SourceNumberMapTel2Ip]
For example:
SourceNumberMapTel2Ip 0 = 22,03,$$,0,0,$$,2,$$,667,$$,0,$$,$$;
SourceNumberMapTel2Ip 0 =
10,10,*,255,255,3,0,5,100,$$,255,$$,$$;
Redirect Number Tel-to-IP Table
Web: Redirect Number Tel -> IP This table parameter manipulates the Redirect Number for Tel-to-
EMS: Redirect Number Map Tel IP calls. The format of this parameter is as follows:
to IP [RedirectNumberMapTel2Ip]
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[RedirectNumberMapTel2IP] FORMAT RedirectNumberMapTel2Ip_Index =
RedirectNumberMapTel2Ip_DestinationPrefix,
RedirectNumberMapTel2Ip_RedirectPrefix,
RedirectNumberMapTel2Ip_RemoveFromLeft,
RedirectNumberMapTel2Ip_RemoveFromRight,
RedirectNumberMapTel2Ip_LeaveFromRight,
RedirectNumberMapTel2Ip_Prefix2Add,
RedirectNumberMapTel2Ip_Suffix2Add,
RedirectNumberMapTel2Ip_IsPresentationRestricted,
RedirectNumberMapTel2Ip_SrcTrunkGroupID,
RedirectNumberMapTel2Ip_SrcIPGroupID;
[\RedirectNumberMapTel2Ip]
For example:
RedirectNumberMapTel2Ip 1 = *, *, 4, 0, 255, , , 255, -1, -1;
Note: For a description of this table, see 'Configuring Redirect
Number Manipulation' on page 251.
Phone Context Table
Web: Phone Context Table This table parameter configures the Phone Context table. This
EMS: SIP Manipulations > parameter maps NPI and TON to the SIP 'phone-context'
Phone Context parameter, and vice versa.
[PhoneContext] The format for this parameter is as follows:
[PhoneContext]
FORMAT PhoneContext_Index = PhoneContext_Npi,
PhoneContext_Ton, PhoneContext_Context;
[\PhoneContext]
For example:
PhoneContext 0 = 0,0,unknown.com
PhoneContext 1 = 1,1,host.com
PhoneContext 2 = 9,1,na.e164.host.com
Note: For a detailed description of this table, see 'Mapping
NPI/TON to SIP Phone-Context' on page 253.
Web/EMS: Add Phone Context Determines whether the received Phone-Context parameter is
As Prefix added as a prefix to the outgoing Called and Calling numbers.
[AddPhoneContextAsPrefix]  [0] Disable (default)
 [1] Enable
44.12 Least Cost Routing Parameters

The Least Cost Routing (LCR) parameters are described in the table below.
Table 44-63: LCR Parameters
Web: Routing Rule Groups This table parameter enables the LCR feature and configures the
Table average call duration and default call cost. The default call cost
[RoutingRuleGroups] determines whether routing rules that are not configured with a Cost
Group are considered as a higher or lower cost route compared to
other matching routing rules that are assigned Cost Groups.
[ RoutingRuleGroups ]
FORMAT RoutingRuleGroups_Index =
RoutingRuleGroups_LCREnable,

RoutingRuleGroups_LCRAverageCallLength,
RoutingRuleGroups_LCRDefaultCost;
[ \RoutingRuleGroups ]
Note: For a detailed description of this table, see 'Enabling LCR and
Configuring Default LCR' on page 197.
Web: Cost Group Table This table parameter configures the Cost Groups for LCR, where each
EMS: Cost Group Cost Group is configured with a name, fixed call connection charge,
Provisioning > Cost Group and a call rate (charge per minute).
[CostGroupTable] [ CostGroupTable ]
FORMAT CostGroupTable_Index =
CostGroupTable_CostGroupName,
CostGroupTable_DefaultConnectionCost,
CostGroupTable_DefaultMinuteCost;
[ \CostGroupTable ]
For example: CostGroupTable 2 = "Local Calls", 2, 1;
Note: For a detailed description of this table, see 'Configuring Cost
Web: Cost Group > Time This table parameter configures time bands and associates them with
Band Table Cost Groups.
EMS: Time Band [CostGroupTimebands]
Provisioning > Time Band FORMAT CostGroupTimebands_TimebandIndex =
[CostGroupTimebands] CostGroupTimebands_StartTime, CostGroupTimebands_EndTime,
CostGroupTimebands_ConnectionCost,
CostGroupTimebands_MinuteCost;
[\CostGroupTimebands]
Note: For a detailed description of this table, see 'Configuring Time
Bands for Cost Groups' on page 200.
44.13 Standalone Survivability Parameters

The Stand-alone Survivability (SAS) parameters are described in the table below.
Table 44-64: SAS Parameters
Web: Enable SAS Enables the Stand-Alone Survivability (SAS) feature.

EMS: Enable  [0] Disable (default)
[EnableSAS]  [1] Enable
When enabled, the device receives the registration requests
from different SIP entities in the local network and then forwards
them to the defined proxy. If the connection to the proxy fails
('Emergency Mode'), the device serves as a proxy by allowing
calls internal to the local network or outgoing to PSTN.
required.
Web: SAS Local SIP UDP Port Defines the local UDP port for sending and receiving SIP
EMS: Local SIP UDP messages for SAS. The SIP entities in the local network need to
[SASLocalSIPUDPPort] send the registration requests to this port. When forwarding the
requests to the proxy ('Normal Mode'), this port serves as the
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source port.
Web: SAS Default Gateway IP Defines the Default Gateway used in SAS 'Emergency Mode'.
EMS: Default Gateway IP When an incoming SIP INVITE is received and the destination
[SASDefaultGatewayIP] Address-Of-Record is not included in the SAS database, the
request is immediately sent to this default gateway.
The address can be configured as an IP address (dotted-
decimal notation) or as a domain name (up to 49 characters).
You can also configure the IP address with a destination port,
e.g., "10.1.2.3:5060". The default is a null string, i.e., the local IP
address of the gateway.
Web: SAS Registration Time Defines the value of the SIP Expires header that is sent in a 200
EMS: Registration Time OK response to an incoming REGISTER message when in SAS
[SASRegistrationTime] 'Emergency Mode'.
The valid range is 0 (Analog) to 2,000,000. The default is 20.
Web: SAS Local SIP TCP Port Defines the local TCP port used to send/receive SIP messages
EMS: Local SIP TCP Port for the SAS application. The SIP entities in the local network
[SASLocalSIPTCPPort] need to send the registration requests to this port. When
forwarding the requests to the proxy ('Normal Mode'), this port
serves as the source port.
Web: SAS Local SIP TLS Port Defines the local TLS port used to send/receive SIP messages
EMS: Local SIP TLS Port for the SAS application. The SIP entities in the local network
[SASLocalSIPTLSPort] need to send the registration requests to this port. When
forwarding the requests to the proxy ('Normal Mode'), this port
serves as the source port.
Web: SAS Connection Reuse Enables the re-use of the same TCP connection for sessions
[SASConnectionReuse] with the same user in the SAS application.
 [0] Disable
The device can use the same TCP connection for multiple SIP
requests / responses for a specific SIP UA. The benefits of this
feature include less CPU and memory usage because fewer
TCP connections are open and reduced network congestion.
For example, assume the following:
 User A sends a REGISTER message to SAS with
transport=TCP.
 User B sends an INVITE message to A using SAS.
In this scenario, the SAS application forwards the INVITE
request using the TCP connection that User A initially opened
with the REGISTER message.
Web/EMS: Enable Record-Route Determines whether the device's SAS application adds the SIP
[SASEnableRecordRoute] Record-Route header to SIP requests. This ensures that SIP
messages traverse the device's SAS agent by including the
SAS IP address in the Record-Route header.
 [1] Enable
The Record-Route header is inserted in a request by a SAS
proxy to force future requests in the dialog session to be routed

through the SAS agent. Each traversed proxy in the path can
insert this header, causing all future dialogs in the session to
pass through it as well.
When this feature is enabled, the SIP Record-Route header
includes the URI "lr" parameter, indicating loose routing, for
example:
Record-Route: <sip:server10.biloxi.com;lr>
Web: SAS Proxy Set Defines the Proxy Set (index number) used in SAS Normal
EMS: Proxy Set mode to forward REGISTER and INVITE requests from users
[SASProxySet] that are served by the SAS application.
The valid range is 0 to 5. The default is 0 (i.e., default Proxy
Set).
Web: Redundant SAS Proxy Set Defines the Proxy Set (index number) used in SAS Emergency
EMS: Redundant Proxy Set mode for fallback when the user is not found in the Registered
[RedundantSASProxySet] Users database. Each time a new SIP request arrives, the SAS
application checks whether the user is listed in the registration
database. If the user is located in the database, the request is
sent to the user. If the user is not found, the request is
forwarded to the next redundant SAS defined in the Redundant
SAS Proxy Set. If that SAS Proxy IP appears in the Via header
of the request, it is not forwarded (thereby, preventing loops in
the request's course). If no such redundant SAS exists, the SAS
sends the request to its default gateway (configured by the
parameter SASDefaultGatewayIP).
The valid range is -1 to 5. The default is -1 (i.e., no redundant
Proxy Set).
Web/EMS: SAS Block Determines whether the device rejects SIP INVITE requests
Unregistered Users received from unregistered SAS users. This applies to SAS
[SASBlockUnRegUsers] Normal and Emergency modes.
 [0] Un-Block = (Default) Allow INVITE from unregistered
SAS users.
 [1] Block = Reject dialog-establishment requests from un-
registered SAS users.
Enables the device to change the SIP Contact header so that it
[SASEnableContactReplace] points to the SAS host and therefore, the top-most SIP Via
header and the Contact header point to the same host.
 [0] (default) = Disable - when relaying requests, the SAS
agent adds a new Via header (with the SAS IP address) as
the top-most Via header and retains the original Contact
header. Thus, the top-most Via header and the Contact
header point to different hosts.
 [1] = Enable - the device changes the Contact header so
that it points to the SAS host and therefore, the top-most Via
header and the Contact header point to the same host.
Note: Operating in this mode causes all incoming dialog
requests to traverse the SAS, which may cause load problems.
Web: SAS Survivability Mode Determines the Survivability mode used by the SAS application.
EMS: Survivability Mode  [0] Standard = (Default) Incoming INVITE and REGISTER
[SASSurvivabilityMode] requests are forwarded to the defined Proxy list of
SASProxySet in Normal mode and handled by the SAS
application in Emergency mode.
Version 6.6 641 MP-11x & MP-124

MP-11x & MP-124
 [1] Always Emergency = The SAS application does not use
Keep-Alive messages towards the SASProxySet, instead it
always operates in Emergency mode (as if no Proxy in the
SASProxySet is available).
 [2] Ignore Register = Use regular SAS Normal/Emergency
logic (same as option [0]), but when in Normal mode
incoming REGISTER requests are ignored.
 [3] Auto-answer REGISTER = When in Normal mode, the
device responds to received REGISTER requests by
sending a SIP 200 OK (instead of relaying the registration
requests to a Proxy), and enters the registrations in its SAS
database.
 [4] Use Routing Table only in Normal mode = The device
uses the IP-to-IP Routing table to route IP-to-IP SAS calls
only when in SAS Normal mode (and is unavailable when
SAS is in Emergency mode). This allows routing of SAS IP-
to-IP calls to different destinations (and not only to the SAS
Proxy Set).
Web: SAS Subscribe Response Defines the SIP response upon receipt of a SUBSCRIBE
[SASSubscribeResponse] message when SAS is in Emergency mode. For example, if this
parameter is set to "200", then SAS sends a SIP 200 OK in
response to a SUBSCRIBE message, when in Emergency
mode.
Web: Enable ENUM Enables SAS to perform ENUM (E.164 number to URI mapping)
[SASEnableENUM] queries when receiving INVITE messages in SAS emergency
mode.
 [1] Enable
Web: SAS Binding Mode Determines the SAS application database binding mode.
EMS: Binding Mode  [0] URI = (Default) If the incoming AoR in the INVITE
[SASBindingMode] requests is using a ‘tel:’ URI or ‘user=phone’ is defined, the
binding is performed according to the user part of the URI
only. Otherwise, the binding is according to the entire URI,
i.e., User@Host.
 [1] User Part only = The binding is always performed
according to the User Part only.
Web: SAS Emergency Numbers Defines emergency numbers for the device's SAS application.
[SASEmergencyNumbers] When the device's SAS agent receives a SIP INVITE (from an
IP phone) that includes one of the emergency numbers (in the
SIP user part), it forwards the INVITE to the default gateway
(configured by the parameter SASDefaultGatewayIP), i.e., the
device itself, which sends the call directly to the PSTN. This is
important for routing emergency numbers such as 911 (in North
America) directly to the PSTN. This is applicable to SAS
operating in Normal and Emergency modes.
Up to four emergency numbers can be defined, where each
number can be up to four digits.
Defines a prefix that is added to the Request-URI user part of
[SASEmergencyPrefix] the INVITE message that is sent by the device's SAS agent
when in Emergency mode to the default gateway or to any other
destination (using the IP-to-IP Routing table). This parameter is
required to differentiate between normal SAS calls routed to the

default gateway and emergency SAS calls. Therefore, this
allows you to define different manipulation rules for normal and
emergency calls.
This valid value is a character string. The default is an empty ""
string.
Web: SAS Entering Emergency Determines for which sent SIP message types the device enters
Mode SAS Emergency mode if no response is received for them from
[SASEnteringEmergencyMode] the proxy server.
 [0] = (Default) SAS enters Emergency mode only if no
response is received from sent SIP OPTIONS messages.
 [1] = SAS enters Emergency mode if no response is
received from sent SIP OPTIONS, INVITE, or REGISTER
messages.
Note: If the keep-alive mechanism is disabled for the Proxy Set
(in the Proxy Set table) and this parameter is set to [1], SAS
enters Emergency mode only if no response is received from
sent INVITE or REGISTER messages.
sas-indialog-mode Defines how the device sends incoming SIP dialog requests
[SASInDialogRequestMode] received from users when not in SAS Emergency mode.
 [0] = (Default) Send according to the SIP Request-URI.
 [1] = Send to Proxy server.
Web: SAS Inbound Manipulation Enables destination number manipulation of incoming INVITE
Mode messages when SAS is in Emergency mode. The manipulation
[SASInboundManipulationMode] rule is done in the IP to IP Inbound Manipulation table.
 [1] Emergency Only
Notes:
 Inbound manipulation applies only to INVITE requests.
 For more information on SAS inbound manipulation, see
'Manipulating Destination Number of Incoming INVITE' on
page 346.
SAS Registration Manipulation Table
Web: SAS Registration This table parameter configures the SAS Registration
Manipulation Manipulation table. This table is used by the SAS application to
EMS: Stand-Alone Survivability manipulate the SIP Request-URI user part of incoming INVITE
[SASRegistrationManipulation] messages and of incoming REGISTER request AoR (To
header), before saving it to the registered users database. The
format of this table parameter is as follows:
[SASRegistrationManipulation]
FORMAT SASRegistrationManipulation_Index =
SASRegistrationManipulation_RemoveFromRight,
SASRegistrationManipulation_LeaveFromRight,
SASRegistrationManipulation_RuleApplyTo;
[\SASRegistrationManipulation]
For example, the manipulation rule below routes an INVITE with
Request-URI header "sip:[email protected]" to user
"[email protected]" (i.e., keep only four digits from right of
user part):
SASRegistrationManipulation 0 = 0, 4, 2;
Note: For a detailed description of this table, see 'Manipulating
Version 6.6 643 MP-11x & MP-124

MP-11x & MP-124
URI user part of Incoming REGISTER' on page 345.
Web: SAS IP-to-IP Routing Table
[IP2IPRouting] This table parameter configures the IP-to-IP Routing table for
SAS routing rules. The format of this parameter is as follows:
[IP2IPRouting]
FORMAT IP2IPRouting_Index = IP2IPRouting_SrcIPGroupID,
IP2IPRouting_SrcUsernamePrefix, IP2IPRouting_SrcHost,
IP2IPRouting_DestUsernamePrefix, IP2IPRouting_DestHost,
IP2IPRouting_DestType, IP2IPRouting_DestIPGroupID,
IP2IPRouting_DestSRDID, IP2IPRouting_DestAddress,
IP2IPRouting_DestPort, IP2IPRouting_DestTransportType,
IP2IPRouting_AltRouteOptions;
[\IP2IPRouting]
For example:
IP2IPRouting 1 = -1, *, *, *, *, 0, -1, -1, , 0, -1, 0;
Note: For a detailed description of this table parameter, see
'SAS Routing Based on IP-to-IP Routing Table' on page 349.
44.14 Auxiliary and Configuration File Name Parameters

The configuration files (i.e., auxiliary files) can be loaded to the device using the Web
interface or a TFTP session. For loading these files using the ini file, you need to configure
these files in the ini file and configured whether they must be stored in the non-volatile
memory. The table below lists the ini file parameters associated with these auxiliary files.
For more information on the auxiliary files, see 'Loading Auxiliary Files' on page 369.
Table 44-65: Auxiliary and Configuration File Parameters
General Parameters
[SetDefaultOnIniFileProcess] Determines if all the device's parameters are set to their defaults
before processing the updated ini file.
 [0] = Disable - parameters not included in the downloaded ini file
are not returned to default settings (i.e., retain their current
settings).
 [1] = Enable (default).
Note: This parameter is applicable only for automatic HTTP update
or Web ini file upload (not applicable if the ini file is loaded using
BootP).
[SaveConfiguration] Determines if the device's configuration (parameters and files) is
saved to flash (non-volatile memory).
 [0] = Configuration isn't saved to flash memory.
 [1] = (Default) Configuration is saved to flash memory.
Auxiliary and Configuration File Name Parameters

Web/EMS: Call Progress Tones Defines the name of the file containing the Call Progress Tones
File definitions. For more information on how to create and load this file,
[CallProgressTonesFilename] refer to DConvert Utility User's Guide.

Web/EMS: Prerecorded Tones Defines the name (and path) of the file containing the Prerecorded
File Tones.
[PrerecordedTonesFileName] Notes:
 PRT is not supported by MP-124 Rev. E.
Web: Dial Plan File Defines the name (and path) of the Dial Plan file. This file should be
EMS: Dial Plan File Name created using AudioCodes DConvert utility (refer to DConvert Utility
[DialPlanFileName] User's Guide).
[UserInfoFileName] Defines the name (and path) of the file containing the User
Information data.
Version 6.6 645 MP-11x & MP-124

MP-11x & MP-124
44.15 Automatic Update Parameters

The automatic update of software and configuration files parameters are described in the
table below.
Table 44-66: Automatic Update of Software and Configuration Files Parameters
General Automatic Update Parameters

[AutoUpdateCmpFile] Enables the Automatic Update mechanism for the cmp file.
 [0] = (Default) The Automatic Update mechanism doesn't apply
to the cmp file.
 [1] = The Automatic Update mechanism includes the cmp file.
[AutoUpdateFrequency] Defines the number of minutes that the device waits between
automatic updates. The default is 0 (i.e., the update at fixed intervals
mechanism is disabled).
[AutoUpdatePredefinedTime] Defines schedules (time of day) for automatic updates. The format
of this parameter is: 'HH:MM', where HH denotes the hour and MM
the minutes, for example, 20:18.
Notes:
 The actual update time is randomized by five minutes to reduce
the load on the Web servers.
[AupdHttpUserAgent] Defines the information sent in the HTTP User-Agent header in the
HTTP Get requests sent by the device to the provisioning server for
the Automatic Update mechanism.
The valid value is a string of up to 511 characters. The information
can include any user-defined string or the following string variable
tags (case-sensitive):
 <NAME>: product name, according to the installed Software
License Key
 <MAC>: device's MAC address
 <VER>: software version currently installed on the device, e.g.,
"7.00.200.001"
 <CONF>: configuration version, as configured by the ini file
parameter, INIFileVersion or CLI command, configuration-version
The device automatically populates these tag variables with actual
values in the sent header. By default, the device sends the following
in the User-Agent header:
User-Agent: Mozilla/4.0 (compatible; AudioCodes;
<NAME>;<VER>;<MAC>;<CONF>)
For example, if you set AupdHttpUserAgent = MyWorld-
<NAME>;<VER>(<MAC>), the device sends the following User-
Agent header:
User-Agent: MyWorld-
Mediant;7.00.200.001(00908F1DD0D3)
Notes:
 The variable tags are case-sensitive.
 If you configure the parameter with the <CONF> variable tag,

you must reset the device with a burn-to-flash for your settings to
take effect.
 The tags can be defined in any order.
 The tags must be defined adjacent to one another (i.e., no
spaces).
EMS: AUPD Verify Certificates Determines whether the Automatic Update mechanism verifies
[AUPDVerifyCertificates] server certificates when using HTTPS.
 [1] = Enable
[AUPDCheckIfIniChanged] Determines whether the Automatic Update mechanism performs
CRC checking to determine if the ini file has changed prior to
processing.
 [0] = (Default) Do not check CRC. The ini file is loaded whenever
the server provides it.
 [1] = Check CRC for the entire file. Any change, including line
order, causes the ini file to be re-processed.
 [2] = Check CRC for individual lines. Use this option when the
HTTP server scrambles the order of lines in the provided ini file.
[ResetNow] Invokes an immediate device reset. This option can be used to
activate offline (i.e., not on-the-fly) parameters that are loaded using
the parameter IniFileUrl.
 [0] = (Default) The immediate restart mechanism is disabled.
 [1] = The device immediately resets after an ini file with this
parameter set to 1 is loaded.
Software/Configuration File URL Path for Automatic Update Parameters
[CmpFileURL] Defines the name of the cmp file and the path to the server (IP
address or FQDN) from where the device can load the cmp file and
update itself. The cmp file can be loaded using HTTP/HTTPS, FTP,
FTPS, or NFS.
For example: http://192.168.0.1/filename
Notes:
 When this parameter is configured, the device always loads the
cmp file after it is reset.
 The cmp file is validated before it's burned to flash. The
checksum of the cmp file is also compared to the previously
burnt checksum to avoid unnecessary resets.
 The maximum length of the URL address is 255 characters.
[IniFileURL] Defines the name of the ini file and the path to the server (IP
address or FQDN) on which it is located. The ini file can be loaded
using HTTP/HTTPS, FTP, FTPS, or NFS.
For example:
http://192.168.0.1/filename
http://192.8.77.13/config<MAC>
https://<username>:<password>@<IP address>/<file name>
Notes:
 When using HTTP or HTTPS, the date and time of the ini file are
validated. Only more recently dated ini files are loaded.
Version 6.6 647 MP-11x & MP-124

MP-11x & MP-124
 The optional string <MAC> is replaced with the device's MAC
address. Therefore, the device requests an ini file name that
contains its MAC address. This option allows the loading of
specific configurations for specific devices.
[PrtFileURL] Defines the name of the Prerecorded Tones (PRT) file and the path
to the server (IP address or FQDN) on which it is located.
For example: http://server_name/file, https://server_name/file.
Notes:
 PRT is not supported by MP-124 Rev. E.
[CptFileURL] Defines the name of the CPT file and the path to the server (IP
address or FQDN) on which it is located.
For example: http://server_name/file, https://server_name/file.
Note: The maximum length of the URL address is 99 characters.
[TLSRootFileUrl] Defines the name of the TLS trusted root certificate file and the URL
from where it can be downloaded.
[TLSCertFileUrl] Defines the name of the TLS certificate file and the URL from where
it can be downloaded.
[TLSPkeyFileUrl] Defines the URL for downloading a TLS private key file using the
Automatic Update facility.
[UserInfoFileURL] Defines the name of the User Information file and the path to the
server (IP address or FQDN) on which it is located.
For example: http://server_name/file, https://server_name/file
Note: The maximum length of the URL address is 99 characters.

User's Manual 45. DSP Templates
45 DSP Templates
The tables below list the maximum supported channel capacity.
Notes:
• To select the DSP Template that you want to use on the device, see Configuring
DSP Templates on page 185.
• Installation and use of voice coders is subject to obtaining the appropriate license
and royalty payments.
• The number of channels refers to the maximum channel capacity of the device.
• The G.727 coder is currently not supported by MP-124 Rev. E.
• For additional DSP templates, contact your AudioCodes representative.
Table 45-1: Maximum Channel Capacity for MP-11x and MP-124 Rev. D
DSP Template
0 1
Maximum Channels
Default Default
Model SRTP Enabled SRTP Enabled
(no SRTP (no SRTP
MP-112 FXS/FXO 2 2 2 2
MP-114 FXS/FXO 4 3 3 3
MP-118 FXS/FXO 8 6 6 6
MP-124 Rev. D 24 18 18 18
Voice Coder
G.711 A/Mu-law PCM √ √ √ √
G.726 ADPCM √ √ √ √
G.727 ADPCM √ √ √ √
G.723.1 √ √ √ √
G.729 A, B √ √ √ √
EG.711 √ √ - -
G.722 - - √ √
Version 6.6 649 MP-11x & MP-124

MP-11x & MP-124
Table 45-2: Maximum Channel Capacity for MP-124 Rev. E
Maximum Channels
Voice Coder
Default (no SRTP) SRTP Enabled
G.711 A/Mu-law PCM 24 17
G.726 ADPCM 24 17
G.723.1 24 17
G.729 A, B 24 17
G.722 21 16

User's Manual 46. Selected Technical Specifications
46 Selected Technical Specifications

The main technical specifications of the MP-11x and MP-124 devices are listed in the table
below.
Notes:
• All specifications in this document are subject to change without prior notice.
• The compliance and regulatory information can be downloaded from AudioCodes
Web site at http://www.audiocodes.com/library.
Table 46-1: MediaPack Technical Specifications
Function Specification
Interfaces
Voice Ports  MP-112: 2 ports
 MP-114: 4 ports
 MP-118: 8 ports
 MP-124: 24 ports
Telephone Interfaces  MP-112: FXS, RJ-11
 MP-114 & MP-118: FXS, FXO or mixed FXS/FXO, RJ-11
 MP-124: FXS, 50-pin Telco
Lifeline Automatic cut through of a single analog line (FXS version only,
refers only for the middle column – 4/8 ports)
Network Interface 10/100Base-TX, RJ-45
Indicators Channel Status and activity LEDs
Voice, Fax, Modem
Voice over Packet G.168-2004 compliant Echo Cancellation, VAD, CNG, Dynamic
Capabilities programmable Jitter
Buffer, modem detection and auto switch to PCM
Voice Compression G.711, G.723.1, G.726 ADPCM, G.727 ADPCM, G.729A/B, G.722
Note: The G.727 is currently not supported by MP-124 Rev. E.
Fax over IP T.38 compliant
Group 3 fax relay up to 14.4 kbps with automatic switching to PCM or
ADPCM
3-Way Conference 3-Way conference with local mixing
Quality Enhancement DiffServ, TOS, 802.1 P/Q VLAN tagging, RTCP XR
IP Transport RTP/RTCP per IETF RFC 3550 and 3551, Multiplexing (aggregated
RTP streams of several channels for saving network bandwidth)
Stand Alone Survivability (SAS) Application
Max. Registered Users SAS ensures call continuity between LAN SIP clients upon
connectivity failure with IP Centrex services (e.g., WAN IP PBX).
Capacity
Registered Users 25
Version 6.6 651 MP-11x & MP-124

MP-11x & MP-124
Signaling
Signaling  MP-112: FXS Loop-start
 MP-114 & MP-118: FXS, FXO Loop-start
 MP-124: FXS Loop-start
In-band Signaling DTMF (TIA 464B)
User-defined and call progress tones
Out-of-Band Signaling DTMF Relay (RFC 2833), DTMF via SIP INFO
Control SIP (RFC 3261)
Provisioning
Protocols  BootP, DHCP, TFTP and HTTP for Automatic Installation
 DHCP options 66,67 in auto update mode
 Remote management using Web browser
 EMS (Element Management System) / SNMP V3
 Syslog support
 RS-232 for basic configuration (via CLI)
 Voice Menu using touch-tone phone (FXS interface) for basic
configuration
 TR-069
Security
Media SRTP
Control H.235, IPSec, TLS/SIPS
Management HTTPS, Access List, IPSec
Physical
Power Single universal power supply 100-240V 0.3A max. 50-60 Hz or -48V
DC
Note: -48V DC is supported only on MP-124D.
Environmental  Operational: 5 to 40°C (41 to 104°F)
 Storage: -25 to 85°C (-13 to 185°F)
 Humidity: 10 to 90% non-condensing
Dimensions  MP-112: 42 x 172 x 220 mm
 MP-114 & MP-118: 42 x 172 x 220 mm
 MP-124: 44 x 445 x 269 mm
Weight  MP-1xx: 0.5 kg (1.1 lbs.) approx.
 MP-124: 1.8 kg (4 lbs.)
Mounting Rack mount, Table top, Wall mount
Additional Features
Message Waiting Indication High (Neon) and Low (LED) Voltage, FSK, Stutter Dial Tone
High Availability PSTN Fallback: Support of PSTN fallback due to Power failure, if the
IP connection is down or due to customer defined IP QoS thresholds
Stand Alone Survivability (SAS): Supports SAS of up to 25 SIP
users (UA)
Ring voltage Sine: 54 V RMS typical (balanced ringing only)

User's Manual 46. Selected Technical Specifications
Ring Frequency 25-100Hz

Maximum Ringer Load Ringer Equivalency Number (REN) 3
Loop Impedance (including Up to 1500 ohm for the MP-11x, Up to 1600 ohm for the MP-124
phone impedance)
Off-hook Current  MP-11x: 20/25/32 mA (Note: For MP-118, 25/32 mA is supported
only on the first 4 ports)
 MP-124: 20 mA (minimum), 20/25/32 mA on first 4 ports
Lifeline Supported in all ports of Mixed FXS/FXO and in first port of MP-
114/FXS and MP-118/FXS using special Lifeline cable
Caller ID Bellcore GR-30-CORE Type 1 using Bell 202 FSK modulation, ETSI
Type 1, NTT, Denmark, India, Brazil, British and DTMF ETSI CID
(ETS 300-659-1)
Polarity Reversal / Wink Immediate or smooth to prevent erroneous ringing
Metering Tones 12/16 KHz sinusoidal bursts, Generation on FXS
Distinctive Ringing By frequency (15-100 Hz) and cadence patterns
Over-voltage protection and Routing of FXS telephony cables outdoors can be done only in
surge immunity conjunction with AudioCodes’ approved primary surge protector and
proper installation and grounding. When done correctly, the
installation will meet ITU-T K.21 (basic) standards.
Version 6.6 653 MP-11x & MP-124

International Headquarters
1 Hayarden Street,
Airport City
Lod 7019900, Israel
Tel: +972-3-976-4000
Fax: +972-3-976-4040
AudioCodes Inc.
27 World’s Fair Drive,
Somerset, NJ 08873
Tel: +1-732-469-0880
Fax: +1-732-469-2298
Contact us: www.audiocodes.com/contact

Website: www.audiocodes.com
©2017 AudioCodes Ltd. All rights reserved. AudioCodes, AC, HD VoIP, HD VoIP Sounds Better, IPmedia, Mediant,
MediaPack, What’s Inside Matters, OSN, SmartTAP, User Management Pack, VMAS, VoIPerfect, VoIPerfectHD, Your
Gateway To VoIP, 3GX, VocaNom, AudioCodes One Voice and CloudBond are trademarks or registered trademarks of
AudioCodes Limited. All other products or trademarks are property of their respective owners. Product specifications
are subject to change without notice.
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User's Manual

AudioCodes Analog Media Gateways

Getting Started with Initial Connectivity ................................................................25

Management Tools ..................................................................................................35

Version 6.6 3 MP-11x & MP-124

4.7 Configuring Web and Telnet Access List ............................................................... 75

General System Settings ......................................................................................109

User's Manual 4 Document #: LTRT-65437

11 Date and Time .................................................................................................. 119

General VoIP Configuration ..................................................................................121

Version 6.6 5 MP-11x & MP-124

14.1.2 Echo Cancellation ..................................................................................................166

User's Manual 6 Document #: LTRT-65437

19 Coders and Profiles ........................................................................................ 219

Gateway Application .............................................................................................231

Version 6.6 7 MP-11x & MP-124

User's Manual 8 Document #: LTRT-65437

Stand-Alone Survivability Application .................................................................327

Version 6.6 9 MP-11x & MP-124

32.1.2 Prerecorded Tones File .........................................................................................375

Status, Performance Monitoring and Reporting .................................................409

User's Manual 10 Document #: LTRT-65437

36 Carrier-Grade Alarms ...................................................................................... 413

Version 6.6 11 MP-11x & MP-124

42.2 Configuring DTMF Tones for Test Calls............................................................... 466

User's Manual 12 Document #: LTRT-65437

44.11.2 DTMF and Hook-Flash Parameters.......................................................................572

Version 6.6 13 MP-11x & MP-124

This page is intentionally left blank.

User's Manual 14 Document #: LTRT-65437

Abbreviations and Terminology

Version 6.6 15 MP-11x & MP-124

SIP CPE Release Notes

Notes and Warnings

User's Manual 16 Document #: LTRT-65437

Document Revision Record

Version 6.6 17 MP-11x & MP-124

65436  Updated with patch version 6.60A.340.001

User's Manual 18 Document #: LTRT-65437

Version 6.6 19 MP-11x & MP-124

1.1 MediaPack Models

Combined FXS / Number of

1.2 SIP Overview

User's Manual 20 Document #: LTRT-65437

 F1 INVITE - 10.8.201.108 to 10.8.201.161:

Version 6.6 21 MP-11x & MP-124

CSeq: 18153 INVITE

 F4 200 OK - 10.8.201.161 to 10.8.201.108:

User's Manual 22 Document #: LTRT-65437

 F6 BYE - 10.8.201.108 to 10.8.201.10:

Version 6.6 23 MP-11x & MP-124

This page is intentionally left blank.

User's Manual 24 Document #: LTRT-65437

2 Assigning the OAMP IP Address

IP Address  FXS and FXS / FXO devices: 10.1.10.10

2.1 Web Interface

 To assign an OAMP IP address using the Web interface:

Version 6.6 27 MP-11x & MP-124

User's Manual 28 Document #: LTRT-65437