Cisco IP Telephony CIPT

Authorized Self-Study Guide
Cisco IP Telephony (CIPT)

Second Edition
Jeremy D. Cioara, CCIE No. 11727
Cisco Press
800 East 96th Street
Indianapolis, IN 46240 USA
ii
Cisco IP Telephony (CIPT) (Authorized Self-Study Guide), Second Edition

Jeremy D. Cioara, CCIE No. 11727
Copyright© 2007 Cisco Systems, Inc.
Cisco Press logo is a trademark of Cisco Systems, Inc.
Published by:
Cisco Press
800 East 96th Street
Indianapolis, IN 46240 USA
All rights reserved. No part of this book may be reproduced or transmitted in any form or by any means, electronic or mechanical,
including photocopying, recording, or by any information storage and retrieval system, without written permission from the pub-
lisher, except for the inclusion of brief quotations in a review.
Printed in the United States of America 1 2 3 4 5 6 7 8 9 0
First Printing October 2006
Library of Congress Cataloging-in-Publication Number: 2006926698
ISBN: 1-58705-261-x
Warning and Disclaimer

This book is designed to provide information about the Cisco Unified CallManager 4.x platform as it relates to Cisco CIPT certifica-
tion exam 642-444. Every effort has been made to make this book as complete and as accurate as possible, but no warranty or fitness
is implied.
The information is provided on an “as is” basis. The authors, Cisco Press, and Cisco Systems, Inc., shall have neither liability nor
responsibility to any person or entity with respect to any loss or damages arising from the information contained in this book or from
the use of the discs or programs that may accompany it.
The opinions expressed in this book belong to the author and are not necessarily those of Cisco Systems, Inc.
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tion, please contact: U.S. Corporate and Government Sales 1-800-382-3419 [email protected]
For sales outside of the U.S. please contact: International Sales 1-317-581-3793 [email protected]
Trademark Acknowledgments
All terms mentioned in this book that are known to be trademarks or service marks have been appropriately capitalized. Cisco Press
or Cisco Systems, Inc., cannot attest to the accuracy of this information. Use of a term in this book should not be regarded as affect-
ing the validity of any trademark or service mark.
iii
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iv
About the Author

Jeremy D. Cioara, CCIE No. 11727, CCVP, MCSE, CNE, is the owner of AdTEC Networks and
works as a network consultant, instructor, and author. He has been working in network technologies
for more than a decade and has deployed networks worldwide. His current consulting work
focuses on network and voice-over-IP (VoIP) implementations. Jeremy has written many books
on Cisco network technology but has a true passion for educating individuals both in the classroom
and through e-learning environments. Jeremy currently lives with his wife in Phoenix, AZ.
v
About the Technical Reviewers

Michael Coffin, CCIE No. 9789, is a technical marketing engineer at Cisco Systems and currently
focuses his time on Cisco Mobility products. He began his career at Cisco in the Cisco Technical
Assistance Center, where he worked on troubleshooting large service provider core networks.
Then moved into the Cisco IP Communications Business Unit as a CallManager escalation
engineer. Mike has contributed to several design guides and white papers, worked on supporting
many large Cisco IP Telephony deployments, and often provides internal and external training
around the world. Mike holds CCIE certifications in Routing & Switching, Service Provider, and
Voice.
Dennis Hartmann, CCIE No. 15651, CCVP, CCIP, CCNP, MCSE, is a consultant for White Pine
Communications. He has been a technical instructor teaching Cisco classes for Global Knowledge
and providing network architecture work for customers. He teaches all of the classes associated
with the CCVP certification, as well as all classes associated with the CCNP, CCIP, and optical
specialist certifications.
Kevin Wallace, CCIE No. 7945, CCSI, CCVP, CCNP, CCDP, MCSE 4, CNE 4/5, is a full-time
instructor for Thomson NETg. With 17 years of Cisco internetworking experience, Kevin has been
a network design specialist for The Walt Disney World Resort and a network manager for Eastern
Kentucky University. Kevin holds a bachelor of science degree in electrical engineering from the
University of Kentucky. Among Kevin’s publication credits are Cisco Voice over IP (CVoice)
Authorized Self-Study Guide, Second Edition; Voice over IP First-Step; CCDA/CCDP Flash
Cards and Exam Practice Pack (coauthored with Anthony Sequeira); CCIE Routing and Switching
Flash Cards and Exam Practice Pack (coauthored with Anthony Sequeira); and Cisco IP
Telephony Flash Cards and Exam Practice Pack, all of which are available from Cisco Press.
Additionally, Kevin authored the Cisco Enterprise Voice over Data Design (EVoDD) 3.3 course,
was a contributing author for the Cisco IP Telephony Troubleshooting (IPTT) 2.0 course, and has
written for Packet magazine from Cisco Systems. Kevin also holds the IP Telephony Design
Specialist, IP Telephony Operations Specialist, and IP Telephony Support Specialist CQS
certifications.
vi
Dedications
First and foremost, I’d like to thank Jesus Christ who continues to bless me in every way and keeps
me from killing myself by doing something stupid. You will always be at the core of everything I
do. Second, to my darling wife, Susan: Thank you for your support through all these projects that
keep me glued to a computer screen through all hours of the day and night. You are a more
wonderful companion than I could have ever hoped for. I love you! To the cricket chirping in the
room right now: I hate you. If I ever find you, I’m going to feed you to my fish and dance merrily
around the room as they eat you slowly. To my cat, Snuggles: Thanks for being soft. To the dog,
Buttercup: I have nothing to say to you at this time, but at least I’ve acknowledged your existence.
To all my readers: Be warned; this is what happens to you after months of writing hundreds of
pages.
vii
Acknowledgments
I’d like to give special recognition to Kevin Wallace, Larry Roberts, Michael Coffin, and Dennis
Hartman for providing their expert technical knowledge in editing the book. As usual, they’re not
afraid to tell you when you’re wrong.
A big “thank you” goes out to the production team for this book. Brett Bartow and Andrew Cupp
have been incredibly professional and a pleasure to work with. I couldn’t have asked for a finer
team.
viii
Contents at a Glance
Foreword xxvi
Introduction xxvii
Part I: Cisco CallManager Fundamentals 3
Chapter 1 Introduction to Cisco Unified Communications and Cisco Unified

CallManager 5
Chapter 2 Cisco Unified CallManager Clustering and Deployment Options 17
Chapter 3 Cisco Unified CallManager Installation and Upgrades 35

Part II: IPT Devices and Users 51
Chapter 4 Cisco IP Phones and Other User Devices 53

Chapter 5 Configuring Cisco Unified CallManager to Support IP Phones 73
Chapter 6 Cisco IP Telephony Users 101
Chapter 7 Cisco Bulk Administration Tool 119

Part III: IPT Network Integration and Route Plan 143
Chapter 8 Cisco Catalyst Switches 145
Chapter 9 Configuring Cisco Gateways and Trunks 161
Chapter 10 Cisco Unified CallManager Route Plan Basics 187
Chapter 11 Cisco Unified CallManager Advanced Route Plans 209
Chapter 12 Configuring Hunt Groups and Call Coverage 235
Chapter 13 Implementing Telephony Call Restrictions and Control 253
Chapter 14 Implementing Multiple-Site Deployments 275

Part IV: VoIP Features 307
Chapter 15 Media Resources 309
Chapter 16 Configuring User Features, Part 1 347
Chapter 18 Configuring Cisco Unified CallManager Attendant Console 409
Chapter 19 Configuring Cisco IP Manager Assistant 437

ix
Part V: IPT Security 457
Chapter 20 Securing the Windows Operating System 459
Chapter 21 Securing Cisco Unified CallManager Administration 483
Chapter 22 Preventing Toll Fraud 503
Chapter 23 Hardening the IP Phone 523
Chapter 24 Understanding Cryptographic Fundamentals 535
Chapter 25 Understanding the Public Key Infrastructure 551
Chapter 26 Understanding Cisco IP Telephony Authentication and Encryption

Fundamentals 573
Chapter 27 Configuring Cisco IP Telephony Authentication and Encryption 597

Part VI: IP Video 617
Chapter 28 Introducing IP Video Telephony 619
Chapter 29 Configuring Cisco VT Advantage 647

Part VII: IPT Management 671
Chapter 30 Introducing Database Tools and Cisco Unified CallManager

Serviceability 673
Chapter 31 Monitoring Performance 695

Chapter 32 Configuring Alarms and Traces 713
Chapter 33 Configuring CAR 737
Chapter 34 Using Additional Management and Monitoring Tools 763
Part VIII: Appendix 789
Appendix A Answers to Review Questions 791
Index 832
x
Contents
Foreword xxvi
Introduction xxvii
Part I: Cisco CallManager Fundamentals 3
Chapter 1 Introduction to Cisco Unified Communications and Cisco Unified
CallManager 5
Cisco Unified Communications 6
Understanding Cisco Unified CallManager 7
Cisco Unified CallManager and IP Phone Interaction 8
The Components of Cisco Unified CallManager 10
Cisco Unified CallManager Servers 11
Summary 12
Review Questions 12
Chapter 2 Cisco Unified CallManager Clustering and Deployment Options 17
The Two Sides of the Cisco Unified CallManager Cluster 17
The SQL Database Cluster 17
Intracluster Run-Time Data 18
Cluster Redundancy Designs 19
1:1 Redundancy Design 19
2:1 Redundancy Design 21
Call-Processing Deployment Models 23
Single-Site Deployment 23
Multisite Deployment with Centralized Call Processing 25
Multisite Deployment with Distributed Call Processing 27
Clustering over the IP WAN 30
Summary 31
Review Questions 32
Chapter 3 Cisco Unified CallManager Installation and Upgrades 35
Cisco Unified CallManager 4.x Clean Installation Process 35
Installation Disks 35
Installation Configuration Data 36
Sample Configuration Data Worksheet 38
Postinstallation Procedures 40
Activating Cisco Unified CallManager Services 41
Upgrading Prior Cisco Unified CallManager Versions 45
Summary 47
Review Questions 47
xi
Part II: IPT Devices and Users 51

Chapter 4 Cisco IP Phones and Other User Devices 53
Cisco IP Phones 53
Entry-Level Cisco IP Phones 54
Midrange and Upper-End Cisco IP Phones 56
Additional Cisco IP Telephony Endpoints 59
IP Phone Startup Process 64
Cisco IP Phone Codec Support 67
Summary 68
Review Questions 68
Chapter 5 Configuring Cisco Unified CallManager to Support IP Phones 73
Configuring Intracluster IP Phone Communication 73
Removing DNS Reliance 73
Configuring Device Pools 75
Cisco Unified CallManager Group Configuration 79
Date/Time Group Configuration 80
Region Configuration 82
Softkey Template Configuration 83
IP Phone Button Templates 84
Putting It All Together 87
IP Phone Configuration 87
Manual IP Phone and Directory Number Configuration 87
Automatic IP Phone Registration 90
Case Study: Device Pool Design 92
Summary 95
Review Questions 96
Chapter 6 Cisco IP Telephony Users 101
Cisco CallManager User Database 101
Cisco CallManager User Configuration 102
Adding a User 102
Configuring Device Pools 104
User Logon and Device Configuration 105
CallManager User Options: Log On 105
CallManager User Options: Welcome 106
CallManager User Options: Call Forwarding 107
CallManager User Options: Speed Dials 108
CallManager User Options: IP Phone Services 110
CallManager User Options: Personal Address Book and Fast Dial 111
CallManager User Options: Message Waiting Lamp Policy 112
CallManager User Options: Locale Settings 113
Summary 115
Review Questions 115
xii
Chapter 7 Cisco Bulk Administration Tool 119

The Cisco Bulk Administration Tool 119
BAT Components 120
BAT Installation 121
Adding Cisco IP Phones Using BAT 123
Using BAT Wizard Step 1: Choosing Devices 123
Using BAT Wizard Step 2: Device Specifics 124
Using BAT Wizard Step 3: Proceed with Task 125
Adding Phones Step 1: Configuring BAT Templates 126
Adding Phones Step 2: Creating CSV Files 128
Adding Phones Step 3: Validating Phones 131
Adding Phones Step 4: Inserting Phones 132
Updating IP Phones Using BAT 134
Using the Tool for Auto-Registered Phone Support 136
Summary 139
Part III: IPT Network Integration and Route Plan 143
Chapter 8 Cisco Catalyst Switches 145
Catalyst Switch Role in IP Telephony 145
Powering the Cisco IP Phone 146
Two Types of PoE Delivery 146
PoE Device Detection 147
Catalyst Family of PoE Switches 148
Configuring PoE 150
Verifying PoE 151
Data and Voice VLANs 152
Configuring and Verifying Dual VLANs Using the CatOS 153
Configuring and Verifying Dual VLANs Using the NativeOS 153
Configuring Class of Service 155
Summary 156
Chapter 9 Configuring Cisco Gateways and Trunks 161
Cisco Gateway Concepts 161
Analog and Digital Gateways 161
Core Gateway Requirements 162
Gateway Communication Overview 163
Configuring Access Gateways 164
H.323 Gateway Configuration 165
Call Classification 167
MGCP Gateway Configuration 169
Non-IOS MGCP Gateway Configuration 172
xiii
Cisco Trunk Concepts 174

Configuring Intercluster Trunks 176
SIP and Cisco CallManager 179
SIP Components 179
CallManager SIP Integration and Configuration 181
Summary 183
Chapter 10 Cisco Unified CallManager Route Plan Basics 187
External Call Routing 187
Route Plan Configuration Process 189
Route Group Overview and Configuration 189
Route List Overview and Configuration 192
Route Pattern Concepts and Configuration 194
Route Pattern: Commonly Used Wildcards 197
Route Pattern Examples 199
Digit Analysis 200
Digit Collection 201
Closest Match Routing 201
Interdigit Timeout 202
Simple Route Plan Example 204
Summary 205
Chapter 11 Cisco Unified CallManager Advanced Route Plans 209
Route Filters 209
Route Filter Tags 211
Configuring Route Filters 212
Applying Route Filters 214
Practical Route Filter Example 215
Discard Digit Instructions 217
Transformation Masks 219
Calling-Party Transformations 220
Called-Party Transformations 221
Configuring Calling- and Called-Party Transformation Masks 223
Transformation Example 224
Translation Patterns 226
Translation Pattern Configuration 226
Practical Use of a Translation Pattern 227
Route Plan Report 229
Summary 230
xiv
Chapter 12 Configuring Hunt Groups and Call Coverage 235

Call Distribution Components 235
Line Groups 236
Call-Distribution Algorithms 237
Hunt Options 238
Call Distribution Scenarios: Top-Down Example 239
Hunting and Forwarding 240
Configuring Line Groups, Hunt Lists, and Hunt Pilots 241
Configuring Line Groups 242
Configuring Hunt Lists 243
Configuring Hunt Pilots 245
Summary 247
Chapter 13 Implementing Telephony Call Restrictions and Control 253
Class of Service Overview 253
Partitions and Calling Search Spaces Overview 254
Calling Search Spaces Applied to Gateways and Intercluster Trunks 256
Partition Configuration 256
Calling Search Space Configuration 258
Time-of-Day Routing Overview 260
Time Periods 260
Time Schedule 261
Time-of-Day Routing Effect on Users 262
Configuring Time-of-Day Routing 262
Configuring Time Periods 263
Configuring Time Schedules 265
Applying Time Schedules 266
Time-of-Day Routing Usage Scenario 268
Summary 270
Chapter 14 Implementing Multiple-Site Deployments 275
Call Admission Control 275
Locations-Based Call Admission Control 276
Configuring Location-Based Call Admission Control 278
AAR Overview 280
Configuring AAR 282
Gatekeeper-Based Call Admission Control 288
Gatekeeper Communication 289
Gatekeeper-Based Call Admission Control Configuration 293
Survivable Remote Site Telephony 296
SRST Router Configuration 298
SRST Reference Configuration 300
Summary 301
xv
Part IV: VoIP Features 307

Chapter 15 Media Resources 309
Introduction to Media Resources 309
Conference Bridge Resources 310
Conference Bridge Hardware 311
Conference Bridge Hardware CallManager Configuration 315
Media Termination Point Resources 316
MTP Configuration 318
Annunciator Resources 319
Annunciator Configuration 320
Transcoder Resources 321
Transcoder Configuration 323
Music on Hold Resources 324
Creating Audio Source Files 326
Configuring MoH 327
Step 1: Configure the Audio Translator 327
Step 2: Configure the MoH Server 329
Step 3: Add and Configure Audio Source Files 330
Step 4: Set MoH Servicewide Settings 331
Step 5: Find and Configure the Fixed Audio Source 332
Step 6: Assigning Audio Source IDs 334
Media Resource Management 335
Media Resource Design 335
Media Resource Groups 336
Media Resource Group Configuration 337
Media Resource Group List Configuration 338
Assigning Media Resource Group Lists 339
MRGL Design Strategy: Grouping Resources by Type 340
MRGL Design Strategy: Grouping Resources by Location 340
Summary 342
Basic IP Phone Features 347
Speed Dial and Abbreviated Dial Configuration 348
Auto Answer Configuration 349
Call Forward and Configurable Call Forward Display Configuration 349
Softkey Templates 351
Adding Softkey Templates 352
Modifying Softkey Availability and Positioning 353
Assigning Softkey Templates to Devices 354
Deleting Softkey Templates 355
Enhanced IP Phone Features 355
Multiple Calls per Line Appearance 355
Direct Transfer 357
xvi
Call Join 358

Immediate Divert to Voice Mail 358
Call Park 359
Call Pickup and Group Call Pickup 360
Other Group Call Pickup 361
Call Pickup Configuration 361
AutoCall Pickup 362
Cisco Call Back 364
Barge and Privacy 365
Configuring Barge 366
Configuring Privacy 367
IP Phone Services 368
Configuring IP Phone Services 369
Services Phone Display Examples 372
Summary 373
Cisco CallManager Extension Mobility 377
Extension Mobility Example 377
Configuring Extension Mobility 378
Cisco CallManager Extension Mobility Service Parameters 381
Configuring a Default Device Profile 382
Configuring User Device Profiles 383
Associating Users with Device Profiles 384
Creating the Extension Mobility Service 386
Updating Cisco IP Phones to Add Extension Mobility Support 387
Client Matter Codes and Forced Authentication Codes 389
FAC Concepts 389
CMC Concepts 391
FAC and CMC Configuration 392
Creating CMC Codes 392
Creating FAC Codes 393
Enabling Route Patterns for FAC and CMC 394
Call Display Restrictions 395
Understanding Calling Line ID Presentation 396
Understanding Connected Line ID Presentation 397
Understanding Ignore Presentation Indicators 397
Configuring Caller ID Restrictions 398
Malicious Call Identification 399
Configuring MCID 399
Configuring CallManager to Support CDRs 400
Configuring MCID Alarms 401
Adding the MCID Softkey 402
xvii
Multilevel Precedence and Preemption 403

Summary 405
Chapter 18 Configuring Cisco Unified CallManager Attendant Console 409
Introduction to Cisco CallManager Attendant Console 409
Terms and Definitions 411
The Telephony Call Dispatcher and Attendant Console Directory 411
Cisco Telephony Call Dispatcher 411
Cisco CallManager Attendant Console Directory 412
Pilot Points and Hunt Groups 413
Call Routing and Call Queuing 414
Call Routing 414
Call Queuing 415
Server and Administration Configuration 416
Adding Attendant Console Users 416
Adding the “ac” User 417
Pilot Point Configuration 419
Hunt Group Configuration 420
Activating Cisco TCD and CTIManager Services 423
Using the Attendant Console Configuration Tool 424
Installing and Configuring the Attendant Console Client 425
Cisco Attendant Console Features 426
Attendant Console GUI: Call Control 428
Attendant Console GUI: Custom Speed Dials 429
Attendant Console GUI: Directory Lookup 430
Attendant Console GUI: Parked Calls 430
Attendant Console GUI: Broadcast Calls 431
Summary 432
Chapter 19 Configuring Cisco IP Manager Assistant 437
Cisco IP Manager Assistant Overview 437
Cisco IPMA with Proxy-Line Support 438
Cisco IPMA with Shared-Line Support 438
Cisco IP Manager Assistant Architecture 439
Configuring Cisco IPMA for Shared-Line Support 440
IPMA Step 1: Activating the Cisco IPMA Service 441
IPMA Step 2: Configuring Service Parameters 442
IPMA Step 3: Restarting the Cisco IPMA Service 444
IPMA Step 4: Configuring Managers and Assistants 444
IPMA Step 5: Configuring Call Divert Target 449
IPMA Step 6: Installing the IPMA Assistant Console 450
Summary 453
xviii
Part V: IPT Security 457

Chapter 20 Securing the Windows Operating System 459
Threats Targeting the Operating System 459
Lowering the Threats in Windows Operating System 460
Security and Hot Fix Policy 461
Operating System Hardening 463
IP Telephony Operating System Security Scripts 464
File-Sharing Considerations 467
Antivirus Protection 467
Cisco Security Agent 468
Cisco Security Headless Agent 469
Cisco Security Managed Agent 469
CSA Supported Applications 470
CSA Protection 471
CSA Guidelines 471
Administrator Password Policy 472
Account and Password Considerations 474
Common Windows Exploits 475
Security Taboos 476
Summary 478
Chapter 21 Securing Cisco Unified CallManager Administration 483
Threats Targeting Remote Administration 483
Securing CallManager Communications Using HTTPS 483
HTTPS Certificates 484
Accessing CallManager When Using Self-Signed Certificates 485
Multilevel Administration 487
Enabling MLA 488
MLA Functional Groups 489
MLA User Groups 490
Assigning MLA Access Privileges 491
Creating New MLA Functional and User Groups 492
Summary 499
Chapter 22 Preventing Toll Fraud 503
Toll Fraud Exploits 503
Preventing Call Forward and Voice-Mail Toll Fraud Using Calling Search Spaces 505
Blocking Commonly Exploited Area Codes 507
Using Time-of-Day Routing 509
Using FAC and CMC 510
xix
Restricting External Transfers 511

Defining OnNet and OffNet 512
Configuring Call Transfer Restrictions 514
Call Transfer Restriction Example 515
Dropping Conference Calls 516
Summary 517
Chapter 23 Hardening the IP Phone 523
Threats Targeting Endpoints 523
Blocking Endpoint Attacks 525
Stopping Rogue Images from Entering IP Phones 525
Disabling Phone Settings in Cisco CallManager Administration 526
Disabling PC Port and Settings Access 526
Disabling IP Phone Web Access 527
Ignoring Gratuitous ARP 528
Disabling Voice VLAN Access 529
Enabling IP Phone Encryption and Authentication 530
Summary 531
Chapter 24 Understanding Cryptographic Fundamentals 535
What Is Cryptography? 535
Authentication and Encryption 537
Symmetric Encryption 538
Symmetric Encryption Example: AES 539
Asymmetric Encryption 540
Symmetric Encryption Example: RSA 541
Hash Functions 541
Digital Signatures 544
Digital Signatures and RSA 545
Summary 546
Chapter 25 Understanding the Public Key Infrastructure 551
The Need for a PKI 551
Key Exchange in Symmetric Cryptography 551
Key Exchange Protected by Asymmetric Encryption 552
The Pitfall of Asymmetric Key Exchange 553
PKI as a Trusted Third-Party Protocol 553
PKI Entities 556
X.509v3 Certificates 557
Self-Signed Certificates 558
PKI Enrollment 559
Man-in-the-Middle PKI Enrollment Attack 560
Secure PKI Enrollment 560
xx
PKI Revocation and Key Storage 561

PKI Revocation Methods 562
Key Storage 562
Smart Cards and Smart Tokens 563
PKI Example 563
PKI and SSL/TLS 563
Web Server Certificate Exchange 564
Summary 568
Chapter 26 Understanding Cisco IP Telephony Authentication and Encryption
Fundamentals 573
Threats Targeting the IP Telephony System 573
How CallManager Protects Against Threats 574
Secure Signaling and Media Transfer 574
Authentication of Phone Images 575
Authentication of Phone Configuration Files 576
PKI Topologies in Cisco IP Telephony 576
Self-Signed Certificates PKI Topology 577
Manufacturing Installed Certificates PKI Topology 577
Locally Significant Certificates PKI Topology 578
Independent, Separated PKI Topology 579
CTL Client 580
CTL Client Application 581
CTL Verification on the IP Phone 582
Initial Deployment Issue 582
PKI Enrollment in Cisco IP Telephony 583
CAPF Acting as a CA 583
CAPF Acting as a Proxy to an External CA 584
Keys and Certificate Storage in Cisco IP Telephony 585
Authentication and Integrity 585
Certificate Exchange in TLS 586
Server-to-Phone Authentication 586
Phone-to-Server Authentication 587
TLS SHA-1 Session Key Exchange 588
Encryption 588
TLS AES Encryption 589
SRTP Media Encryption 589
SRTP Packet Format 590
Secure Call Flow Summary 591
Summary 591
xxi
Chapter 27 Configuring Cisco IP Telephony Authentication and Encryption 597

Authentication and Encryption Configuration Overview 597
Enabling Services Required for Security 599
Using the CTL Client 600
Installing the CTL Client 600
Working with Locally Significant Certificates 602
Issuing a Phone Certificate Using an Authentication String 605
Issuing a Phone Certificate Using the CAPF 607
Configuring the Device Security Mode 607
Negotiating Device Security Mode 608
Generating a CAPF Report 609
Summary 611
Part VI: IP Video 617
Chapter 28 Introducing IP Video Telephony 619
IP Video Telephony Solution Components 619
Video Call Concepts 621
Cisco CallManager Involvement in Video Calls 621
Video Call Flow 622
Video Codecs Supported by Cisco CallManager 623
Video Protocols Supported in Cisco CallManager 624
SCCP Video Call Characteristics 626
H.323 Video Call Characteristics 627
SCCP and H.323 in Cisco CallManager 628
Bandwidth Management 628
Video Call Bandwidth Requirement 629
Calculating the Total Bandwidth 630
Actual Bandwidth Used Per Video Call 630
Call Admission Control in Cisco CallManager 631
Call Admission Control Within a Cluster 632
Region Configuration 634
Location Configuration 635
Call Admission Control Example 637
Retry Video Call as Audio 638
Call Admission Control Between Clusters 640
Gatekeeper Call Admission Control Options 641
Gatekeeper Call Admission Control Example 642
Summary 643
xxii
Chapter 29 Configuring Cisco VT Advantage 647

Cisco VT Advantage Overview 647
Cisco VT Advantage Components 647
Cisco VT Advantage Supported Standards 648
Protocols Used by Cisco VT Advantage 649
How Calls Work with Cisco VT Advantage 651
Cisco VT Advantage Video Modes 652
Configuring Cisco CallManager for Video 653
Call-Routing Considerations 654
VT Advantage Deployment Tool 655
Configuring Cisco IP Phones for Cisco VT Advantage 657
Verifying Phone Loads 657
Configuring IP Phones to Support Video 658
Verification of Phone Configuration 661
Installing Cisco VT Advantage on a Client 661
Cisco VT Advantage Hardware and Software Requirements 661
Cisco VT Advantage Installation Preparation Checklist 662
Installing Cisco VT Advantage 663
Cisco VT Advantage Installation Verification 665
Summary 667
Part VII: IPT Management 671
Chapter 30 Introducing Database Tools and Cisco Unified CallManager
Serviceability 673
Database Management Tools 673
Database Management Tools Overview 674
Microsoft SQL Server 2000 Enterprise Manager 674
DBLHelper 677
Cisco CallManager Serviceability Overview 678
Serviceability Components 678
Alarm 679
Trace 680
Tools 681
Application 683
Help 684
Control Center 684
Service Activation 686
Tools Overview 688
Summary 690
xxiii
Chapter 31 Monitoring Performance 695

Performance Counters 695
Performance Analysis 696
Microsoft Event Viewer 697
Microsoft Performance Monitor 699
Microsoft Performance Monitor Report View 699
Microsoft Performance Monitor Graph and Histogram View 700
Microsoft Windows Task Manager 702
Real-Time Monitoring Tool Overview 702
Real-Time Monitoring Tool Configuration Profiles 704
Using the Real-Time Monitoring Tool Effectively 705
Viewing Call Activity 706
Using Device Search 707
Visiting Alert Central 707
Summary 708
Chapter 32 Configuring Alarms and Traces 713
Alarm Overview 713
Alarm Configuration 714
Configuring Alarms 715
Configuring Alarms for Java Applications 716
Alarm Details 717
Trace Configuration 719
Types of Traces 720
Trace Configuration and Analysis Overview 720
Trace Configuration 721
Enabling Clusterwide Trace Settings 723
Trace Analysis 725
Trace Collection 728
The Trace Collection Tool 729
Bulk Trace Analysis 730
Bulk Trace Analysis Features 730
Using Bulk Trace Analysis 731
Additional Trace Tools 732
Summary 733
Chapter 33 Configuring CAR 737
CAR Overview 737
CDRs and CMRs 737
CAR Users 739
CAR Report Types and User Levels 740
CAR Configuration 740
xxiv
Initial Configuration 741

CAR System Parameter Menu Configuration 742
Administration Rights and Mail Parameters 743
Dial Plan Configuration 744
Gateway and System Preferences 746
Report Scheduling 748
Managing the CDR Load 749
Configuring the Schedulers 750
System Database Configuration 752
CDR and CAR Database Alerts 753
Database Purges 754
User Report Configuration 756
Viewing the Report 757
Sending the Report 757
Summary 758
Chapter 34 Using Additional Management and Monitoring Tools 763
Remote Management Tools 763
CiscoWorks ITEM Overview 763
Simple Network Management Protocol 764
SNMP Basics 765
SNMP Configuration on Cisco CallManager 766
Syslog Overview 769
Syslog Configuration in Cisco CallManager 769
Dependency Records 771
Enabling Dependency Records 771
Accessing Dependency Records 772
Password Changer Tool 774
Using the Password Changer Tool 775
Cisco Dialed Number Analyzer 776
Installing Cisco Dialed Number Analyzer 777
Using the Cisco Dialed Number Analyzer 778
Quality Report Tool 780
Activating the QRT Softkey 781
Viewing QRT Logs 782
Summary 785
Part VIII: Appendix 789
Appendix A Answers to Review Questions 791
Index 832
xxv
Icons Used in This Book
Si
IP
Communication Headphones SIP Server PC Multilayer Switch Access ISDN/Frame Relay
Server with Text Server Switch
Directory Terminal File Web Ciscoworks ATM Modem

Server Browser Workstation Switch
Camera Laptop Cisco IP IBM Front End Cluster Multilayer

PC/Video Communicator Mainframe Processor Controller Switch
Gateway Router Bridge Branch Office Cisco Unity Cisco CallManager Catalyst
Switch
Headquarters Cisco Cisco IP Network Cloud Ethernet Serial Switched

IP Phone Contact Center Serial
Command Syntax Conventions

The conventions used to present command syntax in this book are the same conventions used in
the IOS Command Reference. The Command Reference describes these conventions as follows:
■ Boldface indicates commands and keywords that are entered literally as shown. In actual
configuration examples and output (not general command syntax), boldface indicates
commands that are manually input by the user (such as a show command).
■ Italic indicates arguments for which you supply actual values.
■ Vertical bars (|) separate alternative, mutually exclusive elements.
■ Square brackets [ ] indicate optional elements.
■ Braces { } indicate a required choice.
■ Braces within brackets [{ }] indicate a required choice within an optional element.
xxvi
Foreword
Cisco IP Telephony (CIPT), Second Edition is an excellent self-study resource for the CCVP CIPT
exam. Whether you are studying to become CCVP certified or are simply seeking to gain a better
understanding of VoIP and PSTN components and technologies, you will benefit from the
information presented in this book.
Cisco Press Self-Study Guide titles are designed to help educate, develop, and grow the
community of Cisco networking professionals. As an early-stage exam preparation product, this
book presents a detailed and comprehensive introduction to the technologies used to install,
configure, and support Cisco CallManager 4.1 in a Cisco network, including such features as
security and video. Developed in conjunction with the Cisco certifications team, Cisco Press
books are the only self-study books authorized by Cisco Systems.
Most networking professionals use a variety of learning methods to gain necessary skills. Cisco
Press self-study titles are a prime source of content for some individuals and can also serve as an
excellent supplement to other forms of learning. Training classes, whether delivered in a
classroom or on the Internet, are a great way to quickly acquire new understanding. Hands-on
practice is essential for anyone seeking to build, or hone, new skills. Authorized Cisco training
classes, labs, and simulations are available exclusively from Cisco Learning Solutions Partners
worldwide. Please visit http://www.cisco.com/go/training to learn more about Cisco Learning
Solutions Partners.
I hope and expect that you’ll find this guide to be an essential part of your exam preparation and
a valuable addition to your personal library.
Don Field
Director, Certifications
Cisco Systems, Inc.
August 2006
xxvii
Introduction
Professional certifications have been an important part of the computing industry for many years
and will continue to become more important. Many reasons exist for these certifications, but the
most popularly cited reason is that of credibility. All other considerations held equal, the certified
employee/consultant/job candidate is considered more valuable than one who is not.
Goals and Methods

The most important and somewhat obvious goal of this book is to help you pass the Cisco IP
Telephony (CIPT) exam (642-444). In fact, if the primary objective of this book was different, then
the book’s title would be misleading; however, the methods used in this book to help you pass the
CCVP Cisco IP Telephony exam are designed to also make you much more knowledgeable about
how to do your job. Although this book has more than enough questions to help you prepare for
the actual exam, the method in which they are used is not to simply make you memorize as many
questions and answers as you possibly can.
One key methodology used in this book is to help you discover the exam topics that you need to
review in more depth, to help you fully understand and remember those details, and to help you
prove to yourself that you have retained your knowledge of those topics. So, this book does not try
to help you pass by memorization, but helps you truly learn and understand the topics. The Cisco
IP Telephony exam is just one of the foundation topics in the CCVP certification and the
knowledge contained within is vitally important to consider yourself a truly skilled voice engineer
or specialist. This book would do you a disservice if it didn’t attempt to help you learn the material.
Who Should Read This Book?

This book is not designed to be a general networking topics book, although it can be used for that
purpose. This book is intended to tremendously increase your chances of passing the CCVP Cisco
IP Telephony exam. Although you will achieve other objectives from using this book, the book is
written with one goal in mind: to help you pass the exam.
The placement of CallManager in the CCVP series of exams is unique in that it could be one of
the very first CCVP exams you take, or it could be one of the last. It stands as an isolated product
in the CCVP series of exams. This means you could be getting into this book with an extensive
knowledge of voice gateways, switches, and quality of service (from reading the CVOICE and/or
QoS books) with a basic CCNA-level knowledge. Regardless, your learning curve will be the
same; CallManager is configured differently, troubleshot uniquely, and managed distinctively
from the rest of the voice network.
So why should you want to pass the CCVP Cisco IP Telephony exam? Because it is one of the
milestones toward getting the CCVP certification; no small feat in itself. What would getting the
xxviii
CCVP mean to you? A raise, a promotion, recognition? How about to enhance your résumé? To
demonstrate that you are serious about continuing the learning process and that you are not content
to rest on your laurels. To please your reseller-employer, who needs more certified employees for
a higher discount from Cisco. Or one of many other reasons.
How This Book Is Organized

Although this book could be read from cover to cover, it is designed to be flexible and allow you
to easily move between chapters and sections of chapters to cover just the material that you need
more work with. If you do intend to read them all, the order in the book is an excellent sequence
to use.
The chapters cover the following topics:
Chapter 1, “Introduction to Cisco Unified Communications and Cisco Unified

CallManager”—This chapter provides an overview of the Cisco Unified Communications
(AVVID) strategy, how the Cisco Unified CallManager product suite fits into the scheme, the
interaction between CallManager and Cisco IP Phones, and the CallManager server platforms.
Chapter 2, “Cisco Unified CallManager Clustering and Deployment Options”—This chapter

discusses the design strategies behind a Cisco CallManager cluster, cluster replication, and
CallManager deployment models.
Chapter 3, “Cisco Unified CallManager Installation and Upgrades”—This chapter covers the
requirements to perform a CallManager server installation, the CallManager installation and
upgrade process, and postinstallation procedures.
Chapter 4, “Cisco IP Phones and Other User Devices”—This chapter examines the basic
features of all Cisco IP Phones, the entry-level, midrange, and upper-end Cisco IP Phone models,
the IP Phone startup process, and audio codec communication.
Chapter 5, “Configuring Cisco Unified CallManager to Support IP Phones”—This chapter

examines the configuration of Cisco CallManager to support IP Phone registration and
communication within a cluster, the creation of device pools, and manually or automatically
registering Cisco IP Phones with the SQL Database.
Chapter 6, “Cisco IP Telephony Users”—This chapter covers the addition of IP telephony users
to the CallManager LDAP database, associating users with devices, and allowing users to access
the User Options web page to configure common features.
xxix
Chapter 7, “Cisco Bulk Administration Tool”—This chapter covers the features and
components of the BAT application, the installation of BAT, and the configuration of BAT to apply
bulk moves, adds, and changes to the Cisco CallManager cluster.
Chapter 8, “Cisco Catalyst Switches”—This chapter examines the functions that Catalyst
switches perform in a Cisco IP telephony solution, the three options for powering Cisco IP Phones,
the two types of power supplied by PoE switches, inline power configuration, dual VLAN
configuration, and CoS configuration.
Chapter 9, “Configuring Cisco Gateways and Trunks”—This chapter discusses the role of the
gateway in the IP telephony infrastructure, core gateway requirements, and gateway
communication protocols; it also examines the configuration of analog and digital gateways, Cisco
CallManager trunk configuration, and enabling the CallManager to use SIP capabilities.
Chapter 10, “Cisco Unified CallManager Route Plan Basics”—This chapter covers the
fundamentals of the Cisco CallManager route plan, including the identification of the route plan
building blocks, the configuration of route groups, route lists, and route patterns, and the design
of a basic route plan.
Chapter 11, “Cisco Unified CallManager Advanced Route Plans”—This chapter covers the
concepts and configurations behind route filters, digit discard instructions, transformation masks,
translation patterns, and route plan reports.
Chapter 12, “Configuring Hunt Groups and Call Coverage”—This chapter examines the call
distribution components and algorithms supported by CallManager; examines call hunting
concepts; discusses the concepts and configurations of line groups, hunt lists, and hint pilots; and
provides a scenario-based design discussion for hunting and forwarding calls.
Chapter 13, “Implementing Telephony Call Restrictions and Control”—This chapter

discusses the concepts behind class of service as it relates to users of a phone system; it also covers
the design and configuration of partitions, calling search spaces, and time-of-day routing.
Chapter 14, “Implementing Multiple-Site Deployments”—This chapter discusses why call

admission control is important to maintain voice QoS across an IP WAN; describes the Cisco
CallManager locations feature and how it provides the necessary call admission control for
centralized call-processing environments; and explains the concepts and configurations behind
locations, H.323 gatekeepers, and SRST.
Chapter 15, “Media Resources”—This chapter examines the necessary Cisco CallManager
resources you should use as media resources, the configuration of conference bridges, media
termination points, transcoders, and music on hold.
xxx
Chapter 16, “Configuring User Features, Part 1”—This chapter discusses the core Cisco IP
Phone features supported by Cisco CallManager along with the enhanced IP Phone features
configurable by an administrator; this chapter also examines the configuration of softkey
templates, call park, call pickup, call back, barge, shared line appearances, and IP Phone services.
Chapter 17, “Configuring User Features, Part 2”—This chapter covers the configuration of the
Cisco CallManager Extension Mobility service, FAC and CMC call accounting and restrictions,
call display restrictions, malicious caller ID, and multilevel precedence and preemption.
Chapter 18, “Configuring Cisco Unified CallManager Attendant Console”—This chapter

explains the functions and features of the Cisco CallManager Attendant Console application, and
defines the key Attendant Console components and redundancy process. This chapter also covers
the configuration of the Cisco CallManager server to support the Attendant Console and the client-
side Attendant Console installation and configuration.
Chapter 19, “Configuring Cisco IP Manager Assistant”—This chapter covers the features of
Cisco IPMA and the two modes of operation, the IPMA components, and the configuration of
Cisco IPMA for shared-line mode installations.
Chapter 20, “Securing the Windows Operating System”—This chapter examines the security
threats to the Windows operating system, the Cisco security and hotfix policy, the included
enhanced security scripts, and the antivirus protection options for the CallManager server. This
chapter also covers the suggested administrative password policy, the protection of the server from
common exploits, the Cisco Security Agent, and the not-recommended security settings for the
Cisco CallManager server.
Chapter 21, “Securing Cisco Unified CallManager Administration”—This chapter discusses

the threats targeting remote Cisco CallManager Administration and other applications, the
function of HTTPS in remote communication, the HTTP certificate operations, and the
CallManager multilevel administration configuration.
Chapter 22, “Preventing Toll Fraud”—This chapter examines the vulnerability of legitimate
devices to be used for fraudulent use in the IP telephony network, the use of partitions and calling
search spaces to restrict call forwarding, the blocking of specific area codes, the configuration of
CallManager to route calls based on time-of-day, the implementation of FAC to implement user
authorization, and the restriction of conference call features to limit toll fraud.
Chapter 23, “Hardening the IP Phone”—This chapter covers the potential threats against IP
Phones and the attack tools and methods a hacker can use, signed firmware images, CallManager
IP Phone security techniques, and IP Phone authentication and encryption.
xxxi
Chapter 24, “Understanding Cryptographic Fundamentals”—This chapter discusses the

foundations of cryptography and the four cryptographic services, basic operation and uses for
symmetric and asymmetric encryption algorithms, hashing, and digital signatures.
Chapter 25, “Understanding the Public Key Infrastructure”—This chapter examines the
problem of secure, scalable distribution of public keys, the concepts behind a trusted introducer,
certificates, CAs, certification paths, certificate enrollment, and certificate revocation.
Chapter 26, “Understanding Cisco IP Telephony Authentication and Encryption

Fundamentals”—This chapter explains how file manipulation, tampering with call signaling,
man-in-the-middle attacks, eavesdropping, and IP Phone theft can compromise a Cisco
CallManager system. This chapter also discusses the authentication and encryption mechanisms
supported by CallManager, the roll of CAPF, MICs, LSCs, CTLs, and the CTL Client. This
chapter covers the processes and protocols used for signaling encryption and media encryption.
Chapter 27, “Configuring Cisco IP Telephony Authentication and Encryption”—This

chapter examines the steps to configure a Cisco CallManager system for authentication and
encryption, the activation of the CTL Provider and CAPF services, the installation of the CTL
Client, and configuring the IP Phones for a hardened security structure.
Chapter 28, “Introducing IP Video Telephony”—This chapter covers the functions and
components of the Cisco IP video telephony solution, the connection of a video call, H.323 versus
SCCP video signaling, the two factors in determining the bandwidth requirement for video, and
video call admission control.
Chapter 29, “Configuring Cisco VT Advantage”—This chapter discusses the features and
function of the VT Advantage, placing and receiving calls with VT Advantage, configuring the
Cisco CallManager to support video, and the VT Advantage client installation process.
Chapter 30, “Introducing Database Tools and Cisco Unified CallManager Serviceability”—
This chapter examines the database structure and replication status of broken database connection,
the services provided by CallManager Serviceability, the CallManager Control Center, the
CallManager Service Activation window, and the various tools used to monitor Cisco
CallManager listed by function.
Chapter 31, “Monitoring Performance”—This chapter defines performance objects, covers the
Microsoft Event Viewer and Performance Monitor, and the Cisco Real-Time Monitoring Tool.
Chapter 32, “Configuring Alarms and Traces”—This chapter identifies the functions of the
Cisco CallManager Alarm interface, and discusses the configuration of alarms and traces, the trace
analysis process, and the Bulk Trace Analysis tool.
xxxii
Chapter 33, “Configuring CAR”—This chapter discusses the usage, features, and operations of
the CAR tool; the contents of CDR and CMR records; the three levels of CAR users and their
reporting capabilities; the configuration of CAR system parameters, system schedule, and alerts;
and the generation of user reports using CAR.
Chapter 34, “Using Additional Management and Monitoring Tools”—This chapter examines
the use of SNMP, Syslog, and CiscoWorks ITEM in remotely managing and maintaining a Cisco
CallManager system, the use of dependency records, the Password Changer tool, the Dialed
Number Analyzer, and the Quality Reporting tool.
Finally, Appendix A, “Answers to Review Questions,” contains the solutions to the review
questions throughout the book.
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Part I: Cisco CallManager
Fundamentals
Chapter 1 Introduction to Cisco Unified Communications and

Cisco Unified CallManager
Chapter 2 Cisco Unified CallManager Clustering and Deployment Options
Chapter 3 Cisco Unified CallManager Installation and Upgrades

This chapter covers the
following topics:
■ An introduction to the Cisco Unified

Communications strategy (formerly Cisco
AVVID) and Cisco Unified CallManager
technology
■ An understanding of the placement of Cisco

Unified CallManager in an IP telephony
design
■ The interaction between Cisco Unified

CallManager and Cisco IP Phones
■ Cisco Unified CallManager server platforms

CHAPTER 1
Introduction to Cisco Unified
Communications and Cisco
Unified CallManager
Although IP telephony might seem like the new, emerging technology in IT environments, it has
actually been around for many years. As WAN and LAN data connections became more stable
and the total amount of available bandwidth increased, legacy PBX vendors rushed to add IP
processing functions to their chassis-based management systems. Cisco Systems also saw the
writing on the wall and in 1999 unveiled both the Cisco 7900 series IP Phones and the Cisco
Architecture for Voice, Video, and Integrated Data (AVVID) strategy for the future. This was
also the year that Cisco announced Cisco Unified CallManager version 2.4.
A Cisco IP telephony deployment relies on Cisco Unified CallManager for its call-processing
and call-routing functions. Understanding the role that Cisco Unified CallManager plays in a
converged network from a system, software, and hardware perspective is necessary to
successfully install and configure Cisco Unified CallManager. This chapter discusses Cisco
Unified Communications and the Cisco Unified CallManager functions, hardware
requirements, software requirements, and installation and upgrade information.
NOTE With the recent launch of the new Cisco Unified Communications portfolio, Cisco
AVVID has become an obsolete term because it no longer accurately describes the breadth of
Cisco’s converged, unified communications offerings. This chapter still occasionally makes
reference to Cisco AVVID in historical contexts, but for the most part uses the term Cisco
Unified Communications. You might find them used interchangeably during this transition
period.
NOTE Cisco has recently changed the name of the Cisco CallManager product to Cisco
Unified CallManager to reflect the Cisco Unified Communications campaign. For brevity, this
book mainly uses the shortened name of Cisco CallManager or just CallManager.
6 Chapter 1: Introduction to Cisco Unified Communications and Cisco Unified CallManager
Cisco Unified Communications

When Microsoft announced their new .NET platform, there was a huge amount of confusion in
the industry. Systems administrators did not know if they were to expect a new operating system,
programming language, or company direction. The same effect happened when Cisco announced
the Architecture for Voice, Video, and Integrated Data (AVVID), now known as Cisco Unified
Communications. Network administrators did not understand if this was a new router type, IOS
feature set, or marketing campaign. Cisco Unified Communications is not any of these things.
Instead, it is a company strategy that provides the foundation for converged networks. The goal of
Cisco Unified Communications is to create network equipment that has the capability to handle
voice, video, and data traffic within a single network infrastructure.
Figure 1-1 shows the four standard layers of the Cisco Unified Communications voice
infrastructure model: the infrastructure layer, which lays the foundation for network components;
the call-processing layer, which maintains PBX-like functions; the applications layer, where
applications that provide additional network functionality reside; and the client layer, where end-
user devices reside.
Figure 1-1 The Cisco Unified Communications Model for Voice Networks
Client
IP
Distributed
Adaptive
Video Cisco IP Communicator IP Phone PC
Applications
Cisco Unity Cisco IP Contact Center
Call Processing
Manageable
Call Admission, Call Routing Cisco CallManager Directory

Open
Infrastructure
Cisco IOS Network Services Gateway Router Switch

Understanding Cisco Unified CallManager 7
Each of the major areas of the Cisco Unified Communications architecture has a similar model
focused around the technologies of voice, video, and data. The key points about the four standard
layers of the voice model are as follows:
■ Infrastructure layer—The infrastructure carries data between all network devices and
applications and consists of routers, switches, and voice gateways.
■ Call-processing layer—Call processing is physically independent of the infrastructure.

Thus, a Cisco CallManager in Chicago can provide call control for a bearer channel in
Phoenix.
■ Applications layer—Applications are physically independent of call-processing functions

and the physical voice-processing infrastructure; that is, they can reside anywhere within the
network.
■ Client layer—The client layer makes the voice applications available to the user, whether the
end device is a Cisco IP Phone, a PC using a Cisco IP Communicator, or a PC delivering
converged messaging.
At first, this might seem like just another model to commit to memory; however, understanding
Cisco’s design of this model allows you to make the most of your voice network. The OSI model
creates a standard for communication across networks. In addition, it allows vendors to isolate
network functionality and specialize in equipment working at a specific layer. Likewise, one of the
huge benefits of using an IP telephony system over a PBX system is the open standards for
protocols and equipment. A vendor could specialize and design devices that work only at the client
layer of the Cisco Unified Communications voice model. These devices could use an open
standard protocol to communicate with the Cisco CallManager at the applications layer.
NOTE Because of the flexibility of the Cisco IP telephony network, many organizations have
created their own custom applications for the voice network. Cisco has made a Cisco
CallManager Software Development Kit (SDK) freely available to aid in the process of creating
custom Extensible Markup Language (XML) applications.
Understanding Cisco Unified CallManager

The vast majority of this CIPT book is focused on the core product that controls a Cisco IP
telephony network: Cisco Unified CallManager. Cisco CallManager brings enterprise telephony
features and functions to packet telephony devices. These devices include Cisco IP Phones, media-
processing devices, voice over IP (VoIP) gateways, and multimedia applications. Additional data,
voice, and video services, such as unified messaging, multimedia conferencing, collaborative
contact centers, and interactive multimedia response systems, interact with the IP telephony
solution through the Cisco CallManager application programming interface (API).
Cisco CallManager provides the following functions:
■ Call processing—Call processing refers to the complete process of originating, routing, and
terminating calls, including any statistical collection processes.
■ Signaling and device control—Cisco CallManager sets up all of the signaling connections
between call endpoints and directs devices such as phones, gateways, and conference bridges
to establish and tear down streaming connections.
■ Dial plan administration—The dial plan is a set of configurable rules that Cisco
CallManager uses to determine call routing. Cisco CallManager provides the ability to create
flexible dial plans for users.
■ Phone feature administration—Cisco CallManager extends services such as hold, transfer,

forward, conference, speed dial, last-number redial, Call Park, and other features to IP Phones
and gateways.
■ Directory services—Cisco Unified CallManager uses DC Directory as an embedded

Lightweight Directory Access Protocol (LDAP) directory. This directory stores
authentication and authorization information about users and is a standard feature of Cisco
CallManager (it does not require any special configuration or installation). However, Cisco
CallManager can also be integrated with a corporate directory such as the Netscape Directory
Server or Microsoft Active Directory.
■ Programming interface to external applications—Cisco CallManager provides a

programming interface to external applications such as Cisco IP SoftPhone, Cisco IP
Communicator, Cisco IP Interactive Voice Response (IVR), Cisco Personal Assistant, and
Cisco CallManager Attendant Console.
Cisco Unified CallManager and IP Phone Interaction

Cisco CallManager provides the intelligence behind call-processing functions when integrated
with Cisco IP Phones. These IP Phones are virtually a paperweight without the Cisco CallManager
instructing them with what they should do. The Cisco CallManager uses the Skinny Client Control
Protocol (SCCP, or Skinny) signaling protocol over IP to communicate with Cisco IP Phones for
call setup and maintenance tasks. When the call is set up, Cisco IP Phones communicate directly
using Real-Time Transport Protocol (RTP) to carry the audio.
TIP Most Cisco 7900 IP Phones ship with a software image, allowing them to use the Skinny
protocol to communicate with the Cisco CallManager. Cisco has also made a Session Initiation
Protocol (SIP) image that allows Cisco IP Phones to be used with a third-party management
system. Cisco CallManager does not support controlling the IP Phones using the SIP image as
of Cisco CallManager 4.1. However, Cisco has added this support in the Cisco CallManager 5.0
release.
You can better understand how Cisco CallManager performs call processing and signaling
functions by tracking a basic IP telephony call, as shown in Figure 1-2.
Figure 1-2 Cisco CallManager Basic Call Processing and Signaling Capabilities
Cisco CallManager
Skinny Protocol Signaling Skinny Protocol Signaling
IP IP
IP Phone IP Phone
Party A Real-Time Transport Protocol (RTP) Media Path Party B
In Figure 1-2, Party A (left IP Phone) wants to call Party B (right IP Phone). Party A picks up the
handset and dials the number of Party B. In this environment, dialed digits are sent to Cisco
CallManager, the call-processing engine. Cisco CallManager finds the address and determines
where to route the call.
Using the Skinny protocol, Cisco CallManager signals the calling party over IP to initiate a ring
back, and Party A hears ringing. Cisco CallManager also signals the destination phone to initiate
ringing.
When Party B picks up the telephone, the RTP media path opens between the two stations. Party
A or Party B can now initiate a conversation. Because the IP Phones manage this RTP media path
themselves, the Cisco CallManager is able to move out of the call-processing functions for this
call. The IP Phones require no further communication with Cisco CallManager until either Party
A or Party B invokes a feature, such as call transfer, call conferencing, or call termination. Even
if the Cisco CallManager were to fail during the course of the call, the RTP stream would continue
until one of the parties involved in the call decided to disconnect the call.
TIP The Skinny (SCCP) protocol uses TCP port 2000, whereas the RTP bearer stream uses
dynamically negotiated even UDP port numbers in the range from 16,384 to 32,767.
The Components of Cisco Unified CallManager

Cisco CallManager installs using an image-based procedure. The image expands and deploys the
following foundation operating system components:
■ Windows 2000 Server
■ Microsoft SQL Server 2000
■ DC Directory
■ Cisco IP Telephony Backup and Restore System (BARS)
Cisco CallManager server relies on Microsoft Windows 2000 for its operating system and
Microsoft Structured Query Language (SQL) Server 2000 for its database (both provided by Cisco
Systems). The operating system version that Cisco provides is called the Cisco IP Telephony
Operating System. For example, Cisco CallManager 4.1(2) requires Cisco IP Telephony
Operating System Version 2000.2.6 (or later) and the latest Cisco IP Telephony Server Operating
System service release. The latest operating system updates and service releases can be obtained
from Cisco using an authorized CCO login.
Cisco CallManager uses DC Directory as an embedded LDAP directory. This directory stores
authentication and authorization information about users and is standard with Cisco CallManager
(it does not require any special configuration or installation). Maintaining a user database for your
IP telephony system is optional, as the phone system will work just fine without requiring users to
log in to the system. However, if you want to deploy any of the advanced features such as
Extension Mobility or the Cisco IP SoftPhone, user authentication is necessary. Authentication
establishes the right of the user to access the system, whereas authorization identifies the
telephony resources that a user is permitted to use, such as a specific telephone extension.
The Cisco Customer Directory Plugin allows you to integrate Cisco CallManager with one of the
following enterprise directories:
■ Microsoft Active Directory, available with Microsoft Windows 2000
■ Microsoft Active Directory, available with Microsoft Windows 2003
■ Netscape Directory Server, Versions 4.1 and 4.2
■ Sun ONE Directory Server 5.x
The Cisco IP Telephony Backup and Restore System (BARS) can be used to back up Cisco
CallManager. Cisco BARS is installed separately from Cisco CallManager.
Cisco Unified CallManager Servers

In the original 2.4 version of Cisco CallManager, you were able to install the software on any
platform that you desired. By allowing this freedom, a level of hardware-based instability plagued
the original versions of the Cisco CallManager software. To compete with the legacy PBX-based
voice network, VoIP networks should maintain an uptime of 99.999 percent. To achieve this, Cisco
now requires you to install Cisco CallManager on a server that meets Cisco configuration
standards. For this reason, Cisco has collaborated with two server hardware manufacturers,
Hewlett-Packard and IBM, to create Cisco Media Convergence Servers (MCSs). Cisco chose
these platforms because they have proven their reliability in the industry over time. Table 1-1
provides the current hardware specifications for the Cisco CallManager MCSs at the time of this
writing.
Table 1-1 Cisco Media Convergence Server Platforms
Platform Space Processor CPU CPU Maximum Phones

Equipped Maximum Per Server
MCS 7815-I1 Tower Pentium 4 1 1 300

3060 MHz
MCS 7825-I1 1U Rack Pentium 4 1 1 1000

Mount 3400 MHz
MCS 7835-I1 2U Rack Nocona Xeon 1 2 2500

Mount 3400 MHz
MCS 7825-H1 2U Rack Nocona Xeon 2 2 7500

Mount 3400 MHz
MCS 7835-H1 2U Rack Pentium 4 1 1 1000

Mount 3400 MHz

Mount 3400 MHz

Mount 3400 MHz
All of these servers, with the exception of the 7815-I1, are rack-mountable and do not include a
monitor, mouse, or keyboard. Cisco designed the Cisco MCS for local setup, rack mounting, and
remote administration.
NOTE The higher-end Cisco servers (beginning with the MCS 7835-I1) also include
redundant power supplies and hard disk configurations.
Summary
At the turn of the millennium, Cisco moved forward with a new company direction described in
the acronym of AVVID: Architecture for Voice, Video, and Integrated Data. Under this new
banner, all three methods of communication would collapse under a single network infrastructure.
The evolved term to describe this converged, IP-based environment is now Cisco Unified
Communications. Standing at the forefront of the voice technology is the Cisco CallManager, the
Cisco call-processing system that controls and manages converged voice over data networks.
Cisco CallManager runs on a foundation of Windows 2000 and SQL 2000 and can integrate with
many popular directory services for user authentication purposes. Ever since the 3.x versions of
Cisco CallManager, Cisco restricted the hardware platforms capable of running Cisco
CallManager to the MCS series. This ensures a consistent hardware platform to provide stable
support for the critical voice services of an organization.
Review Questions
You can find the solutions to these questions in Appendix A, “Answers to Review Questions.”
1. The Cisco Unified Communications strategy is primarily focused around what goal?
a. to create a new line of routers and switches equipped with increased processor and
memory resources to handle a converged network environment
b. to design network equipment equipped with the hardware resources and software fea-
tures to handle a converged network environment
c. to allow voice, video, and data to flow smoothly across the Internet
d. to upgrade WAN links between locations to a speed capable of handling increased net-
work traffic
2. In which layer of the voice infrastructure model does the Cisco Unified CallManager reside?
a. infrastructure layer
b. call-processing layer
c. applications layer
d. client layer
Review Questions 13
3. Which of the following functions are performed by the Cisco CallManager server?
(Choose three.)
a. call processing
b. directory services
c. PBX integration and signaling
d. cial plan administration
e. RTP audio processing
4. Which of the following protocols does the Cisco CallManager use to communicate with Cisco
IP Phones?
a. H.323
b. MGCP
c. Skinny
d. SIP
5. Which of the following functions is the Cisco CallManager responsible for when setting up
a phone call between Cisco IP Phones? (Choose three.)
a. receiving dialed digits
b. initiating ring tones on dialed IP Phone
c. streaming RTP audio between devices
d. handling call-processing functions when the calling party presses the Hold button
6. Which of the following software components constitute the foundation operating system of
Cisco CallManager? (Choose two.)
a. Windows 2000
b. Windows 2003
c. SQL 2000
d. Exchange 2000
e. Solaris
7. If you decide not to integrate Cisco CallManager into your company LDAP-compliant
directory, what built-in option does Cisco CallManager provide to store user information?
a. DC Directory
b. Active Directory
c. LW Directory
d. Cisco LM Directory
8. You have purchased a Cisco MCS 7825 server. What is the maximum number of Cisco IP
Phones you will be able to support?
a. 250
b. 500
c. 1000
d. 2500
e. 7500
9. Maintaining a user directory is essential to the day-to-day operation of the IP telephony
network. (True/False)
10. Cisco IP Phones can use which of the following signaling protocols? (Choose two.)
a. H.323
b. Skinny
c. SIP
d. MGCP
following topics:
■ The concepts and design strategies behind a

Cisco Unified CallManager cluster
■ The types of replication that occur in a Cisco

Unified CallManager cluster
■ Properly designing the Cisco Unified

CallManager cluster for redundancy
■ The four Cisco Unified CallManager

deployment models
CHAPTER 2
Clustering and Deployment
Options
Legacy PBX systems have quite the reputation for reliability. As a general track record, PBXs
provide an uptime of 99.999 percent (known as the “five-nines” of reliability). This translates to
an unbelievable 5.3 minutes of downtime per year! To match this standard of reliability, multiple
Cisco CallManager servers are grouped into a cluster. Most administrators recognize the term
“cluster” as an ability to allow multiple servers to act as one device, which is the type of
clustering provided by Windows clustering services. The Cisco CallManager cluster operates
quite differently.
After you understand the clustered relationship of Cisco CallManager, you can plan your
deployment strategy using the four deployment models provided by Cisco CallManager. This
chapter discusses the cluster relationship and deployment options provided by the Cisco
CallManager and the options available to enterprises to deploy a highly available IP telephony
network.
The Two Sides of the Cisco Unified CallManager Cluster

As a network administrator, you are accustomed to quite a bit of fiscal responsibility riding on
your shoulders. Under the Cisco Unified Communications architecture, that responsibility has
grown exponentially. Not only are you responsible for the operation of the data environment;
you are now responsible for the company’s voice network, which rides on top of this
infrastructure. This voice network is critical to day-to-day business operations. Because of this,
you should approach it just like any key network service: the more redundancy, the better.
A Cisco CallManager cluster is two or more servers grouped together to support a Cisco IP
telephony network. The cluster relationship between Cisco CallManager servers provides
redundancy and load balancing for the voice network. Cisco defines this cluster relationship in
two ways: the SQL database structure and the intracluster run-time data.
The SQL Database Cluster

As discussed in Chapter 1, Cisco CallManager relies on Microsoft SQL 2000 as an information
store for the data of the voice network. This data includes the phone extensions on the network,
calling restrictions, route plan information, and so on. The database replication capability
18 Chapter 2: Cisco Unified CallManager Clustering and Deployment Options
provided by Microsoft SQL Server makes clustering possible by allowing the same database to be
on multiple machines. Database replication makes it appear as if a single machine is handling call
processing along with other functions of the voice network and ensures that standby processors
(Cisco CallManager servers) can seamlessly step in and fulfill the functions if the primary
processor fails. This SQL database replication also ensures that all clustered Cisco CallManager
servers have access to the same information.
You must have at least two Cisco CallManager servers to obtain this redundancy, and one of these
servers must be a publisher database server. The publisher database server manages the only
writable copy of the Microsoft SQL Server 2000 database. The subscriber database servers
maintain read-only copies of the database. You can have only one publisher server and up to eight
subscriber servers per cluster. It is these database servers that are able to actively participate in the
call-processing functions of the Cisco voice network. This SQL limitation is also a key factor in
determining the maximum size of a cluster, which is covered later in this chapter.
When you make changes to the Cisco CallManager configuration, these changes are made directly
to the publisher server database. The publisher then replicates these changes to the subscriber
servers. When the publisher server is offline, the Microsoft SQL Server 2000 database
automatically locks, and thus prevents any database changes. The IP telephony network continues
to operate, but you will not be able to add or configure any devices that are managed by Cisco
CallManager. The only exception to this rule is the Call Detail Records (CDRs), which record
information regarding the calls occurring within the cluster. When the publisher is down, the
subscribers store CDRs until the publisher comes back online, and then the subscribers update the
publisher with the CDRs.
In Cisco CallManager Release 3.3 and later, a single cluster is capable of handling approximately
30,000 Cisco IP Phones. This cluster limitation does not restrict the size of the voice over IP (VoIP)
network. By creating additional clusters, you can increase the network size. However, the more
clusters you create in a network, the more management the network requires to operate.
NOTE The 30,000 IP Phone maximum cluster size is only possible if you are using the MCS-
7845 servers throughout your cluster deployment.
Intracluster Run-Time Data

The second communication method that defines the cluster relationship between Cisco
CallManager servers is the intracluster run-time data, which is also called Intra-Cluster
Communication Signaling (ICCS). This type of communication encompasses the “happenings” of
the cluster. For example, when a new Cisco IP Phone connects to the network, it registers with its
primary Cisco CallManager. That primary Cisco CallManager tells all the other servers in the
cluster, “Hey everyone, a new phone just registered with me! I’ve given it the extension 4003.” (IP
address 10.5.5.1 in this example). All the other servers now know to send calls directed at
extension 4003 to the Cisco CallManager at the IP address 10.5.5.1, which, in turn, makes the
Cisco IP Phone ring.
After the initial phone registration, the Cisco IP Phone sends keepalive messages to the primary
Cisco CallManager server every 30 seconds and sends a TCP connect message (which is
technically a TCP three-way handshake) to its secondary Cisco CallManager server to ensure it is
online and ready to accept a device failover, if necessary. When the Cisco IP Phone detects the
failure of its TCP keepalive messages with the primary Cisco CallManager, the device attempts to
register with a secondary Cisco CallManager server. The secondary Cisco CallManager server
accepts the registration from the device and announces the new registration (using intracluster run-
time data) to all of the Cisco CallManager servers in the cluster.
NOTE If the Cisco CallManager service is ever stopped manually (through the Services
Windows 2000 Control Panel or a system shutdown), the Cisco CallManager server will clear
all of its active TCP connections with the IP Phones. This causes them to failover immediately
to their backup server rather than wait for the keepalive failure.
Cisco IP Phone registration is just one example of intracluster run-time data. You will discover
many other types of intracluster communication as this book introduces new concepts in
upcoming chapters.
Cluster Redundancy Designs

Because the IP telephony network is critical to business operation, you should never configure a
single Cisco CallManager to support any organization. You should maintain at least two Cisco
CallManager servers for the voice network to support the network should one server fail. Smaller
organizations (240 users or less) might decide to use Cisco CallManager Express, which operates
on an IOS-based router platform. The number of Cisco CallManager servers you use typically
depends on the server platform you purchase and your cluster design strategy. Chapter 1,
“Introduction to Cisco Unified Communications and Cisco Unified CallManager,” discussed the
server platforms and the number of phones that each server model supports. This section discusses
the proper way to use those server platforms to build a Cisco CallManager cluster.
1:1 Redundancy Design

In a 1:1 Cisco CallManager redundancy deployment design, you can have a dedicated backup
server for each primary server. This design guarantees that Cisco IP Phone registrations will never
overwhelm the backup servers, even if multiple primary servers fail. However, the 1:1 redundancy
design considerably limits the maximum cluster size and can be cost-intensive for the initial Cisco
CallManager deployment.
Each cluster must also have a designated TFTP server. Depending on the number of devices that
a server is supporting, you can combine this TFTP server functionality with the publisher or
subscriber Cisco CallManager servers, or you can deploy the TFTP functionality on a separate,
standalone server. The TFTP server is responsible for delivering IP Phone configuration files to
each telephone, along with streamed media files, such as music on hold (MOH) and ring files;
therefore, the TFTP server can experience a considerable network and processor load.
In the example shown in Figure 2-1, a Cisco 7835 Media Convergence Server (MCS) is used
because each Cisco CallManager server installed on that platform supports a maximum of 2500
Cisco IP Phones.
Figure 2-1 1:1 Server Redundancy Design
2500 IP Phones 5000 IP Phones 10,000 IP Phones

(5000 Device Units) (10,000 Device Units) (20,000 Device Units)
Cisco MCS 7835 Cisco MCS 7835 Cisco MCS 7835

Publisher and
Publisher and Publisher and
TFTP Server
TFTP Server TFTP Server
(Not Req. <1000)
Primary 1 to 1 to
1 to 2500 2500 2500
Backups Backups
2501 to 2501 to
Backup
5000 5000
5001 to
7500
Backups
7501 to
10,000
The shading in the figure divides up the functions performed within a cluster. The lighter shading
(outside the Primary/Backup server area) represents the server that is not participating in call
processing. This is the SQL publisher server, which is also acting as a TFTP server. In cluster
environments with less than 1000 IP Phones, the SQL publisher can also participate in the call-
processing functions. However, in clusters larger than 1000 devices, you should isolate the SQL
publisher because it will be quite busy handling updates to the SQL database and TFTP requests.
The darker shaded box (containing the Primary/Backup servers) represents Cisco CallManager
servers handling the call-processing functions (intracluster run-time data) of the cluster. In the
example showing cluster design for 2500 IP Phones, a single Cisco CallManager is the primary
server, with a secondary server acting as a dedicated backup. The primary or backup server can
also serve as the Microsoft SQL publisher and the TFTP server in smaller IP telephony
deployments.
TIP Be sure you understand how the two types of intracluster communication apply to the
Cisco CallManager cluster design. The publisher and subscriber relationship relates to the SQL
database replication. The Primary and Secondary server roles (shown in Figure 2-1 as Primary
and Backup) relate to the intracluster run-time relationship and device registration. The Primary
and Backup servers will be SQL subscribers if you isolate the SQL publisher from call-
processing functions.
When you increase the number of IP Phones, you must increase the number of Cisco CallManager
servers that are required to support the telephones. Some network engineers might consider the
1:1 redundancy design excessive because a well-designed network is unlikely to lose more than
one primary server at a time. With the low possibility of server loss and the increased server cost,
many network engineers elect to use a 2:1 redundancy design. However, other engineers choose
to deploy the 1:1 redundancy to ensure maximum uptime of the IPT network. In addition, you can
load-balance your IP Phones between the primary and backup servers in a 1:1 redundancy design.
If any one server fails, only half of the phones must failover to the backup server (which provides
a faster failover process).
2:1 Redundancy Design

In a 2:1 Cisco CallManager redundancy deployment design, you have a backup server shared
between every two primary servers, as shown in Figure 2-2. Although this design offers some
redundancy, there is the risk of overwhelming the backup server if multiple primary servers fail.
In addition, upgrading the Cisco CallManager servers can cause a temporary loss of service
because you must reboot the Cisco CallManager servers after the upgrade is complete.
NOTE Figure 2-2 assumes that each Cisco CallManager server can support a maximum of
2500 IP Phones.
Figure 2-2 1:2 Server Redundancy Design
2500 IP Phones 5000 IP Phones 10,000 IP Phones

(5000 Device Units) (10,000 Device Units) (20,000 Device Units)
Cisco MCS 7835 Cisco MCS 7835 Cisco MCS 7835

Publisher and
Publisher and Publisher and
TFTP Server
TFTP Server TFTP Server
(Not Req. <1000)
Primary 1 to 1 to
1 to 2500 2500 2500
Backup Backup
2501 to 2501 to
Backup
5000 5000
5001 to
7500
Backup
7501 to
10,000
Network administrators use this 2:1 redundancy model in most IP telephony deployments because
of the reduced server costs. If you are using a Cisco MCS 7835 (shown in the figure), that server
is equipped with redundant, hot-swappable power supplies and hard drives. When you properly
connect and configure these servers, it is unlikely that multiple primary servers will fail at the same
time, which makes the 2:1 redundancy model a viable option for most businesses.
Because of tight budgets, some network administrators are forced into 3:1 and even 4:1
redundancy designs (three or four primary servers supported by a single backup server). Cisco
does not support these designs because of the high risk involved.
TIP The Cisco CallManager architecture limits a cluster to nine servers (one publisher and
eight subscribers) that have access to the SQL database. Only these servers can participate in
call-processing functions. The redundancy model and server platform you choose has a huge
impact on the total size of your Cisco CallManager cluster. Although it might be possible to
create a cluster capable of supporting a tremendous amount of IP Phones by minimizing
redundancy, Cisco TAC only supports a maximum cluster size of 30,000 IP Phones. If you reach
this maximum cluster size, you will need to divide your network into multiple clusters.
Call-Processing Deployment Models

After you understand the concepts and design of a Cisco CallManager cluster, you can move on
to the design of the organization IP telephony network. Cisco proposes four primary design
models:
■ Single-site deployment
■ Multisite deployment with centralized call processing
■ Multisite deployment with distributed call processing
■ Clustering over the IP WAN
Single-Site Deployment
In a single-site deployment model, all Cisco CallManager servers, applications, and IP Phones are
in the same physical location. A single Cisco CallManager cluster can support a maximum of
30,000 IP Phones. If you need to support more IP Phones at the site, you can implement multiple
clusters within the single location and interconnect them through intercluster trunks (intercluster
trunks are covered in Chapter 10, CallManager Route Plan Basics). Gateways that connect directly
to the Public Switched Telephone Network (PSTN) handle external calls. If an IP WAN exists
between sites, it is used to carry data traffic only; no telephony services are provided over the
WAN. Figure 2-3 shows a typical network diagram of a single-site deployment.
Figure 2-3 Single-Site Cisco CallManager Design
Applications
Cisco
CallManager
Cluster
IP
V
PSTN
V
IP
Use the following guidelines for single-site deployments:
■ You must understand the current calling patterns within the enterprise. How and where are
users making calls? How many calls are intersite or interbranch versus intrasite? If calling
patterns dictate that most calls are intrasite, use the single-site model to deploy IP telephony
and make use of the relatively inexpensive PSTN. This design also simplifies the dial plans
and avoids provisioning dedicated bandwidth for voice in the IP WAN.
■ The IP Phones should use the G.711 coder-decoder (codec). Because the call will stay in the
LAN, G.711 is a simple and high-quality codec for deployment. It does not require dedicated
Digital Signal Processor (DSP) resources for transcoding (which means converting between
codec types, such as between G.711 and G.729), and older voice-mail systems might support
only G.711. You can allocate these DSP resources to other functions, such as conferencing
and Media Termination Point (MTP). Although the 64 kbps per-call bandwidth that G.711
consumes is higher than that of other commonly used codecs, it is not a concern in this design
because the call is not traversing the WAN, where bandwidth is generally limited.
■ All off-net calls will be diverted to the PSTN or sent to the legacy PBX for call routing if the
PSTN resources are being shared during migratory deployments.
■ Use Media Gateway Control Protocol (MGCP) gateways for the PSTN if the network does
not require H.323 functionality. Centralize the gateway dial plans by using H.323 gatekeepers
when deploying multiple clusters, rather than using MGCP gateways. In addition, PSTN
gateway redundancy should also be a consideration because a single gateway failure could
affect the inbound and outbound PSTN calls for an entire corporation.
■ Deploy the recommended network infrastructure for high-availability connectivity options

for telephones (inline power), quality of service (QoS) mechanisms, and other services.
■ Do not oversubscribe Cisco CallManager to scale larger installations. In earlier software

releases, Cisco used various schemes to allow the capacity of a system to be calculated using
device weights, busy hour call attempt (BHCA) multipliers, and dial plan weights. With Cisco
CallManager Release 4.0, this scheme has been replaced by a capacity tool to allow for more
accurate planning of the system. The capacity planning tool is currently available only to
Cisco employees. If your system does not meet the guidelines found in the Cisco IP
Telephony Solution Network Reference Design, or if you consider the system to be more
complex and would like to verify the capacity, please contact your Cisco Systems engineer.
The Cisco IP Telephony Solution Reference Network Design (SRND) is found at: http://
www.cisco.com/univercd/cc/td/doc/solution/esm/ (or can be browsed to through http://
www.cisco.com/go/srnd).
Multisite Deployment with Centralized Call Processing

Figure 2-4 illustrates the multisite centralized call-processing deployment model with a Cisco
CallManager cluster at a central site and a connection to several remote sites through a QoS-
enabled IP WAN. The remote sites rely on the centralized Cisco CallManager cluster to handle call
processing. Applications such as voice mail and interactive voice response (IVR) systems usually
reside at the central site, thus reducing the overall cost of ownership and centralizing
administration and maintenance. However, centralizing these application servers is not a
requirement.
Figure 2-4 Multisite Deployment with Centralized Call Processing
Application
Servers
SRST-Enabled
Cisco PSTN Router
IP
CallManager
Cluster
IP
V
IP WAN Branch A
IP IP IP
Headquarters
SRST-Enabled
Router
IP
IP
Branch B
Routers that reside at WAN edges require QoS mechanisms, such as low latency queuing (LLQ)
and traffic shaping, to protect voice traffic from data traffic across the WAN (where bandwidth is
typically scarce).
To avoid oversubscribing the WAN links with voice traffic (thus causing deterioration of the
quality of established calls), the network might need a call admission control scheme. With the
introduction of Cisco CallManager Release 3.3, centralized call-processing models can take
advantage of automated alternate routing (AAR) features. AAR allows Cisco CallManager to
dynamically reroute a call over the PSTN if the call exceeds the WAN bandwidth.
NOTE AAR is a feature specific to centralized call-processing environments and is discussed

in depth in Chapter 14, Implementing Multiple-Site Deployments.
You can provide PSTN access for the voice network through a variety of Cisco gateways. When
the IP WAN is down, the users at remote branches can place their calls through the PSTN using
the Cisco Survivable Remote Site Telephony (SRST) feature that is available for Cisco IOS
gateways that can provide call processing during the outage.
Follow these best-practices guidelines when deploying a centralized call-processing model:
■ Installations adopting the centralized call-processing deployment model are limited to hub-
and-spoke topologies because the locations-based call admission control mechanism used in
centralized call-processing deployments records only the available bandwidth in and out of
each location.
■ There is no limit to the number of IP Phones at each individual remote branch. However, the
capability that is provided by the SRST feature in the branch router limits remote branches to
720 Cisco IP Phones using a 3845 router during failover. Smaller platforms have lower limits.
SRST is covered in its entirety in Chapter 14, “Implementing Multiple-Site Deployments.”
■ Controlling WAN bandwidth utilization in a centralized call-processing deployment model is

quite simple. Through the use of the Cisco CallManager Locations feature, the central Cisco
CallManager cluster can decide if the remote-site call should use the WAN or PSTN for its
connection.
■ When communicating over the IP WAN, the G.729 compressed codec should be used to save
considerable WAN bandwidth.
Multisite Deployment with Distributed Call Processing

A multisite distributed call-processing deployment has one or more call-processing agents at each
site, and each site has its own Cisco CallManager cluster consisting of two or more servers. By
separating these sites into different clusters, the Cisco CallManager servers do not share the SQL
database and intracluster run-time data between sites. This requires some additional configuration
because it will be necessary to trunk these sites together through an IP WAN. Figure 2-5 illustrates
a typical multisite design using distributed call processing.
When deciding between a centralized or distributed call-processing model, you will usually base
your decision on two factors: how many phones must you support at each site and what features
does the company require to be available during failover. If you are using a smaller number of IP
Phones and only require basic call-processing functionality, the centralized call-processing model
might be most appropriate. If you are using a larger number of IP Phones or require more advanced
features (such as call center applications), a distributed model might be more appropriate.
Figure 2-5 Multisite Deployment with Distributed Call Processing
Cisco
Application CallManager
Servers Cluster
Cisco PSTN
V
CallManager
Cluster
IP IP IP
Branch A
V
IP WAN
IP IP IP GK
Application
Gatekeeper Servers
Headquarters
Cisco
CallManager
Cluster
IP IP IP
Branch B
Depending on your network design, an individual site in the distributed call-processing design
might consist of the following:
■ A single site with its own call-processing agent, which can be a Cisco CallManager or a
third-party call agent
■ A centralized call-processing site (and all of its remote sites) that the network views as a
single site for distributed call processing
■ A legacy PBX with a VoIP gateway or a legacy PBX that is attached using a time-division
multiplexing (TDM) interface to a VoIP gateway
The distributed call-processing model requires all sites to be connected through an IP WAN. Cisco
considers a site connected only through the PSTN to be a standalone site.
Multisite distributed call processing allows each site to be completely self-contained. In the event
of an IP WAN failure or insufficient bandwidth, the site does not lose call-processing service or
functionality. Cisco CallManager simply sends all calls between the sites across the PSTN.
The main benefits of this deployment model are as follows:
■ Cost savings when you are using the IP WAN for intersite calls. The IP Phones and voice
gateways can compress calls using the G.729 codec, reducing the bandwidth required for the
audio and increasing efficiency over the standard PSTN connections.
■ Toll-bypass savings when you are using remote gateways to drop off into the PSTN (known
as “tail end hop off,” or TEHO). For example, if one of your sites is located in Arizona, all the
other sites can forward calls across the IP WAN to obtain toll-bypass when calling a number
in Arizona.
■ No loss of functionality during an IP WAN failure.
■ Scalability to hundreds of clusters.
The multisite WAN with distributed call-processing deployment model is a superset of the single-
site and multisite WAN with centralized call-processing models. You should follow the best-
practices guidelines mentioned previously for single-site and multisite deployments in addition to
those listed here, which are specific to this deployment model.
The H.323 gatekeeper or Session Initiation Protocol (SIP) proxy servers are among the key
elements in the multisite WAN with distributed call processing. Both provide centralized dial plan
resolution, with the gatekeeper also providing call admission control.
A gatekeeper is an H.323 device that provides call admission control and centralized dial plan
resolution. Otherwise, the network administrator at each of the sites must configure a full dial plan
to reach all devices on the network. Additional gatekeeper guidelines include the following:
■ Cisco recommends that you use backup gatekeeper support to provide a gatekeeper solution
with high availability. It is also recommended that you use multiple gatekeepers to provide
spatial redundancy within the network.
■ Cisco recommends that you use a single WAN codec. This design makes capacity planning
easy and does not require you to overprovision the IP WAN to allow for worst-case scenarios.
SIP proxy servers provide resolution of dialed numbers (also called E.164 numbers) as well as SIP
uniform resource identifiers (URIs) to enable endpoints to place calls to each other. Cisco
CallManager supports the use of E.164 numbers only.
The following best practices apply to the use of SIP proxies:
■ Provide adequate redundancy for the SIP proxies.
■ Ensure that the SIP proxies have the capacity for the call rate and number of calls required in
the network.
For more detail on bandwidth capacity planning and call admission control for each deployment
model, refer to the Cisco IP Telephony Solution Reference Network Design (SRND) for Cisco
CallManager 4.0 at: http://www.cisco.com/univercd/cc/td/doc/solution/esm/iptele/index.htm.
Clustering over the IP WAN

Cisco supports Cisco CallManager clusters over a WAN, shown in Figure 2-6.
Figure 2-6 Clustering over the IP WAN
Cisco
CallManager Cluster
Voice-Mail Voice-Mail
Server Server
IP IP
IP IP
IP Phones IP Phones
Los Angeles San Diego
Although there are stringent requirements, this design offers the following advantages:
■ Single point of administration for users for all sites within the cluster
■ Feature transparency; features such as call transfers and conference calls work seamlessly
using an internal dialing scheme
■ Shared line appearances over a WAN connection, which allows multiple phones to share a line
instance
■ Extension mobility within the cluster, allowing users to log in to remote phones and have their
phone profile follow them
This design is useful for customers who require more functionality at remote sites than the limited
feature set that is offered by SRST. This network design also allows remote offices to support more
Cisco IP Phones than SRST in the event that the connection to the primary Cisco CallManager is
lost.
Summary 31
Although the distributed single-cluster call-processing model offers some significant advantages,
it must adhere to these strict design guidelines:
■ Two Cisco CallManager servers in a cluster must have a maximum round-trip delay of 40 ms
between them. In comparison, high-quality voice guidelines dictate that one-way end-to-end
delay should not exceed 150 ms. Because of this strict guideline, you can use this design only
between closely connected, high-speed locations (Metropolitan Ethernet would be a good
example of this connectivity type).
■ For every 10,000 Busy Hour Call Attempts (BHCAs) within the cluster, you must support an
additional 900 kbps of WAN bandwidth for intracluster run-time communication. The BHCA
represents the number of call attempts made during the busiest hour of the day.
■ Up to eight small sites are supported using the remote failover deployment model. Remote
failover allows you to deploy the backup servers over the WAN. Using this deployment
model, you can have up to eight sites with Cisco CallManager subscribers being backed up
by Cisco CallManager subscribers at another site.
■ SRST can function in this model but is not necessary to protect against Cisco CallManager
failure. The telephones can failover across the WAN to other Cisco CallManager servers. This
design may require significant additional bandwidth, depending on the number of telephones
at each location.
NOTE If the WAN connection between sites fails, users will receive a fast busy signal for any
intersite call. This might be confusing to users who are used to ringing followed by voice mail.
Summary
This chapter covered the key factors you must consider when you are initially designing a Cisco
CallManager–based IP telephony deployment. The key to understanding your deployment options
is first understanding the Cisco CallManager cluster relationship as defined by the Microsoft SQL
replication and intracluster run-time data. The Cisco CallManager cluster provides redundancy
and failover in an IP telephony environment. This redundancy can be implemented using the 1:1
server redundancy design, which offers the most redundancy available, or a 2:1 redundancy
design, which balances the cost factors with overall redundancy. The chapter ended with a
discussion of the four call-processing deployment models:
■ Single-site deployment
■ Multisite deployment with centralized call processing
■ Multisite deployment with distributed call processing
■ Clustering over the IP WAN

Review Questions
1. Which four of the following are IP telephony deployment models that are supported by Cisco?
(Choose four.)
a. a single site with one call-processing agent
b. multiple sites with centralized call processing
c. multiple sites each with its own call-processing agent
d. a single cluster with distributed call processing
e. multiple clusters with no call-processing agent
2. A 1:1 redundancy design offers _________.
a. increased redundancy; however, the increased server cost is often prohibitive
b. some redundancy; however, a server reboot is required after an upgrade
c. maximum uptime; however, no more than a 20-ms round-trip delay can exist between
servers
d. high availability; however, you might overwhelm the backup servers
3. A single cluster that spans multiple sites can have which two benefits compared to a branch
office in a multisite WAN with a centralized call-processing deployment? (Choose two.)
a. completely self-contained individual sites
b. WAN bandwidth cost savings
c. a common dial plan across all sites
d. more IP Phone features during failover
e. scalability to hundreds of sites
4. Which two of the following enable the cluster to achieve redundancy? (Choose two.)
a. Windows clustering
b. database replication
c. at least two servers
d. directory access
5. When configuring the SQL database relationship between Cisco CallManager servers, how
many publisher servers should you configure for the cluster?
a. It depends on the server platform; you should have one publisher for each group of
phones followed by a backup.
b. You should have only one publisher server for the entire cluster.
Review Questions 33
c. You should have at least two publisher servers to provide database redundancy.
d. You must configure a publisher for each subscriber you install.
6. What is the maximum number of SQL subscribers that can exist in a Cisco CallManager
cluster?
a. 3
b. 5
c. 8
d. 9
e. 12
7. To support a centralized, multisite Cisco CallManager design, what feature should be
employed at the remote offices?
a. QoS
b. SRST
c. Classification
d. SQL database replication
8. When using a multisite, single-cluster Cisco CallManager design, it is not necessary to
employ SRST features to support the IP Phones in the case of Cisco CallManager failure.
(True/False)
9. When designing a distributed, multicluster Cisco CallManager design, what pieces of
equipment might be necessary to ensure the IP telephony network maintains a consistent and
centralized dial plan? (Choose two.)
a. H.323 gatekeeper
b. H.323 gateways
c. SIP proxy server
d. Cisco CallManager
10. What is the maximum Cisco CallManager cluster size Cisco will support using Cisco
CallManager 4.x?
a. 7500 IP Phones
b. 10,000 IP Phones
c. 12,000 IP Phones
d. 15,000 IP Phones
e. 30,000 IP Phones
following topics:
■ The requirements to perform Cisco Unified

CallManager server installation
■ Cisco Unified CallManager installation

process
■ Cisco Unified CallManager postinstallation

procedures
■ Cisco Unified CallManager upgrade process

CHAPTER 3
Installation and Upgrades
When Cisco first announced that the Cisco CallManager platform would use Microsoft
Windows 2000 as a foundation operating system and Microsoft SQL 2000 as a database store,
concern arose that the Cisco administrator would require a thorough understanding of these
applications to successfully operate a Cisco IP telephony network. Although an understanding
of these components can be useful for monitoring and troubleshooting purposes, it is not
necessary for Cisco CallManager installation and setup. Cisco has done a fantastic job of
“wizard-izing” and scripting the complete installation of both Windows 2000 and SQL 2000,
hiding any complexity behind a friendly Next button.
The upgrade process of Cisco CallManager is not quite as friendly as a clean install; however,
if you keep the potential pitfalls in mind, a Cisco CallManager upgrade can go quite smoothly.
This chapter discusses both the clean install and upgrade process of Cisco CallManager.
Cisco Unified CallManager 4.x Clean Installation Process

A clean installation of Cisco CallManager has always been an extremely simple process. As you
perform the Cisco CallManager installation, the automated setup process prompts you for the
information that is necessary to build Windows 2000, Microsoft SQL Server 2000, and Cisco
CallManager with a base configuration. The entire operating system installation process,
excluding preinstallation tasks, takes approximately 25 to 45 minutes per server, depending on
your server type. Installing Cisco CallManager, excluding pre- and postinstallation tasks, takes
45 to 90 minutes per server, depending on your server type.
Installation Disks
All Cisco MCSs and customer-provided servers that meet approved Cisco configuration
standards ship with a blank hard drive. When you purchase a Cisco IP telephony application,
you use the appropriate disks to install or upgrade the operating system and application:
■ Disk 1: Cisco IP Telephony Server Operating System Hardware Detection Disk—

Checks the server and displays an error message if it detects an unsupported server. After
you boot the server using the Hardware Detection CD-ROM, the automated installation
process prompts for the correct CD-ROMs to use based on the type of hardware platform
detected.
36 Chapter 3: Cisco Unified CallManager Installation and Upgrades
■ Disk 2: Cisco IP Telephony Server Operating System Installation and Recovery Disk—
Installs the operating system. Use only one of the server-specific Cisco IP Telephony Server
Operating System Installation and Recovery disks that come in your software kit. Depending
on your platform, the Operating System disc could be CD-ROM or DVD-based. After the
operating system installation, a prompt instructs you to insert the appropriate Cisco
CallManager software disk into the drive.
■ Disk 3: Cisco CallManager 4.1 Software Disk— This disk installs the Cisco CallManager
application on the server.
You might also receive a Cisco IP Telephony Server Operating System Upgrade Disk. Use this
disk to upgrade the operating system on existing (not new) servers in the cluster. You do not need
to use this disk if you are performing a new operating system installation.
Installation Configuration Data

As mentioned previously, the installation process for Cisco CallManager is automated by a step-
by-step wizard. You will initially boot off the Hardware Detection CD-ROM, which will walk you
through a wizard prompting you for the basic configuration data to get the server running. The
process erases all data on the server hard disk. During the installation, you are prompted for the
following items:
■ New installation or server replacement—Choose this option if you are installing the Cisco
IP telephony application for the first time, overwriting an existing installation, or replacing a
server. To replace the server, you must store the data to a network directory or tape device
before the operating system installation. Choosing this setting erases all existing drives.
■ Cisco product key—Cisco supplies a product key when you purchase a Cisco IP telephony
product. The product key is based on a file encryption system that allows you to install only
the components that you have purchased. It also prevents you from installing other supplied
software for general use. The product key consists of alphabetical characters only.
■ Username and organization name—The system will prompt you for a username and an
organization name to register the software product that you are installing. Do not leave the
field blank. You can enter letters, numbers, hyphens (-), and underscores (_).
■ Computer name—The system will prompt you to assign a unique computer name, using 15
characters or fewer, to each Cisco CallManager server. The computer name can contain
alphabetic and numeric characters, hyphens, and underscores, but it must begin with a letter
of the alphabet. Follow your local naming conventions, if possible. If you want to change the
computer name after the application installation, you must completely reinstall the operating
system and the application.
Cisco Unified CallManager 4.x Clean Installation Process 37
■ Workgroup—The system will also prompt you for a workgroup name. A workgroup consists
of a collection of computers that share the same workgroup name. Computers in the same
workgroup can more easily communicate with each other across the network. Ensure that this
entry, which must also be 15 characters or fewer, follows the same naming conventions as the
computer name.
■ Domain suffix—When prompted, you must enter the Domain Name System (DNS) suffix in
the format “mydomain.com” or “mycompany.mydomain.com.” If you are not using DNS, use
a fictitious domain suffix, such as fictitioussite.com.
■ TCP/IP properties—You must assign an IP address, subnet mask, and default gateway when
installing a Cisco CallManager server. Changing the Cisco CallManager IP address after you
install the software can be a tedious process, so be sure to plan accordingly.
CAUTION It is strongly recommended that you choose static IP information, which ensures
that the Cisco CallManager server obtains a fixed IP address. With this selection, Cisco IP
Phones can register with Cisco CallManager when the telephones are plugged into the network.
Using Dynamic Host Configuration Protocol (DHCP) can cause problems, including failure of
the telephony system.
■ DNS—You can identify a primary DNS server for this optional field. By default, the
telephones will attempt to connect to Cisco CallManager using DNS. Therefore, you must
verify that the DNS server contains a mapping of the IP address and the fully qualified domain
name (FQDN) of the Cisco CallManager server. If you do not use DNS, use the server IP
address, instead of a server name, to register the telephones with Cisco CallManager.
NOTE Before you begin installing multiple servers in a cluster, you must have a name
resolution method in place, such as DNS, Windows Internet Naming Service (WINS), or local
name resolution using a configured LMHOSTS file. If you use DNS, you must verify that the
DNS server contains a mapping of the IP address and the hostname of the server that you are
installing. This verification must take place before you begin the installation. If you use local
name resolution, ensure that the LMHOSTS file is updated on the existing servers in the cluster
before you begin the installation on the new subscriber server. You must add the same
information to the LMHOSTS file on the new server during installation.
TIP Although it might seem tedious, Cisco considers the creation of LMHOST file IP address
to hostname mappings on each Cisco CallManager server a better practice. Using DNS services
introduces another point of failure for the voice network.
■ SNMP community string—The Windows 2000 Simple Network Management Protocol

(SNMP) agent provides security through the use of community names and authentication
traps. Cisco sets the community rights to none for security reasons. If you want to use SNMP
with this server, you must configure it.
■ Database server—You must determine whether you will configure this server as a publisher
database server or as a subscriber database server through a radio button selection during the
Cisco CallManager installation. This selection is permanent. You must reinstall the Cisco
CallManager server if you want to reassign the database server type at a later date.
NOTE You must install a Cisco CallManager publisher server before you can install any
subscriber servers. When you are configuring a subscriber database server, ensure that the server
that you are installing can connect to the publisher database server during the installation. This
connection facilitates the copying of the publisher database to the local drive on the subscriber
server. You must supply the name of the publisher database server and a username and password
with administrator access rights on that server. The installation will be discontinued if, for any
reason, the publisher server cannot be authenticated.
■ New password for the system administrator—Cisco CallManager Releases 3.0 and later
support password protection. A prompt at the end of the installation procedure will ask you
to supply a new password for the system administrator.
NOTE For Cisco CallManager database replication, you must enter the same Administrator
account password for the publisher and all of the subscribers in the cluster. The installation
wizard will request this password.
Sample Configuration Data Worksheet

Table 3-1 shows the configuration information that you need to install the Cisco CallManager
software on your server. You should complete all of the fields in the table, unless otherwise noted.
You must gather this information for each Cisco CallManager server that you are installing in
the cluster. Make copies of this table, and record your entries for each server in a separate table.
Table 3-1 summarizes the data you should have available when you begin the installation.
Table 3-1 Configuration Data for Cisco MCS
Configuration Data
Cisco product key
Username
Name of your organization

Table 3-1 Configuration Data for Cisco MCS (Continued)
Configuration Data
Computer name
Workgroup
Microsoft NT domain (optional)
DNS domain suffix
Current time zone, date, and time
DHCP parameters It is recommended that you program a fixed

IP address in TCP/IP properties for the
server instead of using DHCP.
TCP/IP properties (required if DHCP is not used):

• IP address
• Subnet mask
• Default gateway
DNS servers (optional):

• Primary
• Secondary
WINS servers (optional):
• Primary
• Secondary
• LMHOSTS file (optional)
Database server (choose one):

• Publisher
• Subscriber
If you are configuring a subscriber server, supply the
username and password of the publishing database
server:
• Publisher username
• Publisher password
Backup (choose one or both):

• Server
• Target
New Windows 2000 administrator password

Postinstallation Procedures
After you complete the Cisco CallManager software installation, the installation wizard will
prompt you to change all passwords used in the Cisco CallManager cluster. These passwords
should be the same on all servers you install into the cluster. In addition, many supporting services
are running on your server that you might be able to stop. The fewer services you have running on
your server, the more server resources you will have available to support the IP telephony network.
In addition, running more services on the Cisco CallManager server introduces more security
vulnerabilities for the underlying Windows operating system. You should stop all of the following
services on both the Publisher and Subscriber servers in your cluster and set them to manual-start
status unless they are otherwise needed on the system:
■ DHCP client
■ Fax service
■ FTP Publishing Service
■ Smart Card (unless using security tokens)
■ Smart Card Helper
■ Computer browser
■ Distributed File System
■ License Logging Service
By default, the installation wizard configures all Subscribers with Internet Information Server
(IIS) Services running. This allows you to make changes to the cluster by accessing the web
interface on your subscriber servers. Even though you are accessing the web interface on the
Subscriber server, the changes are actually being made on the Publisher server (because it has the
only writable copy of the database). In addition, allowing the web services to run on all Subscriber
servers introduces more security risk as there are now multiple points of access for the Cisco
CallManager administration interface. Because of this, it is usually best to save the Subscriber
resources by stopping the web services on all servers except the Publisher. You can accomplish
this by stopping the following services:
■ Microsoft Internet Information Server (IIS) Admin Service
■ World Wide Web Publishing Service
You can stop all of these services through the Windows 2000 Services console. To open this
console, click Start > Programs > Administrative Tools > Services. When the console opens,
Windows lists all services in alphabetic order. Right-click on the service you want to disable and
choose Properties. In the Properties window shown in Figure 3-1, use the drop-down box to select
either a Manual startup or to Disable the service. These will take effect the next time you reboot
the Cisco CallManager server. You can also choose to stop the service without restarting the server
from this page.
TIP Because Cisco CallManager requires these services to be active when upgrading to new
Cisco CallManager versions, setting them to a state of Manual is suggested.
Figure 3-1 Windows 2000 Service Properties
Activating Cisco Unified CallManager Services

If you are installing Cisco CallManager for the first time, all services that are required to run Cisco
CallManager automatically install on the system; however, none of the services are activated at the
completion of the installation (except for the Cisco Database Layer Monitor service). Cisco
CallManager Serviceability provides a web-based Service Activation tool that is used to activate
or deactivate multiple services and to select default services to activate.
It is recommended that you activate only the required components for each server in the cluster.
Each component that you activate adds to the server load.
If you are upgrading Cisco CallManager, the services that you have already started on your system
will start after the upgrade.
Each service performs specific functions for the IP telephony network. Some services might need
to run on a single Cisco CallManager server in a cluster; other services might need to run on all of
the Cisco CallManager servers in the cluster.
CAUTION Be sure to activate at least the Cisco CallManager service before you apply any
configuration to your Cisco CallManager server. Failure to do so can lead to unpredictable
results, potentially leading to a server reinstall.
The following information briefly describes each available Cisco CallManager service:
■ Cisco CallManager Service—Allows the server to participate in telephone registration, call

processing, and other Cisco CallManager functions. Cisco CallManager Service is the core
service of the Cisco CallManager platform.
■ Cisco TFTP—Activates a TFTP server on Cisco CallManager. The TFTP service delivers
Cisco IP Phone loads and configuration files to IP Phones, along with streamed media files,
such as music on hold (MOH) and ring files.
■ Cisco Messaging Interface—Allows Cisco CallManager to interface with a Simplified

Message Desk Interface (SMDI)-compliant, external voice-mail system.
■ Cisco IP Voice Media Streaming Application—Allows Cisco CallManager to act as a

Media Termination Point (MTP), a conference bridge, a music on hold (MOH) server, and an
annunciator. The voice network uses these media resources for feature functionality.
■ Cisco CTIManager—Allows Cisco CallManager to support computer telephony integration

(CTI) services and provides Telephony Application Programming Interface (TAPI) or Java
Telephony Application Programming Interface (JTAPI) client support. Cisco CTIManager
allows you to use applications such as Cisco IP SoftPhone.
■ Cisco Telephony Call Dispatcher—Distributes calls to multiple telephone numbers (hunt

groups). Cisco WebAttendant, Attendant Console, and Auto Attendant require Cisco
Telephony Call Dispatcher (TCD).
■ Cisco MOH Audio Translator—Allows Cisco CallManager to convert MP3 or WAV audio
files into voice codec format used for MOH.
■ Cisco Real-Time Information Server (RIS) Data Collector—Allows Cisco CallManager

to write trace and alarm file information to a database, Microsoft Event Viewer, or alert an
SNMP server.
■ Cisco Database Layer Monitor—Monitors aspects of the Microsoft SQL 2000 database, as
well as call detail records (CDRs).
■ Cisco CDR Insert—Allows Cisco CallManager to write CDRs to the local database and
replicates CDR files to the Microsoft SQL publisher at a configured interval.
■ Cisco CTL Provider—Works with the Cisco Certificate Trust List (CTL) client to change the
security mode for the cluster from nonsecure to secure (called mixed mode).
■ Cisco Extended Functions—Provides support for some Cisco CallManager features,

including Cisco Call Back and Quality Report Tool (QRT).
■ Cisco Serviceability Reporter—Generates the following daily reports: Device Statistics,

Server Statistics, Service Statistics, Call Activities, and Alert.
■ Cisco WebDialer—Provides click-to-dial functionality by using a web page or a desktop

application.
■ Cisco IP Manager Assistant—Allows Cisco CallManager to support the Cisco IP Manager

Assistant (IPMA), an application designed to allow a receptionist and manager to support
special functionality between their phones.
■ Cisco CallManager Extension Mobility—Allows Cisco CallManager to support extension

mobility functions for roaming users. By using extension mobility features, you can assign
users roaming phone profiles that activate when they log in to a phone. Works similarly to the
roaming profiles in a Windows-based domain environment.
■ Cisco Certificate Authority Proxy Function—Working in conjunction with the Cisco

Certificate Authority Proxy Function (CAPF) application, the Cisco CAPF service can
perform the following tasks, depending on your configuration:
— Issues locally significant certificates to supported Cisco IP Phone models

— Requests certificates from third-party certificate authorities on behalf of supported
Cisco IP Phone models
— Upgrades existing certificates on the phones
— Retrieves phone certificates for troubleshooting
— Deletes locally significant certificates on the phone
You must activate the Cisco CallManager services from the Service Activation web interface
rather than the Windows 2000 Services control panel. To access this interface, perform the
following steps:
Step 1 Open Internet Explorer, and go to https://<CallManager_IP_Address>/

ccmadmin. The <CallManager_IP_Address> is the IP address of the Cisco
CallManager server that is running IIS web services. Enter the
administrative username and password information.
Step 2 From the Application menu, choose Cisco CallManager Serviceability.
The Cisco CallManager Serviceability interface appears.
Step 3 From the Tools menu, choose Service Activation. A window similar to the
window shown in Figure 3-2 appears.
Figure 3-2 Cisco CallManager Service Activation Interface
Step 4 Click the server that you want to configure from the Servers column. Next
click the services that you want to activate, and click the Update button.
(You will experience a slight delay.) The Service Activation window will
refresh when the process is complete.
Upgrading Prior Cisco Unified CallManager Versions 45
TIP The method shown is just one way to access the Cisco CallManager Serviceability pages.
If you are working on the Cisco CallManager itself, you can get there quicker by using the
Windows 2000 Start menu (Start > Programs > Cisco CallManager > Cisco CallManager
Serviceability) or by accessing https://<CallManager-IP-Address>/CCMService.
CAUTION Remember to activate the Cisco CallManager services from the Service
Activation web interface. Activating the services through the Windows 2000 Services console
will produce unpredictable and unstable results.
When you click the Set Default button in the web interface, the Service Activation tool chooses
the services required to run Cisco CallManager based on a single-server configuration. This is the
bare minimum to have a working Cisco CallManager–based IP telephony network. Because Cisco
highly advises against single-server installations, you will most likely use the Set Default button
in a lab environment.
Upgrading Prior Cisco Unified CallManager Versions

Cisco supports the upgrade to Cisco CallManager 4.1 from Cisco CallManager Release 3.3(4),
3.3(5), 4.0(1), and 4.0(2a). You must first upgrades earlier versions to one of these releases before
you can upgrade to version 4.1.
NOTE Cisco CallManager 4.1(3) is the most recent version available at the time of this
writing. If you are upgrading to a more recent version, refer to the specific upgrade instructions
provided in the documentation for the more recent Cisco CallManager version.
If your server runs a version of Cisco CallManager Release 3.2 or earlier, you must first upgrade
every server in the cluster to the latest version of Cisco CallManager Release 3.3 before you can
upgrade to a version of Cisco CallManager Release 4.1. This is because the database structure and
storage system changes from versions prior to release 3.3. (Earlier versions use SQL 7.0 rather
than SQL 2000.)
Before you perform any upgrade procedures, it is strongly recommended that you install the latest
operating system upgrade and service release, SQL service releases and hotfixes, and Cisco
CallManager service release for the versions that currently run in the cluster. Cisco provides the
service release and corresponding “readme” documentation on Cisco.com. To obtain these
documents, go to http://www.cisco.com/kobayashi/sw-center/sw-voice.shtml (CCO login is
required).
NOTE Cisco releases all service packs, hotfixes, and Windows operating system upgrades
through the CCO website. Updates released by Microsoft should never be applied to the Cisco
CallManager server as they can potentially cause the Cisco CallManager software to become
unstable or to crash. Cisco guarantees that they will release any critical operating system patch
within 24 hours of the Microsoft announcement. Noncritical patches are rolled up into a monthly
update.
Cisco requires that you install Cisco IP Telephony Server Operating System Version 2000.2.6
(available from Cisco CCO) before you upgrade to Cisco CallManager Release 4.1.
CAUTION Because the foundation operating system components might change, the upgrade
procedure will many times require you to back up the database on the Publisher server, reimage
Cisco CallManager completely (using the clean installation method discussed earlier in the
chapter), and then restore the SQL data from backup. If your version of Cisco CallManager
requires this, be absolutely sure to install the newer backup software on the Cisco CallManager
you are upgrading before you initially back up the database. Many times, the backup software
changes between the 3.x and 4.x versions of Cisco CallManager. Because of this, you will
usually find the latest version of the backup software in the /backup folder on the first Cisco
CallManager Installation CD-ROM. If you do not install the new backup software before
performing the Cisco CallManager upgrade, your backup file (and SQL data) might be
unreadable in the new Cisco CallManager version.
Because many of the Cisco CallManager servers are equipped with RAID 1 configurations
(mirrored hard disks), a common upgrade strategy is as follows:
Step 1 Remove the mirrored hard disk from the production Cisco CallManager
server.
Step 2 Boot a lab server using the mirrored hard disk.
Step 3 Perform the Cisco CallManager software upgrade on the mirrored hard
disk.
Step 4 During a scheduled window of downtime, reboot the production server
using the upgraded, mirrored drive.
Step 5 The hard drive running the older Cisco CallManager version will become
the new, backup mirrored drive.
Review Questions 47
By using this upgrade process, you can reduce the amount of voice network downtime to the
amount of time it takes to reboot the Cisco CallManager server on the upgraded drive.
CAUTION The upgrade process described will cause a loss of any database changes
performed from the time you have removed the mirrored drive until the time you reboot your
Cisco CallManager server using the new version. Plan your upgrade windows accordingly.
Summary
This chapter covered the Cisco CallManager installation and upgrade process. When you purchase
a Cisco MCS and Cisco CallManager software, you will receive a number of CDs (and even
DVDs). Of those, you will only need three discs for the installation: the hardware detection CD,
the operating system CD/DVD, and the Cisco CallManager installation CD/DVD. The installation
itself is very automated and disk image-based. It requires very little knowledge of the foundation
Windows 2000 operating system and SQL 2000 database store. After the installation completes,
you should stop any unnecessary services running on your Cisco CallManager and start the Cisco
CallManager services required for your network.
The upgrade process for Cisco CallManager varies wildly depending on the version of Cisco
CallManager you are using. The process can be as simple as inserting the Cisco CallManager
installation disk and following a step-by-step wizard or as complex as a complete reimage of all
servers in the cluster. The key is to make sure that you have backed up all data from the SQL server
with the newest version of the backup software available from the Cisco CallManager installation
CD-ROM or the Cisco website.
Review Questions
1. Cisco CallManager uses which of these operating systems?

a. Linux
b. Windows NT
c. Windows 2000
d. Windows 2003
2. Why is it recommended that you stop IIS on the subscriber servers? (Choose two.)
a. enhances server call processing and redundancy
b. maximizes the number of devices in a cluster
c. disables the use of remote terminal services
d. helps prevent unauthorized access to the server
e. makes more resources available for critical voice services
3. When you first install Cisco CallManager software, which CD-ROM should you use to boot
the server to determine the correct CD-ROM to insert next?
a. Cisco CallManager 4.0 Software Disk
b. Cisco IP Telephony Server Operating System Hardware Detection Disk
c. Cisco CallManager Installation, Upgrade, and Recovery Disk
d. Cisco IP Telephony Server Backup and Restore Disk
e. Cisco Extended Services and Locales Disk
4. If you are not using DNS, what must you configure to resolve server names for the Cisco
CallManager installation process?
a. DHCP
b. backup server
c. LMHOSTS file
d. DNS reverse lookup
5. Which of the following represent Windows services that you are able to stop on both
Publisher and Subscriber servers? (Choose three.)
a. computer browser
b. DHCP client
c. database layer monitor
d. CTL provider
e. FTP Publishing Service
6. If you want to set up a Cisco CallManager as a single-server configuration in a lab
environment, what button can you click on the Service Activation web page to start the
necessary services?
a. Single Server
b. Lab
c. Set Default
d. Minimal Service
Review Questions 49
7. You have just completed the installation of a Cisco CallManager server, entered the new
passwords for all accounts, and restarted. What is the next step you should take?
a. Configure the Cisco CallManager server settings to send IP address information to the
IP Phones rather than hostname.
b. Activate the necessary services through the Service Activation web page.
c. Upgrade the backup software to the latest version.
d. Apply the latest operating system service pack from the Cisco website.
Part II: IPT Devices and Users
Chapter 4 Cisco IP Phones and Other User Devices
Chapter 5 Configuring Cisco Unified CallManager to Support IP Phones
Chapter 6 Cisco IP Telephony Users
Chapter 7 Cisco Bulk Administration Tool

following topics:
■ The basic features of Cisco IP Phones
■ The entry-level Cisco IP Phones and their

features
■ The midrange Cisco IP Phones and their

features
■ The upper-end Cisco IP Phones and their

features
■ The features and functions of additional

Cisco IP telephony endpoints, including
video endpoints, conference stations,
expansion modules for Cisco IP Phones, PC-
based Cisco IP Phones, and analog adapters
■ The six steps of the Cisco IP Phone startup

process in the correct order
■ The two audio codecs that are supported by

Cisco IP Phones
CHAPTER 4
Cisco IP Phones and
Other User Devices
Thus far, you have focused on the Cisco Unified CallManager server as the management system
of the Cisco IP telephony network. It is now time to focus on the devices that Cisco CallManager
controls. An important task of implementing and supporting an IP telephony deployment is
managing the end-user devices, or endpoints. You should be able to distinguish among the
various Cisco IP telephony end-user devices that you might encounter during the course of
deploying and administering a Cisco IP telephony network.
This chapter explains the various models of Cisco IP Phones and how they work within a Cisco
IP telephony solution. You will learn the basic features of Cisco IP Phones, analog adapters, and
conference stations; the IP Phone power-up and registration process; and the audio coders-
decoders (codecs) that are supported by Cisco IP Phones.
Cisco IP Phones
To the user, the telephone is the most visible component of the voice communications network.
Cisco IP Phones are next-generation, intelligent communication devices that deliver essential
business communications. Fully programmable, the growing family of Cisco IP Phones
provides the most frequently used business features.
The majority of Cisco IP Phones provide the following features:
■ Display-based user interface
■ Straightforward user customization
■ Inline Power over Ethernet (PoE)
■ Support for the G.711 and G.729 audio codecs
Each Cisco IP Phone provides toll-quality audio. Because it is an IP-based phone, you can
install it in any location on a corporate local or wide-area IP network. Some corporations have
even made the Cisco CallManager publicly available on the Internet (with appropriate firewall
platforms in place), allowing Cisco IP Phones to register from any location that has an Internet
connection.
54 Chapter 4: Cisco IP Phones and Other User Devices
Entry-Level Cisco IP Phones

Cisco has produced a number of low-cost, entry-level IP Phones for a variety of business
functions. Depending on user requirements, these IP Phones may function well for employees or
for use only in public areas, such as lobbies or break rooms. Figure 4-1 shows the four entry-level
Cisco IP Phones available at the time of this writing.
Figure 4-1 Entry-Level Cisco IP Phones

Cisco IP Phone
Cisco IP Phone 7902G Cisco IP Phone 7905G 7910G+SW Cisco IP Phone 7912G
Entry-level Cisco phones provide the following common features:
■ Display-based user interface (except Cisco IP Phone 7902G)
■ G.711 and G.729 codec
■ Single line (directory number [DN])
■ Cisco inline power, powered patch panel, or local power option support via a power cube (the
same power supply as the Cisco IP Phone 7910, 7940, or 7960)
■ Visual message waiting indicator (MWI)
■ One-way speakerphone (no built-in microphone) and no headset port
The following list briefly describes the major features of each entry-level Cisco IP Phone:
■ Cisco IP Phone 7902G—The Cisco IP Phone 7902G is a single-line, entry-level, no-display

business phone with fixed feature keys that provide one-touch access to the redial, transfer,
conference, and voice-mail access features. The following briefly describes the major features
of the Cisco IP Phone 7902G:
— Fixed features: redial, transfer, conference, messages

— Hard Hold key
— Single 10-Mbps RJ-45 connection (no internal Ethernet switch)
Cisco IP Phones 55
NOTE Even though it might appear from the figure that the Cisco IP Phone 7902G has an
LCD display, Cisco has equipped this low-end IP Phone with a piece of paper for a user to label
key buttons and phone numbers.
■ Cisco IP Phone 7905G and Cisco IP Phone 7912G—The Cisco IP Phone 7905G provides
single-line access and four interactive softkeys that guide a user through call features and
functions via the pixel-based liquid crystal display (LCD). Use this IP Phone for employees
who do not need a full-featured phone or for a common area such as a hallway, manufacturing
floor, break room, reception space, or office cubicle. The Cisco IP Phone 7912G includes an
integrated Ethernet switch that provides LAN connectivity to a collocated PC (allowing only
a single cable drop to each location). The following briefly describes the major features of the
Cisco IP Phone 7905G and Cisco 7912G:
— Pixel-based display (approximately five lines plus softkeys (on-screen feature

buttons) and date, time, and menu title
— Hard Hold key
— Access to all standard IP Phone features through four physical buttons able to access
multiple on-screen softkeys
— Support for limited Extensible Markup Language (XML) script processing
— Support for Cisco Skinny Client Control Protocol (SCCP), H.323 version 2 (Cisco
7905G only), and Session Initiation Protocol (SIP; compliant with RFC 2543)
■ Cisco IP Phone 7911G—The Cisco IP Phone 7911G provides identical features as the Cisco
7912G with the following differences:
— Support for IEEE 802.3af and Cisco proprietary inline power standards
— Enhanced memory and XML application support similar to the Cisco IP Phone 7970
— Support for enhanced security features
■ Cisco IP Phone 7910G+SW—The Cisco IP Phone 7910G+SW is for common-use areas that
require only basic features, such as dialing out, accessing 911, and intercom calls. Locations
that might benefit from these limited features include lobbies, break rooms, and hallways. The
Cisco IP Phone 7910G+SW includes a two-port switch for use in applications where you
require basic IP Phone functionality and a collocated PC. The following briefly describes the
major features of the Cisco 7910G+SW:
— Ability to handle low-to-medium telephone usage

— Single line with call waiting
— Display area of 2 x 24 inches (5.08 x 60.96 cm)
— Cisco 10BASE-T/100BASE-T, two-port FastEthernet switch

— Basic features: line, hold, transfer, settings, messages, conference, forward, speed
dial, redial
— Adjustable foot stand (flat to 60 degrees)
— Basic and optional wall mounting
NOTE Cisco has positioned the Cisco IP Phone 7912G to replace the Cisco IP Phone
7910G+SW. The Cisco 7910+SW should reach End-of-Life (EOL) in January 2006.
Midrange and Upper-End Cisco IP Phones

Cisco designed the IP Phones 7940G, 7941G and 7941G-GE, 7960, 7961G and 7961G-GE,
7970G and 7971G-GE, and 7985G to meet the demand for a corporate-level, full-featured IP
Phone for medium-to-high telephone use. Figure 4-2 shows the midrange and upper-end Cisco IP
Phones available at the time of this writing.
Figure 4-2 Midrange and Upper-End Cisco IP Phones

Cisco IP Phone 7941G Cisco IP Phone 7961G Cisco IP Phone 7970G
Cisco IP Phone 7985G
A description of features that are common to all midrange and upper-end IP Phones follows:
Cisco IP Phones 57
■ Multiline capability
■ Large pixel-based displays, which allow for the inclusion of XML and future features
■ Integrated two-port FastEthernet or Gigabit Ethernet switch
■ Built-in headset connection and quality full-duplex speakerphone (does not come with a
headset)
■ Information key for “online” help with features
■ A minimum of 24 user-adjustable ring tones
■ Adjustable foot stand (flat to 60 degrees) for desktop use or appropriate kit included for wall
mounting
■ SCCP and SIP support
■ XML service support
■ An EIA/TIA-232 port for options, such as line expansion and security access
NOTE The 79XXG-GE IP Phone models offer features identical to the 79XXG models with
the exception of the integrated two-port switch. The 79XXG-GE models have integrated, two-
port 10/100/1000 switchports, whereas the 79XXG models have integrated, two-port 10/100
switchports.
Additional details of these phone models follow:
■ Cisco IP Phone 7941G and 7941G-GE—The Cisco IP Phone 7941G and 7941G-GE is for
medium traffic and has these features:
— Ideal for transaction workers who use cubicle phones

— Two lines and programmable feature buttons, and four interactive softkeys
— PoE compatible (Cisco prestandard PoE and IEEE 802.3af industry standard)
— Increased memory for advanced applications
— Higher resolution display
— Backlit screen and line keys
■ Cisco IP Phone 7961G and 7961G-GE—The Cisco IP Phone 7961G and 7961G-GE is for
high or busy telephone traffic and has these features:
— Ideal for professionals or managers

— Six lines and programmable feature buttons, and four interactive softkeys
— Increased memory for advanced applications
— Higher resolution display
— Backlit screen and line keys
■ Cisco IP Phone 7970G—The Cisco IP Phone 7970G is one of Cisco’s current high-end
devices featuring a high resolution, full-color display that can make a passerby look twice. It
has the following features:
— Ideal for executives, decision makers, and high-visibility areas of your organization
— Eight lines and programmable feature buttons, and five interactive softkeys
— Larger, color display with touch-screen capabilities
— 3.5-mm stereo jack sockets for connection to PC-style speakers or headphones, and
microphone
— Advanced XML development platform for more dynamic applications
■ Cisco IP Phone 7971G-GE—The Cisco IP Phone 7971G looks identical to the 7970G with
the following differences:
— Delivers Gigabit Ethernet bandwidth to the desktop with an integrated 10/100/1000

Mbps switch
— “Feature identical” to the Cisco IP Phone 7970G
— PoE compatible (IEEE 802.3af PoE only; does not support Cisco prestandard PoE)
■ Cisco IP Phone 7980G—The Cisco 7980G harnesses next-generation video technology
offering the following features:
— Ideal for executives, decision makers, and high-visibility areas of your organization
(such as conference rooms)
— Provides a desktop video phone making instant, face-to-face communication
possible using the H.264 video codec
— Offers an integrated, two-port 10/100 Ethernet switch
— Enables digital clarity video
Cisco IP Phones 59
— PoE compatible (IEEE 802.3af PoE only; does not support Cisco prestandard PoE)
TIP Cisco has recently upgraded the 7940G and 7960G IP Phones to the 7941G and 7961G.
The primary features Cisco added are the support for the IEEE 802.3af PoE standard, higher
resolution and backlit display, multiple color line buttons (to support different line states), and
advanced XML application support. All other features remain the same.
Additional Cisco IP Telephony Endpoints

Cisco provides a complete portfolio of IP endpoints to meet business needs for conferencing,
wireless voice communications, PC-based voice calls, and connecting analog devices to the VoIP
network. Figure 4-3 shows the additional IP telephony endpoints available at the time of this
writing.
Figure 4-3 Additional IP Telephony Endpoints
Cisco VT Advantage Cisco Conference Station 7936 Cisco IP Communicator
Cisco ATA 186 and 188 Cisco 7914 Expansion Module Cisco Wireless IP Phone 7920
These products are as follows:
■ Cisco VT Advantage—Cisco VT Advantage is a video telephony solution consisting of the

Cisco VT Advantage software application and Cisco VT Camera, a video telephony Universal
Serial Bus (USB) camera. With the Cisco VT Camera attached to a PC that is connected to
the switch port of a Cisco IP Phone, users can place and receive video calls on their enterprise
IP telephony network. Users make calls from their Cisco IP Phones using familiar phone
interfaces, but calls are enhanced with video on a PC, without requiring any extra button
pushing or mouse clicking. When registered to Cisco CallManager, the Cisco VT Advantage–
enabled IP Phone has the features and functionality of a full-featured IP videophone. System
administrators can provision a Cisco IP Phone with Cisco VT Advantage as they would any
other Cisco IP Phone, which can greatly simplify deployment and management.
■ Cisco IP Conference Station 7936—The Cisco IP Conference Station 7936 is a full-

featured, IP-based, full-duplex, hands-free conference phone for use on desktops, in offices,
and in small-to-medium conference rooms. The Cisco IP Conference Station 7936 offers
external microphone ports, optional external microphone kit, audio-tuned speaker grill, and a
backlit LCD display. The optional microphone kit includes two microphones with 6-foot
(1.8288-m) cords so that you can place the microphones across a 12-foot (3.6576-m) area,
effectively expanding to a suggested conference room size of 20 x 30 feet (6.096 x 9.144 m).
The backlit LCD display improves visibility in low-light conditions. The display font size is
also adjustable for improved distant viewing. The Conference Station 7936 does not support
inline power and requires an external power supply at this time.
■ Cisco IP Communicator—The Cisco IP Communicator is a Microsoft Windows software–

based application that delivers enhanced telephony support through personal computers. This
application endows computers with the functionality of Cisco IP Phones and provides high-
quality voice calls on the road, in the office, or from wherever users might have access to the
corporate network. Cisco IP Communicator has an intuitive design, is easy to use, and delivers
convenient access to a host of features, including eight lines and five softkeys. Cisco IP
Communicator offers handset, headset, and high-quality speakerphone modes. The
application appears as an on-screen Cisco 7970 IP Phone. Cisco plans to add video support
to the IP Communicator by the time this publication reaches press.
■ Cisco Analog Telephone Adaptor (ATA) 186 and 188—The Cisco ATA 186 and Cisco ATA
188 interface analog devices with your IP-based telephony network. These adapters are useful
for customers who have existing analog devices, such as fax machines or telephones, that they
do not want to replace after they have migrated to VoIP. The Cisco ATA 186 provides two
voice ports, each with its own independent telephone number, and a single 10BASE-T
Ethernet port for network connectivity. The Cisco ATA 188 provides two voice ports and two
10/100-Mbps Ethernet connections, which allows for network connectivity and the ability to
collocate a network device with the analog voice equipment.
■ Cisco IP Phone 7914 Expansion Module—The Cisco IP Phone 7914 Expansion Module
extends the capabilities of the Cisco IP Phone 7960G, 7961G, 7970G, and 7971G-GE with
additional buttons and an LCD display. The Cisco IP Phone 7914 helps administrative
assistants and others who must monitor, manage, and cover the various status of a number of
calls beyond the line capability of the standard Cisco IP Phone models. This expansion
module enables you to add 14 buttons to the existing buttons of the Cisco IP Phone 7960G,
Cisco IP Phones 61
7961G, 7970G, and 7971G-GE, increasing the total number of line and speed dial buttons
available. You can use up to two Cisco 7914 Expansion Modules on each of the Cisco IP
Phones listed previously.
TIP The 7914 Expansion Module cannot receive inline power and requires a local power
supply. A single power supply can power both 7914 expansion modules attached to an IP Phone.
■ Cisco Wireless IP Phone 7920—The Cisco Wireless IP Phone 7920 is an easy-to-use IEEE
802.11b wireless IP Phone that provides comprehensive voice communications in
conjunction with Cisco CallManager and the Cisco Aironet 1200, 1100, 350, and 340 Series
of Wi-Fi (IEEE 802.11b) access points and their related security standards (such as TKIP/
MIC, WPA, LEAP, and so on). The Cisco Wireless IP Phone 7920 is designed for ease of use,
with a pixel-based display to access calling features and two softkeys, support for up to six
directory numbers, a four-way rocker switch, a Hold key, a Mute key, and a Menu key that
allows quick access to information such as directories, call history, and phone settings. The
phone also offers XML support on models with a recent firmware upgrade.
Table 4-1 provides a feature-by-feature comparison of the various Cisco IP Phone models. This
table comes from “Cisco IP Telephony Solution Reference Network Design (SRND) for Cisco
CallManager 4.0 and 4.1” (http://www.cisco.com/en/US/products/sw/voicesw/ps556/
products_implementation_design_guide_book09186a00805fdb7b.html) and is printed with
permission from Cisco Systems.
Table 4-1 IP Phone Features
Feature 7902G 7905G 7910G 7910 +SW 7912G 7920G 7935G, 7940G 7960G 7970G
7936G
Ethernet Y1 Y1 Y1 Y2 Y2 N Y3 Y2 Y2 Y2
Connection
Ethernet Switch N N N Y Y N N Y Y Y
Cisco Power- Y Y Y Y Y N N Y Y Y
Over-Ethernet
(PoE)
IEEE 802.3af N N N N N N N N N Y
Power-Over-
Ethernet (PoE)
Localization N Y N N Y N N Y Y Y
Directory 1 1 1 1 1 6 1 2 6 8
Number
Liquid Crystal N Y Y Y Y Y Y Y Y Y
Display
continues
Table 4-1 IP Phone Features (Continued)
7936G
Caller ID N Y Y Y Y Y Y Y Y Y
Call Waiting N Y Y Y Y Y Y Y Y Y
Caller ID on N Y Y Y Y Y Y Y Y Y
Call Waiting
Call Hold Y Y Y Y Y Y Y Y Y Y
Call Transfer Y Y Y Y Y Y Y Y Y Y
Call Forward Y Y Y Y Y Y Y Y Y Y
Auto-Answer Y Y N N Y Y N Y Y Y
Ad Hoc Y Y Y Y Y Y Y Y Y Y
Conference
Meet-Me N Y Y Y Y Y Y Y Y Y
Conference
Call Pickup N Y Y Y Y Y Y Y Y Y
Group Pickup N Y Y Y Y Y Y Y Y Y
4 Y4 Y4 Y4 Y4
Redial Y Y Y Y Y Y
Speed Dial Y Y Y Y Y Y N Y Y Y
On-hook N Y Y Y Y Y Y Y Y Y
Dialing
Voice Mail Y Y Y Y Y Y N Y Y Y
Access
Message Y Y Y Y Y Y N Y Y Y
Waiting
Indicator
(MWI)
Video call N N N N N N N Y Y Y
Survivable Y Y Y Y Y Y Y Y Y Y
Remote Site
Telephony
(SRST) Support
Music on Hold Y Y Y Y Y Y5 Y Y Y Y
(MoH)
Speaker N Y6 Y6 Y6 Y6 N Y Y Y Y
Headset Jack N N N N N Y7 N Y Y Y
Mute N N Y Y N Y Y Y Y Y
Cisco IP Phones 63
7936G
Multilevel Y Y Y Y Y N N Y Y Y
Precedence and
Preemption
(MLPP)
Barge N N N N N N N Y Y Y
cBarge N Y N N Y N N Y Y N
Disable General Y Y Y Y Y N N Y Y Y
Attribute
Registration
Protocol
(GARP)
Signaling and N N N N N Y8 N Y Y Y
Media
Encryption
Signaling N N N N N N N Y Y Y
Integrity
Manufacturing- N N N N N N N N N Y
Installed
Certificate
(X.509v3)
Field-Installed N N N N N N N Y Y N
Certificate
Third-Party N Y N N Y N N Y Y Y
XML Service
External N N N N N N N N N Y
Microphone and
Speaker
Dial Plan N N N N N N N N N N
Mapping
Signaling 0x60 0x68 0x68 0x68 0x60 0x68 0x68 0x60 0x60 0x60
Packet ToS
Value Marking
Media Packet 0xB8 0xB8 0xB8 0xB8 0xB8 0xB8 0xB8 0xB8 0xB8 0xB8
ToS Value
Marking
Skinny Client Y Y Y Y Y Y N Y Y Y
Control
Protocol
(SCCP)
continues
7936G
Session N Y N N Y N N Y Y N
Initiation
Protocol (SIP)
H.323 N Y N N N N Y N N N
Media Gateway N N N N N N N Y Y N
Control
Protocol
(MGCP)
G.711 Y Y Y Y Y Y Y Y Y Y
G.723 N Y N N N N N N N N
G.726 N Y N N N N N N N N
G.729 Y Y Y Y Y Y Y Y Y Y
Voice Activity Y Y Y Y Y Y Y Y Y Y
Detection
(VAD)
Comfort Noise Y Y Y Y Y Y Y Y Y Y
Generation
(CNG)
1 One 10 Base-T
2 Two 10/100 Base-T
3 One 10/100 Base-T
4 Last Number Redial
5 Supports only unicast MoH
6 One-way listen mode
7 The only supported headset for the Cisco IP Phone 7920 is one with a 2.5 mm jack, available at
http://www.cisco.getheadsets.com.
8 Signaling and Media Encryption are available with Static WEP and LEAP security configurations.
IP Phone Startup Process

Figure 4-4 provides an overview of the startup process for a Cisco IP Phone if you are using a
Cisco Catalyst switch that is capable of providing Cisco prestandard Power over Ethernet (PoE).
Understanding the boot process of a Cisco IP Phone is critical to your abilities to troubleshoot an
IP telephony network.
IP Phone Startup Process 65
Figure 4-4 Cisco IP Phone Startup Process

Cisco
DHCP CallManager Cisco TFTP
4 1 3
IP
2
1. IP Phone obtains power from the switch

2. Phone loads stored image
3. Switch provides VLAN information to IP Phone
4. Phone sends DHCP request; receives IP
information and TFTP server address
5. IP Phone gets configuration from TFTP server
6. IP Phone registers with Cisco CallManager server
The following list explains the normal boot process of a Cisco IP Phone:
Step 1 Obtain power from the switch—When a Cisco IP Phone is connected to

a Cisco Catalyst switch model that is capable of providing inline power
(called Power over Ethernet [PoE]), the switch automatically detects an
unpowered phone and sends power down the Ethernet cable to the IP
Phone. The details of how the switch detects an unpowered IP Phone and
how it delivers power to the IP Phone are covered in Chapter 8, “Cisco
Catalyst Switches.”
Step 2 Load the stored phone image—The Cisco IP Phone has nonvolatile flash
memory in which it stores firmware images and user-defined preferences.
At startup, the phone runs a bootstrap loader that loads a phone image
stored in flash memory. Using this image, the phone initializes its software
and hardware.
Step 3 Configure the VLAN—After the IP Phone receives power and boots, the
switch sends a Cisco Discovery Protocol version 2 (CDPv2) trigger packet
to the IP Phone. CDP packets that are “triggered” do not need to wait for
the default 60-second CDP hello interval. This CDPv2 packet provides the
IP Phone with voice VLAN information, if you have configured that feature
(voice VLANs are also discussed in Chapter 8). The IP Phone will then tag
all its traffic with the appropriate voice VLAN information.
Step 4 Obtain the IP address and TFTP server address—Next, the IP Phone
broadcasts a request to a DHCP server. The DHCP server responds to the
IP Phone with a minimum of an IP address, a subnet mask, the default
gateway, and the IP address of the Cisco TFTP server.
TIP A DHCP server can give a Cisco IP Phone the location of a TFTP server in
two different ways:
■ DHCP Option 66—Gives the Cisco IP Phone the hostname of the TFTP
server (requires DNS resolution)
■ DHCP Option 150—Gives the Cisco IP Phone the IP address of the TFTP
server
The DHCP administrator must add one or both of these options to the DHCP scope
used for the Cisco IP Phones. If you cannot use the DHCP server to distribute this
information, you must hard code the IP address of the TFTP server manually on
each IP Phone. Cisco recommends using DHCP Option 150 because it allows you
to configure the IP Phone to use multiple TFTP server addresses for redundancy.
Step 5 Contact the TFTP server for configuration—The IP Phone then contacts
the Cisco TFTP server. The TFTP server has configuration files (.cnf file
format or .cnf.xml) for telephony devices, which define parameters for
connecting to Cisco CallManager. The TFTP server sends the configuration
information for that IP Phone, which contains an ordered list of up to three
Cisco CallManagers. In general, any time that you make a change in Cisco
CallManager that requires a phone (device) to be reset, a change has been
made to the configuration file of that phone. If a phone has an XML-
compatible load, it requests an XMLDefault.cnf.xml configuration file;
otherwise, it requests a .cnf file. If you have enabled auto-registration in
Cisco CallManager, the phones access a default configuration file
(sepdefault.cnf.xml) from the TFTP server. If you have manually entered
the phones into the Cisco CallManager database, the phone accesses a
.cnf.xml file that corresponds to its device name. The .cnf.xml file also
contains the information that tells the phone which image load that it should
be running. If this image load differs from the one that is currently loaded
on the phone, the phone contacts the TFTP server to request the new image
file, which is stored as a .bin or .sgn (secured load) file.
Cisco IP Phone Codec Support 67
Step 6 Register with Cisco CallManager—After obtaining the file from the
TFTP server, the phone attempts to make a TCP connection to a Cisco
CallManager, starting with the highest-priority Cisco CallManager in its
list. After the phone registers with the Cisco CallManager, it receives an
extension number and becomes operational.
Cisco IP Phone Codec Support

Before a VoIP device is able to stream audio, the analog audio signal must be converted to a
digitized format. This is accomplished by an audio codec, which digitizes the audio input at the
transmitting end and converts the digital stream back to analog audio at the receiving end.
Because converted audio streams can consume a significant amount of bandwidth over slower
WAN connections, many of the audio codecs also provide a level of compression, which can
considerably reduce the bandwidth that they consume. However, different types of compression
can cause degraded voice quality, which is why the different audio codecs offer different
compression algorithms and quality levels.
The International Telecommunication Union Telecommunication Standardization Sector (ITU-T)

standards committee specifies several standards (called recommendations) for audio codecs. Cisco
IP Phones natively support two primary codecs: G.711 and G.729. The G.711 and G.729 codecs
deliver relatively equal sound quality (both considered toll quality), with G.711 scoring slightly
higher than G.729 in a mean opinion score (MOS) test, which is a rating of voice quality. Because
of this similarity, some network administrators choose to operate an entirely G.729-based
network, whereas others choose to implement G.729 over the WAN and G.711 on the LAN. A
G.711 conversation consumes 64 Kbps of network bandwidth (not including packet header
overhead), whereas a G.729 conversation consumes 8 Kbps of network bandwidth (not including
packet header overhead).
Although this configuration is ideal for many network environments, you might eventually
encounter a codec mismatch. A codec mismatch occurs when two devices cannot negotiate a
common codec or when the network administrator has forbidden the use of their common codec,
such as using G.711 over the WAN. Regardless of the cause, you now have a need for transcoding.
Transcoding resources perform conversions between the audio codecs. These resources are often
costly and can introduce significant delay and quality degradation into your IP telephony network.
When designing a voice network, you should attempt to limit the amount of transcoding that takes
place between devices.
Although it is not mentioned often, the Cisco 7900 IP Phone models support the Wideband codec.
This is a Cisco proprietary codec that consumes four times the amount of bandwidth as the G.711
codec, using 256 kbps for the audio payload. The codec delivers extremely high-quality 16-bit
audio that gives a clean, crisp sound to voice and music on hold.
Summary
This chapter covered IP Phones and other end devices in a Cisco IP telephony network. Although
there are many similarities between the Cisco IP Phone models, there are also marked differences
as you move through the ever-expanding line of IP Phone models. Each of these phones fit a niche
in businesses worldwide. The entry-level Cisco IP Phones include the 7902G, 7905G, 7910G+SW,
and 7912G. The midrange and upper-end Cisco IP Phones include the 7941G and 7941G-GE,
7961G and 7961G-GE, 7970G and 7971G-GE, and 7985. The additional IP telephony devices
include the Cisco Conference Station 7936, Cisco IP Communicator, Cisco ATA 186 and 188,
Cisco 7920 wireless phone, and Cisco 7914 Expansion Module.
Review Questions
1. Which of the following two codecs do current Cisco IP Phones support? (Choose two.)
a. G.711
b. G.723
c. G.728
d. G.729
2. Which of the following PoE standards does the Cisco IP Phone 7961G support?
(Choose two.)
a. Cisco Pre-Standard Inline Power
b. 802.11af
c. 802.11b
d. 802.3af
3. Which of the following devices has a built-in Ethernet switch? (Choose three.)
a. Cisco IP Phone 7905
b. Cisco IP Phone 7912
c. Cisco IP Phone 7970
d. Cisco ATA 186
e. Cisco ATA 188
Review Questions 69
4. Which of the following Cisco IP Phones provides video-processing capabilities without the
VT Advantage software?
a. Cisco 7970
b. Cisco 7971
c. Cisco 7980
d. Cisco 7936
5. Place these steps in the order that an IP Phone must go through before it can register with
Cisco CallManager.
a. Obtain power.
b. Get configuration from TFTP server.
c. Load firmware image.
d. Obtain VLAN information.
6. Which of the following options can you configure on a DHCP server to deliver the IP address
of a TFTP server to a Cisco IP Phone?
a. Option 6
b. Option 66
c. Option 150
d. Option 170
7. Most Cisco IP Phones provide built-in ______ processing, which allows them to read data
stored in this industry standard format.
a. XML
b. HTTP
c. TXT
d. SQL
8. What is the difference between the Cisco IP Phone 7960 and Cisco IP Phone 7961?
(Choose two.)
a. The 7961 provides a 10/100/1000 built-in switch, whereas the 7960 provides only a 10/
100 built-in switch.
b. The 7961 adds support for the VT Advantage.
c. The 7961 supports six extensions, whereas the 7960 only supports four extensions.
d. The 7961 supports both the Cisco prestandard and 802.3af inline power, whereas the
7960 supports only the Cisco prestandard.
9. A company wants to migrate to a VoIP network, but it has an expensive all-in-one voice-fax-
copier device that it wants to connect to the VoIP network. Which device would you
recommend?
a. Cisco Conference Station 7936
b. Cisco IP SoftPhone
c. Cisco ATA 188
d. Cisco 7905G
e. Cisco 7914 Expansion Module
10. A firm wants to install a Cisco IP Phone in the lobby area. The IP telephony network runs only
the G.729a codec. The IP Phone should support a single line and all standard features. An
LCD display is not required because cost is a concern. Which IP Phone would best meet these
requirements?
a. Cisco IP Phone 7902G
b. Cisco IP Phone 7905G
c. Cisco IP Phone 7910G+SW
d. Cisco IP Phone 7912G
following topics:
■ Configuring Cisco Unified CallManager to

support Cisco IP Phone registration and
communication within the cluster
■ Required elements to create a device pool
■ Applying required settings to a Cisco IP

Phone through a device pool
■ Manually and automatically registering

Cisco IP Phones into the SQL database
CHAPTER 5
Configuring Cisco Unified
CallManager to Support
IP Phones
After reading the previous chapter, you decided to purchase ten Cisco IP Phones, one of each
model. You receive them in the mail, unwrap them (filling the air with the scent of fresh
cellophane), and set them on the desk. Nothing happens. Now what? Now you must configure
the Cisco CallManager to support the IP Phones. Without the Cisco CallManager, these devices
are nothing more than expensive paperweights.
This chapter describes the Cisco CallManager configuration to support Cisco IP Phones. You
will learn how to configure Cisco CallManager to manually and automatically add IP Phones
and assign directory numbers (DNs). You will also learn how to configure device pools to
provide a convenient way to define a set of common characteristics that can be assigned to
devices, such as date or time zone, codec use, and other functionalities.
Configuring Intracluster IP Phone Communication

Unbelievably, you can have an entire Cisco IP telephony network deployed in a matter of
seconds using the Cisco CallManager in its base configuration. However, setting up the voice
network correctly can take a considerable amount of planning and configuration. Thankfully,
this chapter shows you how to set up the network manually, and then shows you how to pull it
off automatically in a matter of seconds.
Removing DNS Reliance

Chapter 4 discussed the boot process of a Cisco IP Phone. It is critical to understand that process
to effectively troubleshoot the IP telephony network. Just to review, the following are three of
the critical steps in the process:
1. Cisco IP Phone obtains an IP address from the DHCP server along with the IP address of
the TFTP server.
2. The Cisco IP Phone contacts the TFTP server and downloads its configuration.
3. The Cisco IP Phone registers with the Cisco CallManager.
These steps (along with a few others) were presented in the preceding chapter. Now you might
be wondering, “The Cisco IP Phone downloads its configuration from the TFTP server.
74 Chapter 5: Configuring Cisco Unified CallManager to Support IP Phones
So…where does that configuration come from?” Great question! It comes from you. As you
configure the Cisco CallManager, it will generate a configuration file that it places on the TFTP
server for the IP Phone to download. In that configuration file will be a list of up to three Cisco
CallManagers for the Cisco IP Phone to contact.
One of the most common problems that occur in the field is a Cisco IP Phone failing to register
with the Cisco CallManager because it cannot resolve the Cisco CallManager’s hostname to an IP
address. This is because, by default, the Cisco CallManager puts a list of up to three Cisco
CallManager hostnames the Cisco IP Phone can contact. If you have not configured the IP Phones
for DNS name resolution (or the IP Phones cannot reach the DNS server), the phone registration
will fail.
Because of this, changing the name of the selected server to the IP address of the server in the
Cisco CallManager Administration window is the first step in configuring Cisco CallManager to
support Cisco IP Phones.
NOTE Removing DNS reliance is considered a best-practice step in configuring Cisco

CallManager. If you prefer, you can keep the Cisco CallManager hostnames in the configuration
and still have a functional network.
Renaming the server to the IP address has the following benefits:
■ It allows IP Phones and other devices to find Cisco CallManager on the network without
having to query the Domain Name System (DNS) server to help resolve the server name to
an IP address.
■ It prevents the IP telephony network from failing if the IP Phones lose the connection to the
DNS server.
■ It decreases the time that is required when a device attempts to contact Cisco CallManager.
Complete these steps to eliminate DNS reliance:
Step 1 In Cisco CallManager Administration, choose System > Server. The Find
and List Servers window appears. Click Find to display a list of available
servers.
Step 2 Click a server name. The Server Configuration window appears.
Step 3 Remove the hostname and enter the IP address for the server in the Host
Name/IP Address field. Click Update.
Step 4 Repeat Steps 1-3 for every Cisco CallManager in the cluster.
Configuring Intracluster IP Phone Communication 75
After the Cisco CallManager hostnames are changed to the appropriate IP addresses, the Cisco
CallManager automatically updates the configuration file on the TFTP server, causing the IP
Phones to contact the Cisco CallManagers via the IP address rather than the hostname. Figure 5-1
illustrates the Server Configuration screen.
NOTE There are other places where server names must be changed to IP addresses to fully
remove DNS resolution from the cluster. Most of these can be found in the Service > Enterprise
Parameters options.
Figure 5-1 Cisco CallManager Server Configuration Window
Configuring Device Pools

Device pools provide a convenient way to define a set of common characteristics that can be
assigned to devices, such as IP Phones, instead of assigning individual characteristics to individual
phones. Device pools enable you to simply assign the phone to the device pool so that the phone
automatically inherits the common configuration items. You must configure the device pool for
Cisco IP Phones before adding them to the network.
To create a new device pool, you must first create (or use default settings where applicable) the
following minimal mandatory components:
■ Cisco CallManager group
■ Date/time group
■ Region
■ Softkey template
■ Cisco Survivable Remote Site Telephony (SRST) reference
NOTE The SRST Reference field allows you to specify the IP address of the Cisco SRST
router. Cisco SRST enables routers to provide call-handling support for Cisco IP Phones when
they lose their connection to remote Cisco CallManager installations or when the WAN
connection is down.
These components (except for the SRST reference) are covered later in this chapter. SRST is
covered in Chapter 14, “Implementing Multiple-Site Deployments.”
The device pool combines all of the individual configurations that you have created into a single
entity. You will eventually assign this entity to individual devices, such as IP Phones. This process
will configure these devices with most of the configuration elements that they need to operate
efficiently in your IP telephony network.
For example, configuring 100 different Cisco IP Phones to use the Arizona time zone, the English
language, and the classical-style music on hold (MOH) option is tedious. Instead, you can create
a device pool, named “Arizona,” that configures the correct time zone, language, and MOH, and
you can make a single assignment of the Arizona device pool to each IP Phone.
Complete these steps to create the device pool:
Step 1 Choose System > Device Pool. The Find and List Device Pools window opens.
Step 2 Click the Add a New Device Pool link to open the Device Pool Configuration
window shown in Figure 5-2. Choose, at a minimum, the Cisco CallManager
group, date/time group, region, and softkey template.
NOTE It is a good practice to name your configuration components in Cisco CallManager to

reflect their purpose. For example, you could name the Arizona Device Pool “Arizona_DP.” You
could name the Arizona Date and Time Group “Arizona_DT.” This allows you to easily identify
these items in the Cisco CallManager database.
Figure 5-2 Cisco CallManager Device Pool Configuration Window
You can configure many options through a device pool. Some of these options are shown here;
others are fully explained later in the book. Table 5-1 provides a quick reference for each of the
configuration options in the device pool.
Table 5-1 Cisco CallManager Device Pool Configuration Options
Field Description
Device Pool Name* Describes a name for the device pool.
Cisco CallManager Group* Selects a redundancy group for the device pool. This redundancy
group can contain a maximum of three redundant Cisco
CallManager servers.
Date/Time Group* Assigns the correct time zone to the device.
Region* Determines the coder-decoder (codec) selection used by the device,
depending on the end location of the call.
Softkey Template* Defines the type and order of the softkeys that are displayed on the
liquid crystal display (LCD) of a Cisco IP Phone.
SRST Reference* Configures SRST and selects the gateway that will support the
device if the connection to the Cisco CallManager is lost.
continues
Table 5-1 Cisco CallManager Device Pool Configuration Options (Continued)
Field Description
Calling Search Space for Defines whom an IP Phone is able to call if it auto-registers with the
Auto-registration Cisco CallManager.
Media Resource Group List Assigns media resource support to a device for functions such as
conferencing, transcoding, or MOH.
Network Hold MOH Audio Selects the audio that Cisco CallManager should play when a user
Source presses the Transfer or Conference button on the Cisco IP Phone.
User Hold MOH Audio Selects the audio that Cisco CallManager should play when a user
Source presses the Hold button on the Cisco IP Phone.
Network Locale Defines the tones and cadences that the device uses.
User Locale Defines the language that the device uses.
Connection Monitor Defines the amount of time that the IP Phone monitors its
Duration connection to Cisco CallManager before it unregisters from SRST
and reregisters to Cisco CallManager. This is to ensure that the link
is stable (not “flapping”). The default for the enterprise parameter
specifies 120 seconds, which can be modified on a device-pool basis
or left at the default value.
MLPP Precedence and Manages Multilevel Precedence and Preemption (MLPP) settings:
Preemption Information
MLPP Indication: Specifies whether devices in the device pool that
are capable of playing precedence tones will use the capability when
the devices plan an MLPP precedence call
MLPP Preemption: Specifies whether devices in the device pool

that are capable of preempting calls in progress will use the
capability when the devices plan an MLPP precedence call
MLPP Domain: A hexadecimal value for the MLPP domain that is

associated with the device pool
*Indicates a required field.
If you make changes to a device pool, you must reset the devices in that device pool before the
changes will take effect.
You cannot delete a device pool that has been assigned to any device or one that is used for device
defaults configuration. To find out which devices are using the device pool, click the Dependency
Records link in the Device Pool Configuration window. If you try to delete a device pool that is
in use, an error message is displayed. Before deleting a device pool that is currently in use, you
must perform one of the following tasks:
■ Update the devices to assign them to a different device pool.
■ Delete the devices that are assigned to the device pool that you want to delete.
Individual components of a device pool are explored in the following subtopics.
Cisco Unified CallManager Group Configuration

A Cisco CallManager group specifies a prioritized list of Cisco CallManager servers for a Cisco
IP Phone to register to, with a maximum of three in the list. The first Cisco CallManager in the list
serves as the primary Cisco CallManager for devices that are assigned to that group. The other
members of the group serve as the secondary and tertiary backups. Changes to the Cisco
CallManager group affect the configuration file that is given to Cisco IP Phones by the TFTP
server when they initially boot.
Complete these steps to configure a Cisco CallManager group:
Step 1 Choose System > Cisco CallManager Group. The default group that was
created by Cisco CallManager during the installation appears.
Step 2 Choose Add New Cisco CallManager Group to create a new Cisco
CallManager group.
Step 3 Move the “Available” Cisco CallManager servers to the “Selected” Cisco
CallManagers using the left and right arrows to place them in the group.
You can change the order of Cisco CallManager servers using the up and
down arrows, which adjusts the priority of the Cisco CallManager servers
where the top server will be the primary server of the group and the bottom
server will be the tertiary server of the group.
In Figure 5-3, you can see the Cisco CallManager group called “Arizona” has three Cisco
CallManager servers. You would assign the Cisco CallManager group to a device pool, and you
then assign this device pool to the Cisco IP Phone. The IP Phone uses the Arizona Cisco
CallManager as its primary Cisco CallManager, the California Cisco CallManager as its
secondary, and the Michigan Cisco CallManager as its tertiary. Cisco CallManager Administration
will present an error message if you attempt to add a fourth Cisco CallManager (for example, the
EAST1A Cisco CallManager) to the list.
NOTE The Auto-registration Cisco CallManager Group check box (shown in Figure 5-3)
dictates that all devices that auto-register will receive the current Cisco CallManager Group
assignment. You can only designate a single Cisco CallManager Group as the auto-registration
group. Auto-registration is covered thoroughly at the end of this chapter.
Figure 5-3 Cisco CallManager Group Configuration Window
NOTE The sample Cisco CallManager group configuration is an example of clustering over
the IP WAN because only Cisco CallManagers in the same cluster can be added to a Cisco
CallManager group. More commonly, Cisco CallManager clustered servers reside in the same
physical location. You can then configure Cisco CallManager groups based on the servers within
your data center. For example, the server Arizona_SUB1 could be the primary, Arizona_SUB2
could be the secondary, and Arizona_PUB could be the tertiary server in a Cisco CallManager
group.
Date/Time Group Configuration

Date/time groups define time zones for the various devices that are connected to Cisco
CallManager. You can assign each device to only one device pool. As a result, the device has only
one date/time group.
Cisco CallManager has a default date/time group called CMLocal. The CMLocal date/time group
synchronizes to the active date and time of the operating system on the Cisco CallManager server.
You can change the settings for CMLocal after installing Cisco CallManager.
NOTE All Subscriber Cisco CallManager servers in a cluster synchronize their time with the
Publisher server through a Microsoft process known as W32TIME. Cisco highly recommends
using the Microsoft NTP client on the Publisher server to synchronize its time with a more
accurate source.
Complete these steps to configure the date/time group:
Step 1 Choose System > Date/Time Group. The default CMLocal group appears.
Step 2 Choose Add a New Date/Time Group to insert additional date/time
groups as required.
Figure 5-4 illustrates the addition of a California date/time group.
Figure 5-4 Cisco CallManager Date/Time Group Configuration Window
TIP For a worldwide distribution of Cisco IP Phones, you will want to create one named date/
time group for each of the time zones where you have deployed Cisco IP Phones.
Region Configuration
The region is typically the most confusing item of the device pool configuration. When you create
a region, you specify the audio codec that can be used for calls between devices (such as IP
Phones) within that region and between that region and other regions. As of Cisco CallManager
Release 4.0, you can also specify the video call bandwidth.
You can create regions to modify the codec selection for any reason; however, most network
administrators create regions based on geographical areas. The default voice codec for all calls
through Cisco CallManager is G.711. If you do not plan to use any other voice codec, you do not
need to use regions. The system will use the default region.
For example, in Figure 5-5, there are four regions: Arizona, California, Default, and Michigan; the
configuration of the Michigan region is displayed. If a device that is assigned to the Michigan
region calls another device in the Michigan region, the devices use the G.711 codec. However, if
a device assigned to the Michigan region calls a device that is assigned to the Arizona region, the
devices use the G.729 codec. Cisco CallManager creates the Default region during the installation
process. You can rename the Default region to a more logical name to avoid confusion, or you can
ignore it and use only regions that you have created.
Figure 5-5 Cisco CallManager Region Configuration

Complete these steps to configure a region:
Step 1 Choose System > Region. The default region that was created during the
Cisco CallManager installation appears.
Step 2 Choose Add a New Region to configure the regions, and choose the codec
and video bandwidth as appropriate between the regions. Click Insert to
save your changes.
Softkey Template Configuration

The Softkey Template Configuration window allows the administrator to manage the on-screen
softkeys that the Cisco IP Phones support (such as the Cisco IP Phone 7960 and 7940 models).
Softkeys govern functionality such as Hold, Conference, Transfer, and so on. You can configure
these softkeys with many Cisco CallManager functions and features. You can also use the softkey
templates to restrict what features the user can access. For example, if you do not want the user to
have the ability to initiate conference calls, you can just remove the conference button from their
softkey template.
To access the Softkey Template Configuration window, open the Cisco CallManager
Administration window and choose Device > Device Settings > Softkey Templates. Cisco
CallManager comes with a few default softkey templates shown in Figure 5-6, which are discussed
fully in future chapters. You cannot modify or delete these default templates; however, you can
make copies of them for your own use by clicking the Copy icon located to the right of the
template name.
Figure 5-6 Cisco CallManager Softkey Template Configuration
After you have made a copy of the softkey template you want to use, you can modify the softkeys
available, and assign the template to a device pool. The full configuration of the softkey templates
are covered in future chapters.
IP Phone Button Templates

Whereas softkey templates focus on modifying the features assigned to Cisco IP Phones, the IP
Phone button template focuses on assigning line and speed dial buttons to the phone. Although
Cisco CallManager does not require the IP Phone button templates for the device pool, it is good
to have all the button templates you plan on using created before you begin to add IP Phones to
the Cisco CallManager SQL database. Figure 5-7 shows the physical buttons affected on the Cisco
7961 IP Phone.
Figure 5-7 Cisco 7961 IP Phone Softkey and Line Buttons

Softkey Template Buttons
Phone Button Template Buttons
You must assign at least one line per IP Phone; usually this line is button 1. Depending on the
Cisco IP Phone model, you can assign additional lines.
Before adding any IP Phones to the system, create phone button templates with all of the possible
combinations for all IP Phone models. An IP Phone model can have various combinations; for
example, a Cisco IP Phone 7960 supports six lines and can use the following phone button
template combinations:
■ One line, five speed dial buttons
■ Two lines, four speed dial buttons (default)
■ Three lines, three speed dial buttons
■ Four lines, two speed dial buttons
■ Five lines, one speed dial button
■ Six lines, no speed dial button

To create a phone button template from Cisco CallManager Administration, click Device > Device
Settings > Phone Button Template. Click the Find button to see the default templates included
with Cisco CallManager (shown in Figure 5-8).
TIP There are additional methods to add speed dial capabilities to a Cisco IP Phone; you are
not limited to just the phone button template buttons. These mechanisms are discussed in future
chapters.
Figure 5-8 Cisco CallManager Phone Button Template Configuration
As with the softkey templates, you are unable to modify the default phone button templates.
Rather, you can make copies of the default templates (by clicking on the Copy icon) and saving
them under a unique name.
Create easily recognizable naming conventions for the phone button template. A suggested best
practice is to use the model number of the Cisco IP Phone followed by the number of lines and
speed dials. For example, a phone button template named “7960 1-5” would indicate a Cisco 7960
IP Phone with one line and five speed dial buttons.
IP Phone Configuration 87
Putting It All Together

The previous sections discussed creating the mandatory, minimum elements to assemble a device
pool. After you have created all the individual elements you need, create device pools that group
the common configuration elements together. For example, you might have the following device
pool configuration:
■ Device Pool Name: Arizona Phones
■ Cisco CallManager Group: Arizona_CG (CCM_AZ Primary, CCM_CA Secondary)
■ Date/Time Group: Arizona_DT (selects Arizona as the time zone)
■ Region Configuration: Arizona_REG (selects G.711 for calls within Arizona, G.729 for
calls elsewhere)
■ Softkey Template: Standard User
■ SRST Reference: Disable (the default, disables SRST)
You would then create additional device pools based on location, Cisco CallManager Groups, or
other unique configurations. The final step is to assign the device pools to the Cisco IP Phones that
will use them. As of yet, you have yet to add these IP Phones to the Cisco CallManager database.
IP Phone Configuration
The bulk of your initial Cisco CallManager configuration comes in adding Cisco IP Phones to the
SQL database. After you add the phones to the database, you can manage and configure them
through the Cisco CallManager interface. You can add the phones to the database using one of
three methods:
■ Manual entry
■ Automatic registration
■ The Bulk Administration Tool (BAT)
This chapter discusses the manual and automatic methods of adding IP Phones to the database.
BAT is covered in Chapter 7, “Cisco Bulk Administration Tool.”
Manual IP Phone and Directory Number Configuration

Manually adding new IP Phones to the network is often tedious, but it can constitute a large part
of day-to-day voice network management. The Bulk Administration Tool (BAT) allows you to add
a large number of IP Phones to the Cisco CallManager database at once, but BAT is not appropriate
for adding or modifying a single IP Phone for a new employee.
Cisco CallManager uses the IP Phone MAC address to track the phone in the voice network. Cisco
CallManager ties all IP Phone configuration settings to the IP Phone MAC address. Before you
can perform any configuration on a Cisco IP Phone through Cisco CallManager, you must find the
MAC address of that IP Phone. Use the following guidelines to locate a MAC address:
■ You can find the MAC address in the text and Universal Product Code (UPC) form, which is
imprinted on the shipping box for the IP Phone. Some administrators use bar code scanners
to simplify the process of adding multiple IP Phones.
■ You can also find the MAC address in the text and UPC form on the back of the IP Phone, on
the middle sticker near the bottom.
■ If you boot the IP Phone, you can press the Settings button on the face of the phone. Use the
arrow keys to navigate, and choose Network Configuration. The MAC address will be
displayed on line 3 of the network configuration.
You can continue the Cisco IP Phone configuration on the Cisco CallManager configuration after
you have the MAC address of the IP Phone, as follows:
Step 1 In Cisco CallManager Administration, choose Device > Phone to open the
Find and List Phones window.
Step 2 Choose Add a New Phone in the upper-right corner of the window.
Step 3 Choose the model of the IP Phone from the drop-down menu, and click
Next.
Step 4 At a minimum, you must configure the MAC Address, Device Pool, and
Phone Button Template fields; then click Insert.
Step 5 Cisco CallManager prompts you to add a DN for line 1; then click OK.
Step 6 When the Directory Number Configuration window appears, enter the DN
of the IP Phone in the appropriate field, and click Insert.
Figures 5-9 and 5-10 illustrate the IP Phone and directory number configuration screens,
respectively.
Figure 5-9 Cisco CallManager IP Phone Configuration

Figure 5-10 Cisco CallManager Directory Number Configuration
Automatic IP Phone Registration

Auto-registration allows Cisco CallManager to issue extension numbers to new IP Phones, which
is similar to the way in which the Dynamic Host Control Protocol (DHCP) server issues IP
addresses. When a new IP Phone boots and attempts to register with Cisco CallManager for the
first time, Cisco CallManager issues an extension number from a configured range. After Cisco
CallManager issues the extension, it records the extension number to the MAC address mapping
in the SQL database.
Although auto-registration simplifies the process of deploying a new IP telephony network, it is

an option that is available only in some new IP telephony deployments. Because administrators
deploy most IP telephony networks as a migration from a PBX environment, users have existing
telephone extensions. These existing telephone extensions typically map to Direct Inward Dial
(DID) numbers from the Public Switched Telephone Network (PSTN) and cannot change.
Therefore, these IP telephony deployments usually use manual configuration rather than auto-
registration.
You should carefully evaluate auto-registration before implementing it because its use can pose a
security risk to the network. Auto-registration allows anyone with physical access to the voice
network to connect an IP Phone and use it, regardless of whether they are authorized. For this
reason, many organizations, as part of their security policy, disable the use of auto-registration or
use auto-registration in a secure staging environment for initial Cisco CallManager configuration.
Complete these steps to configure the Cisco CallManager server to support auto-registration:
Step 1 From Cisco CallManager Administration, choose System > Cisco CallManager.
Step 2 From the list of Cisco CallManager servers, select the server that you want to
support auto-registration.
Step 3 Under the Auto-Registration Information Configuration section, enter the
appropriate DN range in the Starting and Ending Directory Number fields.
Step 4 Ensure that the Auto-registration Disabled on this Cisco CallManager check
box is unchecked.
Step 5 Click Update to save your changes.
Figure 5-11 shows the auto-registration configuration window.
Figure 5-11 Cisco CallManager Auto-Registration Configuration

Installing Cisco CallManager automatically sets device defaults. You cannot create new device
defaults or delete existing ones, but you can change the default settings. Use device defaults to set
the default characteristics of each type of device that registers with a Cisco CallManager. The
device defaults for a device type apply to all auto-registered devices of that type within a Cisco
CallManager cluster. You can set the following device defaults for each device type to which they
apply:
■ Device load—Lists the firmware load that is used with a particular type of hardware device
■ Device pool—Allows you to select the device pool that is associated with each type of device
■ Phone button template—Indicates the phone button template that is used by each type of
device
When a device auto-registers with a Cisco CallManager, it acquires the device default settings for
its device type. After a device registers, you can update its configuration individually to change the
device settings.
Complete these steps to update the device defaults:
Step 1 In Cisco CallManager Administration, choose System > Device Defaults

to open the Device Defaults Configuration window.
Step 2 In the Device Defaults Configuration window, modify the appropriate
settings for the device that you want to change.
Step 3 Click Update to save the changes in the Cisco CallManager configuration
database.
Step 4 Click the Reset icon to the left of the device name to reset all the devices
of that type and load the new defaults on all Cisco CallManager servers in
the cluster. If you choose not to reset all devices of that type, only new
devices that are added after you change the device defaults receive the latest
defaults.
CAUTION Updating device load files and performing a phone reset can affect a large number
of devices and cause significant WAN utilization as firmware files are sent to remote phones.
Cisco recommends saving these large device resets for network off-hours.
Case Study: Device Pool Design

Scenario: You are designing the IP telephony network for Widgets, Inc. The company has two
locations: a central site in Arizona with 1200 users and a remote site in Missouri with 20 users.
These sites are connected through a fractional T1 CAS line supporting a speed of 768 Kbps. The
Case Study: Device Pool Design 93
company has deployed a centralized Cisco CallManager cluster of three servers: AZ_CCM1 (SQL
Publisher), AZ_CCM2 (SQL Subscriber), and AZ_CCM3 (SQL Subscriber). You must configure
the central Cisco CallManager cluster to support phones at both locations.
There is no better way to get a firm grasp on the concept of device pools than to put it in action.
This scenario shows the perfect example of a centralized Cisco CallManager deployment where
the remote site does not have enough phones to justify a multicluster configuration, nor does it
have enough bandwidth to support running a cluster split over the IP WAN. In this scenario, the
customer network will require two device pools to appropriately configure the devices. This is
because the IP Phones reside in two different time zones and will need to use different codecs
when communicating (G.711 uncompressed for the LAN and G.729 compressed for the WAN).
Table 5-2 illustrates the configuration of the device pool elements.
Table 5-2 Device Pool Elements
Device Pool Description

Element
Cisco CallManager Unless the network requires load balancing, the network will only require
Group one Cisco CallManager group:
• Name—Widget_CG
— Primary CCM server—AZ_CCM3
— Secondary CCM server—AZ_CCM2
— Tertiary CCM server—AZ_CCM1
Notice that the SQL Publisher server (AZ_CCM1) is listed last. This is
because you want to keep as much device registration and support load
off the Publisher as possible.
Date/Time Group This network requires two date/time groups to support the phones in
differing locations:
• Name—Widget_AZ_DT
— Time zone—(GMT-07:00) Arizona
— Date format—M-D-Y
— Time format—12-hour
• Name—Widget_MO_DT
— Time zone—(GMT-06:00) Central time
— Date format—M-D-Y
— Time format—12-hour
continues
Table 5-2 Device Pool Elements (Continued)
Device Pool Description

Element
Region Because the WAN has limited bandwidth available, it is necessary to
configure two regions to select different codecs:
• Name—Missouri_RG
— Audio Codec (within Missouri_RG)—G.711
— Audio Codec (to Arizona_RG)—G.729
• Name—Arizona_RG
— Audio Codec (within Arizona_RG)—G.711
— Audio Codec (to Missouri_RG)—G.729
Phones assigned to the Missouri_RG will now use a compressed codec
when communicating with phones assigned to the Arizona_RG, and vice
versa.
Softkey Template Because Widgets, Inc., specified no special requirements for the users,
they should be able to use the Standard User softkey template, which
provides a basic set of functions to the IP Phones.
SRST Reference Although the SRST reference is discussed in later chapters, this scenario
requires you to create a SRST reference for the phones in Missouri. The
SRST reference would allow the router in Missouri to support the IP
Phones and provide basic PSTN calling if the connection to the
centralized Cisco CallManager cluster were lost. In this case, we will
assume you created a SRST reference called Missouri_SRST that
pointed to the Missouri gateway IP address 10.5.1.1.
Now that the device pool elements are created, you can create the two device pools:
Device Pool 1
Name—Arizona_DP
Cisco CallManager Group—Widget_CG
Date/Time Group—Widget_AZ_DT
Region—Arizona_RG
Softkey Template—Standard User
SRST Reference—Disabled
Summary 95
Device Pool 2
Name—Missouri_DP
Cisco CallManager Group—Widget_CG
Date/Time Group—Widget_MO_DT
Region—Missouri_RG
Softkey Template—Standard User
SRST Reference—Missouri_SRST
All that is left is to assign the Cisco IP Phones in each location to the correct device pool.
Summary
This chapter covered the configuration of Cisco CallManager to support Cisco IP Phones. Starting
off, the first configuration administrators typically attack is to change the Cisco CallManager
server configuration to distribute server IP addresses to the IP Phones rather than to hostnames.
This eliminates the IP Phone reliance on DNS. From there, you must configure device pools,
allowing you to assign the correct configuration to the Cisco IP Phones. The required elements of
the device pool were covered in this chapter. They are as follows:
■ Cisco CallManager Group
■ Date/Time Group
■ Region
■ Softkey Template
■ SRST Reference (covered in later chapters)
After you have created these elements, you can group them together in one or more device pools
and assign them to the devices.
The Cisco IP Phones can be added to the Cisco CallManager SQL database manually,
automatically, or by using the Bulk Administration Tool. This chapter discussed the manual and
auto-registration addition of IP Phones. Auto-registration allows the Cisco CallManager to assign
phone extensions like DHCP addresses; however, this configuration is only good for a “green
field” deployment where there are no preexisting phone extension assignments. Most of your day-
to-day administration will consist of adding IP Phones manually to the Cisco CallManager
configuration.
Review Questions
1. Which five settings are required to configure a device pool? (Choose five.)
a. Media Resource Group List
b. Cisco CallManager group
c. date/time group
d. region
e. softkey template
f. SRST reference
g. phone template
h. phone load
i. device defaults
2. Which of the following combinations is a valid phone button template configuration for a
Cisco IP Phone 7960?
a. 0 lines, 6 speed dials
b. 2 lines, 4 speed dials
c. 6 lines, 1 speed dial
d. 4 lines, 4 speed dials
3. Which navigation path would you use to configure Cisco CallManager to automatically
register an IP Phone?
a. System > Server
b. System > Cisco CallManager
c. Service > Service Parameters
d. Device > Phone
e. System > Device Pool
4. How does Cisco CallManager tie configuration information to the IP Phones in the Microsoft
SQL database?
a. IP address
b. unique GUID
c. hostname
d. MAC address
e. device pool ID
Review Questions 97
5. Which of the following device pool characteristics allows you to configure the codec the
Cisco IP Phone will use when communicating with other devices?
a. Region
b. Softkey Template
c. Compression
d. Compression Group
6. By default, the IP Phones will receive a list of Cisco CallManager hostnames they can use to
register from the TFTP server. How can you modify this so the IP Phone receives a list of
Cisco CallManager IP addresses from the TFTP server?
a. You must manually edit the IP Phone configuration files on the TFTP server, replacing
hostnames with IP addresses.
b. You must access the Cisco CallManager server configuration from System > Server
and change the hostnames to IP addresses. The Cisco CallManager will automatically
update the TFTP server.
c. The IP Phones must use hostname resolution to communicate with the Cisco CallMan-
ager. Ensure DNS redundancy is in place.
d. In the Cisco CallManager, access the Service > Service Parameters window and
change the preferred resolution method to IP addresses rather than hostnames.
7. Which of the following device pool elements allows you to configure a redundant group of
Cisco CallManager servers your IP Phone can use for registration?
a. Cisco CallManager Servers
b. CallManager List
c. CallManager Gaggle
d. CallManager Group
8. The Cisco CallManager has one date/time group by default. What is this date/time group?
a. RegionalDT
b. Pacific Time
c. CMLocal
d. CCMServerClock
9. You want to remove the Hold button from a user’s IP Phone. Which device pool configuration
item would you use to accomplish this?
a. IP Phone Button Template
b. Softkey Template
c. Feature Restrictions
d. Feature Template
10. What three locations can you use to find the MAC address of a Cisco IP Phone?
a. You can find the MAC address in the UPC form, which is imprinted on the shipping box
for the IP Phone.
b. You can find the MAC address in the text and UPC form on the back of the IP Phone, on
a sticker near the bottom.
c. You can find the MAC address by pressing the Settings button on the face of the phone,
using the arrow keys to navigate, and choosing Network Configuration.
d. You can find the MAC address by telnetting to the IP address of the Cisco IP Phone and
issuing the show interface command.
following topics:
■ Using Cisco CallManager Administration to

add users and associate users with a device
■ Logging in to the Cisco CallManager User

Options web page and selecting a device to
personalize
■ Activating or subscribing typical user options

from the User Options web pages, such as
call forwarding, speed dial configuration, and
CallManager services
■ Creating personal address books in the Cisco

CallManager User Options web page of
stored names and numbers and assigning Fast
Dial codes to personal address book entries to
enable users to dial those codes in place of
telephone numbers
■ Describing how to change the way that the

voice-message light on the handset works
when a user receives a voice-mail message
■ Describing how to change the language for

the Cisco CallManager User Options web
pages or telephone
CHAPTER 6
Cisco IP Telephony Users
After you have configured the Cisco CallManager to support the various Cisco IP Phones
deployed around your network environment, your next logical step is to configure the IP
telephony users. Configuring users for the voice network is not a mandatory step; however, it
will ease your network administration load considerably because (with proper training) voice
network users can configure their own IP Phones for common features such as speed dials and
call forwarding without adding another support call to the help desk group. In addition, Cisco
CallManager requires user configuration to support many of the advanced features, such as
Cisco Auto Attendant, IP Manager Assistant (IPMA), and Extension Mobility, which are
discussed in Chapters 17-19.
This chapter describes the Cisco CallManager configuration to support voice network users. You
will learn how to add these users to the Cisco CallManager user database and assign these users
access to their respective IP Phone. From there, the chapter discusses the process an end-user
would go through to configure their phone through the Cisco CallManager user web pages.
Cisco CallManager User Database

Cisco CallManager stores most configuration information for the voice network in the
underlying SQL 2000 database infrastructure. However, user information does not fall under
this category; rather, the user information is stored in a separate database accessible through the
Lightweight Directory Access Protocol (LDAP). LDAP is an industry standard protocol that
allows applications to access and potentially modify information stored in the database. Cisco
decided to keep the user database separate from the rest of the CallManager configuration
because network administrators might want to harness their existing corporate directory (that is
already filled with user information used to access the data network) for the voice network.
By default, Cisco CallManager uses an embedded LDAP directory that installs with the
CallManager software image. This directory is called DC Directory. However, you can also
choose to use your existing corporate directory as the CallManager user database. CallManager
supports the following directory integrations as of version 3.3(3) and 4.0(1)sr2:
■ Microsoft Active Directory 2000
■ Microsoft Active Directory 2003

102 Chapter 6: Cisco IP Telephony Users
■ Netscape Directory Server 4.x
■ iPlanet/Sun Directory Server 4.1
Although integrating Cisco CallManager with the corporate directory provides tremendous
advantages, there are equally tremendous caveats to this configuration:
■ To integrate with a corporate directory, the network administrator must modify the schema of
that directory. This is a networkwide change that affects all servers participating in the
replication of the corporate directory and might cross political boundaries in the organization.
■ The CallManager servers must be able to access the corporate directory servers at all times.
Failure of this communication will result in authentication failures (and equipment
inoperability) in the voice network.
■ The additional read/write operations on the corporate directory will result in additional load
for the servers maintaining this directory. An overlay of the voice network to your network
infrastructure can potentially double the number of requests to the corporate directory servers.
NOTE Because the process of integrating the Cisco CallManager into an LDAP corporate
directory differs greatly depending on the type of directory service you are using and its
configuration, we will assume you are using the embedded DC Directory service for the
remainder of this text. The Cisco website and CallManager SRND documentation provides
detailed instructions if you are considering this integration.
The following CCO link also provides excellent information about Active Directory integration:
http://www.cisco.com/en/US/products/sw/voicesw/ps556/
products_implementation_design_guide_chapter09186a0080447505.html#wp1043120
Cisco CallManager User Configuration

Regardless of the directory you use to store user information, the CallManager configuration
remains the same. You can perform all user administration for Cisco CallManager from the same
CallManager Administration web pages you have used thus far for the CallManager configuration.
Adding a User
Figure 6-1 shows an example of adding a user to the Cisco CallManager directory database by
using Cisco CallManager Administration. To add a user in Cisco CallManager Administration,
choose User > Add a New User from the CallManager CCMAdmin web pages.
Cisco CallManager User Configuration 103
Figure 6-1 Cisco CallManager User Configuration Window
Before adding a user, gather the following user information:
■ First name
■ Last name
■ User ID (username)
■ Telephone number or DN
■ Manager user ID (manager username)
■ Department
If the user is going to access the Cisco SoftPhone application, Cisco Auto Attendant, or any other
computer telephony integration (CTI) application, check the Enable CTI Application Use check
box.
When you are first setting up a user, assign a simple password (at least four characters) and a
personal identification number (PIN), which must be at least five digits, for the user to use on the
initial login. The user can then change the initial password and PIN from the Cisco CallManager
User Options page. The Cisco CallManager uses the user password and PIN for different functions
in the voice network. For example, the user can use their password to log in to the User Options
web pages while they can use their PIN to access a phone using Extension Mobility (which is
covered in Chapter 17, “Configuring User Features, Part 2”).
After the directory information is added, you can associate the user with a device or devices.
Configuring Device Pools

After you have added a user, you can associate devices over which the user will have control.
When users have control of an IP Phone, they can control certain settings for that phone, such as
speed dialing and call forwarding. You can associate a user with many devices; however, only one
of the devices can be the primary extension for that user. Figure 6-2 shows the Device Association
window.
Figure 6-2 Cisco CallManager Device Association Window

User Logon and Device Configuration 105
To assign devices to a user, access the User Configuration window for that user and then complete
the following steps to assign the devices:
Step 1 In the Application Profiles pane on the User Configuration page, click
Device Association.
Step 2 Limit the list of available devices by entering the search criteria in the
Available Device List Filters section, if desired, and click Select Devices.
Step 3 Check the check box of one or more devices that you want to associate with
the user. You can assign one primary extension from the devices to which
the user is assigned by clicking the radio button in the Primary Ext. column
for that device.
Step 4 When you have completed the assignment, click Update Selected to assign
the devices to the user.
User Logon and Device Configuration

After you have added users to the LDAP database and associated them with the appropriate
devices, those users will be able to log on to the CallManager User Options web page to configure
their devices. Of course, before the users begin accessing these web pages, some training will be
necessary. You might find it beneficial to create a step-by-step training manual for your
organization that you can distribute to users or post on an intranet website describing each of the
following sections.
TIP You can use the following sections in the book (which walk through the user configuration
options) as a model when creating your organization user training manual.
CallManager User Options: Log On

To open the Cisco CallManager User Options web page, enter this URL:
https://<server_name>/ccmuser
The server name is the hostname or IP address of the Cisco CallManager server.
TIP It would be very beneficial to use a network policy, such as the Active Directory Group
Policy options, to automatically add a desktop shortcut or Microsoft Internet Explorer favorite
linked to the CCMUser web pages for all PCs in the network.
At the Cisco CallManager User Options page (shown in Figure 6-3), enter the correct username
and password. If this is the first time that the user is logging on, the user should obtain the URL,
username, and password from the administrator.
Figure 6-3 Cisco CallManager User Options: Log On Window
From any page within Cisco CallManager User Options web pages, a user can change the
language of the page by choosing the language (locale) from the View Page In drop-down menu,
if additional locales have been configured.
CallManager User Options: Welcome

After logging on, CallManager presents a user with the User Options welcome screen, shown in
Figure 6-4. From there, the user can select an associated device to configure. The user can
customize multiple associated devices by choosing more than one device.
Figure 6-4 Cisco CallManager User Options: Welcome Screen
A web location allowing users to customize Cisco IP Phone settings, such as adding speed dials
and forwarding calls means that users can be more productive in their work environment. For
example, when a user is not going to be in the office to receive an important call, the user can
access the Cisco CallManager User Options web page from home and forward calls from the
Cisco IP Phone to a cellular telephone or any other DN.
CallManager User Options: Call Forwarding

Figure 6-5 shows the Forward Your Calls page. To display this page, click the Forward All Calls
to a Different Number link in the User Options main menu.
Figure 6-5 Cisco CallManager User Options: Call Forwarding Window
From here, this user can forward all incoming calls on line 1 of a device to either voice mail or
another number.
If the user is forwarding calls to another number, the calling search space of the device using Call
Forward All (CFA) will restrict which numbers will be valid. Also, if the number is forwarded off-
net, the user must enter the number as if dialing from that telephone device. For example, a user
who wants to work from home wants all calls to the office telephone to forward to the home
number. If the user dials 92145550122 to call home from the office, the user must enter
92145550122 as the forwarding number.
CallManager User Options: Speed Dials

Figure 6-6 shows the Add/Update Your Speed Dials web page. To display this page, click the Add/
Update Your Speed Dials link in the User Options main menu.
Figure 6-6 Cisco CallManager User Options: Speed Dials Window
Depending on the device, the number of speed dials is limited based on settings in the phone
button template. The user can enter a number and label for each available speed dial button.
Buttons are available if the user is not using them for lines or services.
As you can see from Figure 6-6, the speed dial configuration window is divided into two sections:
Speed Dial Settings on Phone and Speed Dial Settings not associated with a phone button. The
settings on a phone represent the speed dials that correspond to a physical button on the IP Phone.
For example, if the user is using a Cisco IP Phone 7960 with the default template, they will have
two line buttons and four speed dial buttons. These four speed dial buttons represent the buttons
programmable from the “Speed Dial Settings on Phone” section. The settings not associated with
a phone button are accessed when the user first presses the AbbrDial softkey on their Cisco IP
Phone then dials the corresponding number. For example, to access the speed dial listed in field 5
shown in Figure 6-6, the user would press AbbrDial on the phone, then press the 5 button on the
numeric keypad. The user can also perform the reverse steps to activate the speed dial number (for
example, dial 5 followed by pressing the AbbrDial softkey).
When programming speed dials, enter the number precisely as it is dialed from the device. If the
number 9 must be dialed before the telephone number, the 9 must be part of the speed dial number
that is entered. For example, if the telephone number is 4805550199 and the access code for an
outside line is 9, the speed dial number must be entered as 94805550199.
TIP In older versions of CallManager, the Cisco IP Phone required a restart to update speed
dial configurations. Since CallManager version 3.3, a restart is no longer necessary.
CallManager User Options: IP Phone Services

Figure 6-7 shows the Subscribe/Unsubscribe IP Phone Services web page. To display this page,
click the Configure Your Cisco IP Phone Services link in the User Options main menu.
Figure 6-7 Cisco CallManager User Options: IP Phone Services Window
You can configure a number of IP Phone services for user subscription. For example, you could
configure services that store telephone numbers, meeting room availability, traffic reports, and
more. The user can then use the Subscribe/Unsubscribe IP Phone Services page in Cisco
CallManager User Options to subscribe to or unsubscribe from any of the Cisco IP Phone Services
that you have configured. IP Phone service configuration is covered in Chapter 16, “Configuring
User Features, Part 1.”
For example, if you have configured a service that looks up the stock price of a company, a user
can subscribe to that service from the Subscribe/Unsubscribe IP Phone Services page in
CallManager User Options. To view the stock price of a company, the user can press the Services
button on a Cisco IP Phone 7970, 7960, or 7940 and view the stock price on the IP Phone liquid
crystal display (LCD). You can also associate a service with one of the line/speed dial buttons to
allow quick access to XML applications, such as a corporate directory or speed dial list.
CallManager User Options: Personal Address Book and Fast Dial

The Personal Address Book service allows a user to store names and numbers for people internal
and external to the company. A user can also assign Fast Dial codes to Personal Address Book
entries and dial those codes in place of phone numbers. With the Fast Dial service, users can assign
one- or two-digit index numbers (from 1 to 99) for quick dialing from the Cisco IP Phone. Users
can assign index numbers either to Personal Address Book entries or to directory entries that they
add that do not correspond to address book entries.
The Personal Address Book and Fast Dial features are not available from the base Cisco IP Phone
feature set. Rather, Cisco has implemented them as XML services. You must configure these
services in the Cisco CallManager Service Configuration web interface before the CallManager
will make them available for user subscription. Chapter 16 covers XML Service configuration.
Users can configure the Personal Address Book and Fast Dial features from the Personal Address
Book page in Cisco CallManager User Options. After subscribing to these services, a user can
access the Personal Address Book and Fast Dial codes from the Cisco IP Phone by pressing the
Services button.
Figure 6-8 shows the Find/List Address Book Entries page. To display this page, choose the
Configure Your Cisco Personal Address Book link in the User Options main menu. From the
Find/List Address Book Entries page, users can add a new Personal Address Book entry, create or
modify Fast Dial codes to reach address book entries, or search for address book entries by using
either complete or partial strings.
Figure 6-8 Cisco CallManager User Options: Address Book and Fast Dial Window
CallManager User Options: Message Waiting Lamp Policy

Figure 6-9 shows the Change Your Message Waiting Lamp Policy page. To display this page, click
the Change the Message Waiting Lamp Policy for Your Phone link in the User Options main
menu. From here, a user can set the message waiting lamp policy for their device. The mainline
Cisco IP Phones have extremely noticeable message waiting lights (they have been called a
“Beacon in the Night” by some). Because of this, there are five settings that the user can configure:
Use System Policy, Light and Prompt, Prompt Only, Light Only, and None. The light configuration
illuminates the external lamp on the phone. This can be seen from some distance (such as in a
lighthouse from the ocean). The prompt configuration displays an envelope animation on the
display screen of the phone. The default system policy is set to light the lamp and prompt the user.
Figure 6-9 Cisco CallManager User Options: Message Waiting Lamp Policy Window
CallManager User Options: Locale Settings

Cisco CallManager allows users to select their preferred language settings for both the devices
they are using and the user web pages they access. By default, the Cisco CallManager comes with
only the English language selection. However, you can obtain additional languages from Cisco
based on the location(s) of your users. Supported locales include the following:
■ Chinese Simplified (China)

■ Chinese Traditional (Taiwan)
■ Danish (Denmark)
■ Dutch (Netherlands)
■ Finnish (Finland)
■ French (France)
■ German (Germany)
■ Greek (Greece)
■ Hungarian (Hungary)
■ Italian (Italy)
■ Japanese (Japan)
■ Korean (Korea)
■ Norwegian (Norway)
■ Polish (Poland)
■ Portuguese (Portugal)
■ Russian (Russia)
■ Spanish (Spain)
■ Swedish (Sweden)
If a user locale is not selected, the systemwide locale is used.
Figure 6-10 shows the Select a User Locale for Your Phone page. To display the language selection
pages, click the Change the Locale for This Phone or Change the Locale for These Web Pages
link in the User Options main menu. These options allow the user to modify the language selection
for the Cisco IP Phone or the web pages, respectively.
Figure 6-10 Cisco CallManager User Options: Phone Locale Window

Summary
This chapter covered the configuration of the user database for the Cisco IP telephony network.
Although a functional voice network does not require a user database, there are many applications
you will be missing. In addition, adding a user database allows your users to configure many of
the everyday options, such as call forwarding and speed dials, on their IP Phone through an easy-
to-use web interface.
To support a user database, you must first decide whether you will use the integrated DC-Directory
LDAP-compliant database that ships with CallManager or integrate the CallManager with one of
the supported directory services that are already running for your data network environment. After
making this decision, you can perform the user configuration and device association from the
CCMAdmin web interface. After you have associated the user with one or more IP Phones, the
user can access the CCMUser web interface (https://<CCM IP Address>/CCMUser) to manage the
devices.
Review Questions
1. Which of the following URLs can you use to access the Cisco CallManager User web pages
assuming the CallManager IP address is 10.1.1.1?
a. https://10.1.1.1/CCMAdmin
b. https://10.1.1.1/CCMUser
c. https://10.1.1.1/CCM
d. xml://10.1.1.1/CCMAdmin
e. xml://10.1.1.1/CCMUser
f. xml://10.1.1.1/CCM
2. The Cisco CallManager DC Directory is a/an _______-compliant database. (Choose the best
answer.)
a. SQL
b. Oracle
c. LDAP
d. Standards
3. For the Cisco CallManager–based IP telephony network to function correctly, you must add
users to an LDAP-compliant database. (True/False)
4. Which of the following fields are required to create a user who is able to use a Cisco
Softphone application? (Choose five.)
a. First Name
b. Last Name
c. User ID
d. Password
e. PIN Number
f. Manager User ID
g. Department
h. Enable CTI Application Use
i. SoftPhone Support Enabled
5. Before a user is able to configure a Cisco IP Phone through the CCMUser web pages, what
must be true? (Choose two.)
a. The user must have a valid user account.
b. The user must have Java support enabled on their workstation.
c. The user account must be associated with at least one Cisco IP Phone.
d. The user must use a Windows-based operating system.
6. You receive a technical support call from a troubled user of the IP telephony network. The
user has configured a personal address book through the CCMUser pages for their Cisco IP
Phone 7960. They are unable to figure out how to access this personal address book from their
Cisco IP Phone. What do you tell them?
a. You can access the personal address book by pressing the Directories button on the
Cisco IP Phone.
b. You can only view the personal address book through the CCMUser web pages.
c. You need to press the PAB softkey on your Cisco IP Phone.
d. You must first subscribe to the personal address book service and can then access it by
pressing the Services button on your IP Phone.
7. Which of the following is the default Message Waiting Lamp policy for the Cisco
CallManager system configured on each Cisco IP Phone? (Choose the best answer.)
a. Use System Policy
b. Light and Prompt
c. Light Only
d. Prompt Only
8. By default, Cisco CallManager supports three language selections for users: English,
Spanish, and French. (True/False)
9. A user can configure their IP Phone with up to _____ fast dial codes.
a. 20
b. 50
c. 99
d. 150
10. Which of the following options are available for a user who wants to forward their IP Phone
using the CCMUser web pages? (Select all that apply.)
a. Call Forward All
b. Call Forward Busy
c. Call Forward No Answer
d. Call Forward Selected
following topics:
■ Identifying the major features and

components of the BAT application
■ Installing the BAT application on a Cisco

CallManager publisher server
■ Using the BAT Wizard to perform bulk

configuration tasks
■ Creating an IP Phone template to use with

BAT
■ Identifying the two ways to create CSV files

for importing data into BAT
■ Using the BAT Wizard to validate the IP

Phone template and CSV file for errors prior
to inserting the devices into the Cisco
CallManager database
■ Using the BAT wizard to insert the IP Phones

into the Cisco CallManager database
■ Describing how to use BAT to update IP

Phone settings for a group of similar phones
■ Describing the major requirements for proper

installation of TAPS
CHAPTER 7
Cisco Bulk Administration Tool
Adding, updating, and deleting phones, users, and gateway ports are important functions in the
day-to-day activities of a Cisco CallManager administrator. When you use Cisco CallManager
Administration, each database transaction requires an individual manual operation. Manually
adding and configuring large numbers of these entities can be time consuming and tedious. The
Cisco Bulk Administration Tool (BAT) automates the process and achieves faster add, update,
and delete operations so that the system administrator can focus on more business- and network-
critical activities.
As a supplement to BAT, you can further ease the load of adding new phones to the network
through the Tool for Auto-Registered Phones Support (TAPS). You can use TAPS in conjunction
with BAT to update MAC addresses and download a predefined configuration for new phones
after a user has walked through an automated process. This chapter discusses the design and
implementation of both BAT and TAPS.
The Cisco Bulk Administration Tool

After you discover the Cisco BAT utility, it will quickly become your new best friend for most
day-to-day administration of the IP telephony network. Although there will always be individual
changes to the voice network, such as adding a phone extension or a new user, most of the
changes you will perform will need to occur in bulk. For example, the accounting department
needs different hold music or the department managers need specialized calling restrictions. You
can attack these types of tasks one device at a time (which can quickly become tedious and time
consuming) or with a few mouse button clicks using the Cisco BAT.
You can use BAT to work with the following types of devices and records:
■ Add, update, and delete Cisco IP Phones
■ Add, update, and delete users
■ Add, update, and delete user device profiles
■ Add, update, and delete Cisco IP Manager Assistant (IPMA) managers and assistants
■ Add, update, and delete ports on a Cisco Catalyst 6000 FXS Analog Interface Module
120 Chapter 7: Cisco Bulk Administration Tool
■ Add or delete Cisco VG200 analog gateways and ports
■ Add or delete forced authorization codes
■ Add or delete client matter codes
■ Add or delete Call Pickup groups
■ Update or delete locally significant certificates on Cisco IP Phones
BAT provides an optional application, TAPS, which retrieves the predefined configuration for
auto-registered telephones. By using TAPS, CallManager no longer assigns auto-registered
phones random extension numbers from the pool. Instead, after the user enters their extension
number through an automated process, the TAPS system retrieves the complete configuration you
have created for that user.
BAT Components
Every device includes a multitude of individual attributes, settings, and information fields that
enable the device to function in the network and provide its telephony features. Many devices have
the same attributes and settings in common, whereas other values, such as the directory number
(DN), are unique to a user or to a device.
For bulk configuration transactions involving the Cisco CallManager database, the BAT process
uses two components: a template for the device type that includes settings that devices have in
common and a data file in comma-separated values (CSV) format that contains the unique values
for configuring a new device or updating an existing record in the database. The CSV data file
works in conjunction with the device template.
For instance, when you create a bulk transaction to import a group of Cisco IP Phones, you set up
the CSV data file that contains the unique information for each phone, such as the DN and MAC
address. In addition, you set up or choose the BAT template that contains the common settings for
all phones in the transaction, such as a Cisco IP Phone 7960 template. By combining the unique
settings defined in the CSV file with the common settings created in the IP Phone template, you
have everything you need to import a group of IP Phones successfully.
CAUTION Because bulk transactions can affect Cisco CallManager performance and call
processing, use BAT only during off-peak hours.
BAT Installation
BAT must be installed on the same server as the publisher database for Cisco CallManager. During
BAT installation, the setup program stops the following services:
■ Microsoft Internet Information Server (IIS) administration
■ World Wide Web publishing
■ FTP publishing
These services automatically restart when the installation is complete.
When BAT is installed, the Microsoft Excel file BAT.xlt file for the BAT spreadsheet is placed on
the publisher database server at the following path: C:\CiscoWebs\BAT\ExcelTemplate\. You can
use this file to generate the necessary CSV files used to import groups of devices into the Cisco
CallManager SQL database.
NOTE You must install BAT directly on the publisher server (either through the console
directly or using a program such as VNC); do not use Windows Terminal Services.
These steps describe the BAT installation process:
Step 1 Using administrator privileges, log in to the system running the publisher
database for Cisco CallManager.
Step 2 Choose Application > Install Plugins from the CallManager
Administration web interface. The Install Plugins window is displayed, as
shown in Figure 7-1.
Figure 7-1 Cisco CallManager Plugins Installation Window
Step 3 Click the setup icon for the Cisco Bulk Administration Tool.
Step 4 A standard Windows dialog box appears. Determine whether to copy the
BAT installation executable to the system or run it from the current
location.
If an existing version of BAT is detected on the server, you are prompted to
confirm the reinstallation or upgrade. To reinstall BAT or to upgrade from
a previous version, click OK.
Step 5 The Welcome screen appears. Click Next, and the Current Settings window
appears.
Step 6 Click Next to install to the default location C:\CiscoWebs\BAT\. BAT
installs to C:\CiscoWebs\BAT\. You cannot change this path. The Start
Copying Files window appears. The setup program begins copying files.
Step 7 The Setup Complete window appears. You have successfully installed BAT.
Step 8 To close setup, click Finish.
Step 9 After you have installed BAT, from Cisco CallManager Administration,
choose Application > BAT to access BAT. If after you have installed BAT,
BAT is not visible under the Application menu, refresh your browser.
Adding Cisco IP Phones Using BAT

BAT uses a multistep process to prepare the bulk configuration transaction. BAT Release 5.0(1)
introduced a wizard to step you through bulk configurations. The BAT configuration process
includes these tasks:
Step 1 Set up the template for data input. You can create BAT templates for the
following types of device options:
• Phones—All Cisco IP Phone models and Cisco ATA 186, Cisco VGC
phones, CTI ports, and H.323 clients
• Gateways—Cisco VG200 and ports for the Cisco Catalyst 6000 FXS
Analog Interface Module
• User device profiles—Cisco IP Phone 7900 Series and Cisco SoftPhone
Step 2 Define a format for the CSV data file. You can use the BAT Excel
spreadsheet or a text editor to create the CSV data file.
Step 3 Validate the data input files with the Cisco CallManager database. Cisco
CallManager runs a validation routine that checks the CSV file and the
template for errors against the publisher database.
Step 4 Insert the devices into the Cisco CallManager database.
Using BAT Wizard Step 1: Choosing Devices

From the Configure menu shown in Figure 7-2, you can access the wizard by choosing one of these
devices or configuration options:
■ Phones
■ Users
■ Managers/Assistants
■ User Device Profiles
■ Gateways
■ Forced Authorization Codes
■ Client Matter Codes

■ Pickup Group
■ TAPS (optional, when installed)
Figure 7-2 Bulk Administration Tool Configure Menu
Using BAT Wizard Step 2: Device Specifics

After you choose a device or configuration option, the wizard displays a list of configuration tasks
that are specific to that option. For example, when you choose Phones, the following list of tasks
is displayed (shown in Figure 7-3):
■ Insert Phones—Add new phones
■ Update Phones—Locate and modify existing phones
■ Delete Phones—Locate and delete phones
■ Export Phones—Locate and export specific phone records or all phone records
■ Update Lines—Locate and modify lines on existing phones
■ Add Lines—Add new lines to existing phones

■ Reset/Restart Phones—Locate and reset or restart phones
■ Insert Phones with Users—Add new phones and users
■ Generate Phone Reports—Generate customized reports for phones
■ CAPF Configuration—Locate and modify or delete the digital certificates (called locally
significant certificates [LSCs]) issued by the Cisco Authority Proxy Function (CAPF) server
to IP Phones
Figure 7-3 Bulk Administration Tool Wizard Phones Options Menu
Using BAT Wizard Step 3: Proceed with Task

After you choose the configuration task, the wizard provides a list of steps that are specific to the
task. For example, to guide you through the Insert Phones task (shown in Figure 7-4), the wizard
displays the following steps:
Step 1 Add, view, or modify existing phone templates.

Step 2 Create the CSV data file.
Step 3 Validate phone records.
Step 4 Insert phones.
Figure 7-4 Bulk Administration Tool Wizard Insert Phones Window
When you choose a step from the task list, you open a configuration window such as the Phone
Template Configuration window. The configuration window provides the entry fields for defining
a template.
Adding Phones Step 1: Configuring BAT Templates

The first task in creating the data for the BAT configuration process is to set up a template for the
devices that you are configuring. You specify the type of phone or device that you want to add or
modify, and then you create a BAT template that has features that are common to all the phones
or devices in that bulk transaction.
Prior to creating the template, make sure that phone settings, such as device pool, location, calling
search space, button template, and softkey templates, have already been configured in Cisco
CallManager Administration. You cannot create new settings in BAT.
The first steps in configuring an IP Phone template are to give the phone template a unique name,
and choose an IP Phone (device) type in the Phone Template Configuration window, as shown in
Figure 7-5. Choose the template that encompasses all of the IP Phones in the group. If you have
multiple telephone types in a given group, you must create multiple templates.
Figure 7-5 Bulk Administration Tool Phone Template Configuration Window
Next, assign the template to a device pool. After you have configured the initial template
information, click Insert to add the template to the BAT utility.
After configuring the initial template settings, you can modify specific line configurations. From
the initial template window, choose a line to configure, and a new configuration window appears.
These general configuration settings can apply to multiple IP Phones, such as partitions, calling
search spaces, and call waiting settings. BAT obtains line configurations that are specific to the
user from the imported Microsoft Excel spreadsheet.
When you are adding a group of phones that have multiple lines, you can create a master phone
template that provides multiple lines and the most common values for a specific phone model. You
can use the master template to add phones that have differing numbers of lines, but do not exceed
the number of lines in the master phone template. For example, you can create a master phone
template for a Cisco IP Phone 7960 that has six lines. You can use this single template to add
phones that have one line, two lines, or up to six lines (because the template provides the base
configuration for up to six lines).
Adding Phones Step 2: Creating CSV Files

The CSV data file contains the unique settings and information for each individual device, such as
its DN, MAC address, and description. Make sure that all phones and devices in a CSV data file
are the same phone or device model and match the BAT template. The CSV data file can contain
duplicates of some values from the BAT template. Values in the CSV data file override any values
that were set in the BAT template. You can use the override feature for special configuration cases.
You can create CSV files in one of two ways: by using the Microsoft Excel spreadsheet BAT.xlt
file or by using a text editor such as Microsoft Notepad. The BAT Excel spreadsheet simplifies the
creation of CSV data files. You can add multiple devices and view the records for each device in
a spreadsheet format. It allows you to customize the file format within the spreadsheet and
provides validation and error checking automatically to help reduce configuration errors.
Experienced BAT users who are comfortable with working in a CSV-formatted file can use a text
editor to create a CSV data file.
Figure 7-6 shows the BAT.xlt Microsoft Excel spreadsheet. You can find this spreadsheet in the
directory C:\CiscoWebs\BAT\ExcelTemplate\ on the publisher database server. You probably do
not have Microsoft Excel running on the publisher, so you must copy the file to a local machine
(using either a floppy disk or a mapped network drive). After you have copied the file, double-click
BAT.xlt. When prompted, click Enable Macros.
TIP After you have installed BAT, a BAT shared folder is created automatically on the
publisher. This can provide easier access to the BAT files; to access the share, click Start > Run
and type \\<CCM Server Name or IP Address>\BAT.
The BAT spreadsheet includes tabs along the bottom of the spreadsheet for access to the required
data input fields for the various devices and user combinations in BAT. The CSV data file works
in combination with the BAT template. For example, when you choose the Phone tab in the BAT
spreadsheet, you can leave Location, Forward Busy Destination, or Call Pickup Group blank. The
values from the BAT phone template are used for these fields; however, if you specify values for
Forward Busy Destination or Call Pickup Group, those values override the values for these fields
that were set in the BAT phone template.
Figure 7-6 Bulk Administration Tool Excel Spreadsheet
After entering the data into the BAT spreadsheet, click Export to BAT Format to create the CSV
file. The format for CSV files is <tabname><timestamp>.txt. The system saves the file to
C:\XLSDataFiles\ or to a folder of your choice. You must move the converted CSV file from the
C:\XLSDataFiles\ folder on your local computer back to the publisher, where BAT can access the
CSV file and place it in the appropriate folder under C:\BATFiles. (For example, you could save
a phone CSV data file to the C:\BATFiles\Phones\Insert\ folder on the publisher database for Cisco
CallManager.)
Using a text editor is the other way to create a CSV file. When using a text editor, follow these
steps:
Step 1 Create the customized file format using the BAT File Format Configuration
window shown in Figure 7-7. The file format specifies the order in which
you enter values in the text file. This allows you to add attributes to the file
format that are also in the BAT template, and override the template entry
with a specific attribute for a device. For instance, you can choose the route
partition attribute for your file format and enter different partitions for each
phone in the CSV data file.
Figure 7-7 Bulk Administration Tool CSV File Format
Earlier versions of BAT supported only a default file format with a fixed and limited number of
attributes and settings for each device and an All Details format that includes all attributes and
settings for each device.
Step 2 Create the CSV data file using a basic text editor that follows the file format
you defined in Step 1.
Step 3 Associate the file format with the CSV data file in Cisco BAT. You can
associate only one file format with a CSV data file. Use the Add File Format
window, shown in Figure 7-8, to choose the name of the CSV data file
<CSVfilename>.txt from the File Name drop-down menu. Next, you
choose your file format from the File Format Name drop-down menu. The
data in the CSV data file must match the custom file format that you have
chosen.
Figure 7-8 Bulk Administration Tool CSV Format Association
Adding Phones Step 3: Validating Phones

In the next task in the BAT wizard, you use the Validate option, shown in Figure 7-9, to validate
the format of the text file you have chosen. In this task, you choose the name of the CSV data file
and the BAT template for the device or the model when you have a CSV data file with all details.
You have these options for how records are validated:
■ Specific Details—For validating records that follow the default or custom file format
■ All Details—For validating records from a file that was generated with the export utility by
using the All Details option
Figure 7-9 Bulk Administration Tool Validate Phones Window
When you choose Validate, the system runs a validation routine to check for errors against the
publisher database. These checks ensure the following:
■ Fields, such as description, display text, and speed dial label, use valid characters.
■ Cisco CallManager groups, device pools, partitions, and other referenced attributes are
already configured.
■ The number of lines that are configured on a device does not exceed the device template.
Validation does not check for the existence of a user or for mandatory or optional fields that are
BAT-defined, such as the dummy MAC address.
Adding Phones Step 4: Inserting Phones

Inserting the device into the Cisco CallManager database is the last step in using BAT to perform
bulk configurations, shown in Figure 7-10.
Figure 7-10 Bulk Administration Tool Insert Phones Window
The following steps are involved in this procedure:
Step 1 In the File Name field, choose the CSV data file that you created for this
specific bulk transaction.
Step 2 To enable the use of applications such as Cisco IP SoftPhone, check the
Enable CTI Application Use check box (for CTI ports only).
NOTE CTI application use is not required for the Cisco IP Communicator.
Step 3 Choose the Insert option that corresponds to your CSV data file.
Step 4 In the Phone Template Name field, choose the BAT phone template that
you created for this type of bulk transaction.
Step 5 If you did not enter individual MAC addresses in the CSV data file, check
the Create Dummy MAC Address check box.
This field automatically generates dummy MAC addresses in the following format:
XXXXXXXXXXXX, where X represents any 16-character, hexadecimal (0 to 9 and A
to F) number. You can update the phones or devices later with the correct MAC
address by manually entering this information into Cisco CallManager
Administration or by using TAPS.
Step 6 Click Insert to insert the phone records.
Updating IP Phones Using BAT

In addition to using BAT for bulk device addition to the CallManager database, you can also use
BAT to make bulk device updates to the system. This can be extremely useful when modifying a
configuration for a large number of users. For example, a group of users might want to change their
music on hold (MoH) to a new song. Rather than modifying the configuration of each phone
individually, you can use BAT to make a bulk change to all phones based on unique match criteria
(such as the department name or partition).
Figure 7-11 shows the Update Phones window. To update phone settings, such as changing or
adding a device pool or calling search space for a group of similar phones, choose Phones >
Update Phones in the BAT Configure window.
Figure 7-11 Bulk Administration Tool Update Phones Window
You can locate the existing phone records by using a query or a custom file containing the device
names or DNs. You can query using any number of fields, such as the model, device name, DN, or
description. You can also specify search criteria such as “begins with,” “contains,” or “is exactly.”
Choose View Query Result to check that the query returns the information that you need. Choose
Clear Query to remove the query items.
After you have defined the query or custom file to search for phones, follow this procedure to
update the IP Phones or users to the Cisco CallManager database in bulk:
Step 1 Specify the values to be updated within Cisco CallManager, as shown in

Figure 7-12.
Figure 7-12 Bulk Administration Tool Update Phones Detail Window
Step 2 Click Update.

Step 3 Reset or restart the IP Phones through Cisco CallManager.
To check the status of your insertion, read the status line, located above the Insert button.
If the status bar indicates a failure, click View Latest Log File to display a window that will help
you to determine where the operation failed.
Using the Tool for Auto-Registered Phone Support

Use TAPS in conjunction with BAT to provide two features:
■ Update MAC addresses and download predefined configuration for new phones
■ Reload configuration for replacement phones
When new phones are added to Cisco CallManager, TAPS works in conjunction with BAT to
update phones that were added to BAT using dummy MAC addresses. After BAT has been used
to bulk-add the telephones with dummy MAC addresses to Cisco CallManager Administration,
you can plug the telephones into the network. You or the user of the telephone can dial a TAPS
directory number that causes the phone to download its configuration. At the same time, the
telephone is updated in Cisco CallManager Administration with the correct MAC address. You
must make sure that auto-registration is enabled in Cisco CallManager Administration (System >
Cisco CallManager) for TAPS to function.
Prior to BAT Release 5.0(1), TAPS installation and uninstallation for Cisco CallManager was part
of the BAT installation program. With BAT Release 5.0(1) and later, TAPS is installed separately.
In addition, for TAPS to operate, you must also install the Cisco CallManager Extended Services
CD (which ships with current CallManager 4.x versions or can be downloaded from Cisco CCO)
or have purchased the Cisco Customer Response Application (CRA) server, which is typically
used in call center environments.
During TAPS installation or reinstallation on the publisher database server, the setup program
halts the following services:
■ Microsoft IIS administration
■ World Wide Web publishing
■ FTP publishing
These services restart when the installation is finished.
You cannot use Windows Terminal Services to install TAPS. You must install TAPS directly from
the Cisco CallManager publisher server and the CRA server (if you are using a CRA server).
NOTE In earlier versions of CallManager (prior to 4.x), CRA servers were required to be
separate from the CallManager server. In the 4.x versions, Cisco CRA services can reside on the
same server as Cisco CallManager. Cisco CRA allows CallManager integration with outside
applications.
Using the Tool for Auto-Registered Phone Support 137
These prerequisites apply to the TAPS installation for BAT:
■ Make sure that the publisher database for Cisco CallManager is configured and running. The
publisher database can reside on its own server or on the same server as Cisco CallManager.
■ Before installing TAPS, ensure that the latest BAT release is installed on the publisher
database server for Cisco CallManager.
■ Have the IP address for the Cisco CallManager publisher server and the private phrase for the
installation procedure.
■ If you are not using the CallManager Extended Services CD, ensure that the Cisco CRA
server is configured. The Cisco CRA application can reside on its own dedicated server or can
be colocated on the same server as Cisco CallManager.
■ Be sure to use the locale installer to create the country-specific TAPS prompts.
■ Ensure that the Windows 2000 Services window is closed.
NOTE Because TAPS relies on advanced CallManager configuration concepts, many of the
details in the steps pertaining to the TAPS installation and configuration, such as CTI ports and
route points, move outside of the CIPT material. Detailed information about TAPS can be found
in the Cisco CallManager Bulk Administration Guide available on the Cisco website.
Complete the following steps to install TAPS:
Step 1 Log in with administrator privileges to the system that is running the
publisher database for Cisco CallManager (where you installed BAT).
Step 2 Access BAT.
Step 3 Choose Applications > Install Plugins.
Step 4 Double-click the TAPS setup icon.
Step 5 Determine whether to copy the TAPS installation executable to the system
or run it from the current location.
Step 6 The Welcome window for the installation wizard opens. This installation
program installs TAPS on the Cisco CallManager publisher server and the
CRA applications server at the same time, if applications are colocated on
the same server. Click Next.
NOTE When you are installing TAPS in a network with a dedicated CRA server, you must
run the TAPS installation program again on the CRA server. Use CRA online help for assistance
with installation and configuration.
Step 7 Enter the private phrase for the Cisco CallManager publisher server in the
Installing Cisco CallManager Components window and click Next. The
Installing TAPS on CCM window displays a progress bar that shows the
status of the installation.
Step 8 The Installation Completed window is displayed when the installation
ends. Click Finish.
After you have completed the TAPS installation, you must configure TAPS by adding a Computer
Telephony Interface (CTI) route point, CTI ports, and users in Cisco CallManager Administration.
One CTI route point and at least one CTI port are required for TAPS.
The following procedure describes how to configure TAPS in Cisco CallManager Administration:
NOTE To use TAPS, verify that auto-registration is enabled in Cisco CallManager. Because
TAPS can replace a DN, you can protect certain DNs from being overwritten by using the
Secure TAPS option. Turning on Secure TAPS keeps a user from overwriting an extension that
is already in use.
Step 1 Create a CTI route point and assign it a unique DN.

Step 2 Choose the Call Forward Busy, Call Forward No Answer, and Call
Forward on Failure options for the operator number on the TAPS CTI
route point.
Step 3 Create one or more CTI ports with consecutive DNs. You can create CTI
ports in BAT or Cisco CallManager Administration.
Step 4 Create a user. The TAPS route point and ports should be in the Controlled
Devices list of the user.
Step 5 Create an auto-registration partition or calling search space or both to
prevent IP Phones that have auto-registered from dialing any DN other than
the number that is assigned to the TAPS CTI route point. Restricting access
to this DN ensures that users download the proper configuration
information for their IP Phones.
NOTE TAPS supports a maximum number of simultaneous sessions equal to the number of
CTI ports that are configured for TAPS. For example, if you have configured five CTI ports, up
to five users can dial into TAPS at the same time. The sixth caller cannot connect to TAPS.
Summary
This chapter discussed the Bulk Administration Tool (BAT) and the Tool for Auto-Registered
Phone Support (TAPS) utilities from Cisco. These tools ease the administrative load of Cisco
CallManager in a huge way by allowing you to make bulk adds, deletes, and changes to the devices
in the CallManager database. By combining a Comma Separated Value (CSV) text file containing
unique values with a template, you can insert devices (or users) in bulk into the CallManager
system. In addition, you can use BAT to define a query matching multiple existing devices and
perform a mass update to chance settings across your IP telephony network.
Cisco has provided TAPS to create a semiautomated system to add IP Phones to your network.
Using TAPS, new IP Phones auto-register with the Cisco CallManager and receive a generic
extension assignment. When you assign a user to an IP Phone, they dial the TAPS number and
walk through an automated process in which they type in the extension number you have assigned
them. TAPS then queries the data from your imported CSV file to find the specific settings that
CallManager should apply to the IP Phone and sets it up with the correct extension and
configuration settings.
Review Questions
1. Which of the following configurations can you add to the CallManager database in bulk using
BAT? (Choose three.)
a. Route Groups
b. Route Patterns
c. Users
d. IP Phones
e. Phone Templates
f. Device Profiles
2. For TAPS to function correctly, what must you enable on at least one Cisco CallManager
server?
a. Voice Media Streaming Application service
b. Auto-Registration
c. Cisco IPCC
d. All of the above
3. Which two pieces of information are required to perform a successful device import using
BAT? (Choose two.)
a. a tab-delimited text file
b. a comma-separated text file
c. a valid device template
d. CallManager 4.x or later
4. When you install BAT, the installation process stores an Excel template on the hard drive of
the CallManager server. Where is the file located?
a. C:\Cisco\BAT
b. C:\Program Files\Cisco\BAT
c. C:\BAT
d. C:\CiscoWebs\BAT\ExcelTemplate
5. On which CallManager server should you install the Cisco BAT utility?
a. the Publisher server
b. a Subscriber server participating in call processing
c. a Subscriber server not participating in call processing
d. a CallManager server not performing database replication
6. You have created an entry in a CSV file that defines an IP Phone with two lines. You combine
this CSV file with a BAT device template that defines an IP Phone with six lines. What
happens?
a. The BAT utility inserts the IP Phone into the CallManager database with only the two
lines defined in the CSV file.
b. The BAT utility inserts the IP Phone into the CallManager database with the six lines
defined in the device template.
c. The BAT utility rejects the configuration and produces an HTTP 4xx error.
d. The BAT utility prompts you for the configuration for the additional four lines.
7. You are using the BAT Excel template to define a CSV file for a bulk import. You notice a
check box in the template that says “Create Dummy MAC Address.” What does this check
box accomplish?
a. It assigns all IP Phones a MAC address of 0000.0000.0000 until the IP Phone auto-reg-
isters.
b. It enters the IP Phones into the CallManager database without a MAC address.
c. It generates random MAC addresses for the IP Phones until the administrator changes
them.
d. It is for use for updates only with the TAPS utility.
8. You want to perform a bulk update to change the music on hold setting for all Cisco IP Phones
to a classic Frank Sinatra song. How do you specify the IP Phones to change in the BAT
utility?
a. by specifying a range of DNs
b. through a query process
c. using a drop-down menu
d. by checking the Select All check box
9. What additional application does CallManager require to support TAPS?
a. Microsoft Exchange 2000
b. Customer Response Application
c. SQL 2000
d. additional phone ports
10. What two methods can you use to create CSV files? (Choose two.)
a. the Microsoft Excel Template
b. the CSV file generator downloadable from CCO
c. the query utility included with BAT
d. a text editor
Part III: IPT Network Integration
and Route Plan
Chapter 8 Cisco Catalyst Switches
Chapter 9 Configuring Cisco Gateways and Trunks
Chapter 10 Cisco Unified CallManager Route Plan Basics
Chapter 11 Cisco Unified CallManager Advanced Route Plans
Chapter 12 Configuring Hunt Groups and Call Coverage
Chapter 13 Implementing Telephony Call Restrictions and Control
Chapter 14 Implementing Multiple-Site Deployments

following topics:
■ Identifying the functions that Cisco Catalyst

switches perform in a Cisco IP telephony
solution
■ The three options for powering Cisco IP

Phones
■ The two types of PoE that Cisco Catalyst

switches provide and the Cisco Catalyst
switches that support PoE
■ Identifying the commands to configure PoE

on Cisco Catalyst switches
■ Configuring dual VLANs on a single port on

a Cisco Catalyst switch so that the IP Phones
reside in a separate VLAN
■ Configuring CoS on Cisco Catalyst switches

so that voice traffic has priority over data
traffic as it travels throughout the network
CHAPTER 8
Cisco Catalyst Switches
Deploying IP telephony requires planning how you will power the IP Phones and how the voice
network traffic will mix with the data network while ensuring that the data traffic does not
degrade the quality of the voice calls.
Cisco Catalyst switches provide three features that aid an IP telephony deployment: inline
power, voice VLANs, and class of service (CoS). Using the Cisco Catalyst switch to power IP
Phones can save on wiring costs and simplifies management. Enabling multiple VLANs in a
single port and placing voice packets in one VLAN and data in another VLAN saves money by
reducing the number of switch ports. Extending CoS to the IP Phone improves voice quality by
ensuring that voice packets receive priority over data.
This lesson discusses the three major functions that Cisco Catalyst switches perform in an IP
telephony network and describes how to configure a Cisco Catalyst switch to enable these
functions.
Catalyst Switch Role in IP Telephony

Cisco voice-capable Catalyst switches can provide three primary features to assist you with your
IP telephony deployment:
■ Inline power—Inline power capabilities allow a Cisco Catalyst switch to send power
through the Category 3 (or better) copper cabling to a Cisco IP Phone or other inline power-
compatible devices (such as wireless access points) without the need for an external power
supply. Inline power is also referred to as Power over Ethernet (PoE). Inline power was
developed in 2000 by Cisco to support the emerging IP telephony solution and was later
standardized by the IEEE as the 802.3af standard.
■ Auxiliary VLAN support—Auxiliary VLAN support allows a switch to support multiple

VLANs on a single access port. You can connect a PC to the back of the Cisco IP Phone
because some Cisco IP Phones have built-in switches. Auxiliary VLANs allow you to place
the IP Phone, and the devices that are attached through the IP Phone, on separate VLANs.
146 Chapter 8: Cisco Catalyst Switches
■ Class of service (CoS) marking—CoS marking is data link layer (Layer 2) marking that is
used to prioritize network traffic. Prioritizing voice traffic is critical in IP telephony networks.
If voice traffic is not given priority, poor voice quality may result because voice frames wait
in the switch queue behind large data frames. The switch can use existing CoS marking to
prioritize network traffic and can also classify and mark traffic that it receives.
Powering the Cisco IP Phone

Most Cisco IP Phone models are capable of using the following three options for power:
■ PoE—With PoE, the phone plugs into the data jack that connects to the switch, and the user
PC in turn connects to the IP Phone. Power-sourcing equipment (PSE), such as Cisco Catalyst
PoE-capable modular and fixed-configuration switches, insert power over the Ethernet cable
to the powered device, for example, an IP Phone or 802.11 wireless access point.
■ Midspan power injection—Because many switches do not support PoE, the powered device
must support a midspan power source. This midspan device sits between the LAN switch and
the powered device and inserts power on the Ethernet cable to the powered device. A technical
difference between the midspan and inline power mechanism is that power is delivered on the
spare pairs (pins 4, 5, 7, and 8). An example of midspan PSE is a Cisco Catalyst Inline Power
Patch Panel.
■ Wall power—Wall power needs a DC converter for connecting the IP Phone to a wall outlet.
NOTE You must order the wall power supply separately from the Cisco IP Phone. This can
increase the cost of the end devices significantly. Before you decide to use wall power rather than
PoE to power the Cisco IP Phones, be sure to verify the total cost of this decision.
Two Types of PoE Delivery

Cisco provides two types of inline power delivery: the Cisco original implementation and the
IEEE 802.3af PoE standard. You can refer to both inline power types as PoE.
Because Cisco was the first to develop PoE, there was not an established PoE industry standard to
which Cisco could conform. Because of this, the industry considers Cisco’s original
implementation PoE as prestandard and proprietary. Although Cisco will eventually discontinue
this prestandard implementation, Cisco devices will continue to support it for many years to come.
The Cisco prestandard PoE implementation supports the following features:
■ Provides –48 V DC at up to 6.3 to 7.7 watts (W) per port over data pins 1, 2, 3, and 6.
■ Supports most PoE-capable Cisco devices (IP Phones and wireless access points).
■ Uses a Cisco proprietary method of determining whether an attached device requires power.
Power is delivered only to devices that require power.
Since first developing PoE, Cisco has been driving the evolution of this technology toward
standardization by working with the IEEE and member vendors to create a standards-based means
of providing power from an Ethernet switch port. The IEEE 802.3af committee has ratified this
capability. The IEEE 802.3af PoE standard supports the following features:
■ Specifies –48 V DC at up to 15.4 W per port over data pins 1, 2, 3, and 6 or the spare pins 4,
5, 7, and 8 (a PSE can use one or the other, but not both).
■ Enables a new range of Ethernet-powered devices that consume additional power.
■ Standardizes the method of determining whether an attached device requires power. The PSE
delivers power only to devices that require power. This type has several optional elements,
such as power classification, where powered devices can optionally support a signature that
defines the maximum power requirement. PSE that supports power classification reads this
signature and budgets the correct amount of power per powered device, which will likely be
significantly less than the maximum allowed power.
Without power classification, the switch reserves the full 15.4 W of power for every device. This
behavior might result in oversubscription of the available power supplies so that some devices will
not be powered even though there is sufficient power available.
Power classification defines these five classes:
0 (default)—15.4 W reserved
1—4 W
2—7 W
3—15.4 W
4—Reserved for future expansion
All Cisco IEEE 802.3af-compliant switches support power classification.
TIP The Cisco Power Calculator is an online tool that enables you to calculate the power
supply requirements for a specific PoE configuration. The Cisco Power Calculator is available
to registered Cisco.com users at www.cisco.com/go/poe.
PoE Device Detection

The Cisco prestandard PoE and industry standard IEEE 802.3af PoE have slightly different
methods of detecting an inline power-capable device. Figure 8-1 illustrates how a Cisco
prestandard Catalyst switch detects a Cisco IP Phone, wireless access point, or other inline power-
capable device. When a switch port that is configured for inline power detects a connected device,
the switch sends an Ethernet Fast Link Pulse (FLP) to the device. The Cisco powered device (IP
Phone) loops the FLP back to the switch to indicate its inline power capability. The switch then
delivers –48 V DC PoE (inline) power to the IP Phone or other inline power-capable endpoint.
Figure 8-1 Cisco Prestandard PoE Device Detection

Cisco Prestandard
Implementation
Powered Device Port
Pin3
FLP Pin6
Rx
Switch Pin1 FLP
It is an inline device. Pin2 Tx
A Cisco Catalyst IEEE 802.3af-compliant switch detects a Cisco IP Phone, wireless access point,
or other inline power-capable device through a very similar process. The PSE (Cisco Catalyst
switch) detects a powered device by applying a voltage in the range of –2.8 V to –10 V on the cable
and then looks for a 25K ohm signature resistor rather than using the Cisco proprietary FLP signal.
Compliant powered devices must support this resistance method. If the appropriate resistance is
found, the Cisco Catalyst switch delivers power.
Catalyst Family of PoE Switches

The Cisco Catalyst LAN switching portfolio is the industry-leading family of intelligent switching
solutions delivering a robust range of security and quality of service (QoS) capabilities. The Cisco
Catalyst switch portfolio allows organizations to enable new business applications and integrate
new technologies such as wireless and IP telephony into their network infrastructure. Here are the
switches in the Cisco Catalyst family:
■ Cisco Catalyst modular switching—The Cisco Catalyst 6500 Series delivers a 96-port
10BASE-T/100BASE-T line card and 48-port 10BASE-T/100BASE-T and 10BASE-T/
100BASE-T/1000BASE-T line cards. The Catalyst 6500 Series offers a modular PoE
daughter card architecture for the 96-port card and the 48-port 10/100/1000 card. The Cisco
Catalyst 4500 Series delivers 48-port 10/100 and 10/100/1000 line cards. All line cards
support both IEEE 802.3af and Cisco prestandard inline power. The cards are compatible with
any Cisco Catalyst 6500 or 4500 chassis and Supervisor Engine. The Cisco Catalyst modular
chassis switches can deliver 15.4 W per port for all 48 ports on a module simultaneously.
■ Cisco Catalyst stackable switching—The Cisco Catalyst 3750 Series offers 48- and 24-port
Fast Ethernet switches that comply with IEEE 802.3af and Cisco prestandard PoE. The Cisco
Catalyst 3560 Series offers 48- and 24-port Fast Ethernet switches that support both the
industry standard and Cisco standard PoE.
■ Cisco EtherSwitch modules—The Cisco 36- and 16-port 10/100 EtherSwitch modules for
Cisco 2600 and 3700 Series routers offer branch office customers the option to integrate
switching and routing in one platform. These modules can support Cisco prestandard PoE and
provide straightforward configuration, easy deployment, and integrated management in a
single platform. The Cisco 2600 Series requires a separate external PoE power supply; the
Cisco 3700 Series integrates the power supply. Cisco has also released EtherSwitch modules
for the newer Integrated Service Routers (ISRs). These new EtherSwitch modules support
both the Cisco prestandard PoE and the IEEE 802.3af PoE.
Table 8-1 lists the Cisco Catalyst PoE options.
Table 8-1 Catalyst Switch PoE Options
PoE Option Cisco Cisco Cisco Cisco Cisco

Catalyst 6500 Catalyst 4500 Catalyst Catalyst EtherSwitch
3750 3560 Module
PoE Configuration 48-, 96-port 10/ 48-port 10/100 24-, 48-port 24-, 48-port 16-, 36-port
Options 100 or 48-port or 10/100/1000 10/100 10/100 10/100
10/100/1000
IEEE 802.3af- Yes Yes Yes Yes No (older series)
Compliant
Yes (ISRs)
Cisco Prestandard PoE Yes Yes Yes Yes Yes
TIP The switches that are listed here also support multiple VLANs per port and QoS
capabilities.
NOTE The pace of change in IP telephony is intense. By the time you are reading this text,
Cisco will have most likely introduced multiple new product lines and switch models that
support PoE and many other features. Always be sure to check the Cisco website for the latest
product information.
Configuring PoE
By default, PoE is enabled on all Cisco devices that support the PoE feature. The default mode for
PoE is auto, which means the switch will automatically detect if a device is PoE capable and
supply power, if necessary. If you are using a switch that is running Cisco Catalyst Operating
System software (CatOS), use the following syntax to modify the default PoE settings:
CatOS>(enable) set port inlinepower <mod//port> ?

auto Port inline power auto mode
off Port inline power off mode
The two modes are auto and off. In the off mode, the switch does not power up the port even if an
unpowered phone is connected. In the auto mode, the switch powers up the port only if the
switching module has discovered the phone. Examples of devices running Cisco CatOS include
the Cisco Catalyst 6500, 4500, and 4000 Series.
TIP It can be useful to turn PoE capabilities off on ports that you are sure will never use the
feature. If the power supply your switch is equipped with is unable to extend power to all ports,
you can specify ports that should receive power. Otherwise, the switch will allocate power to
ports with lower port numbers until it exhausts the available power supply leaving the higher
port numbers unpowered. The switch allocates power to ports configured with the “auto” setting
regardless of whether the port is using the power.
If you are using a switch that is running Cisco IOS (NativeIOS), use the following syntax to
modify the default PoE settings:
NativeIOS(config-if)# power inline [a

auto | never ]
Use the power inline command on switches that are running native Cisco IOS software (examples
include the Catalyst 6500, 4500, 3750, and 3560 switches). The powered device-discovery
algorithm is operational in the auto mode. The powered device-discovery algorithm is disabled in
the never mode. Other modes exist for allocating power, depending on the version of Cisco IOS,
for example, the ability to allocate power on a per-port basis with the allocation milliwatt
command.
NOTE The Catalyst 6500 Series can run either Cisco Catalyst Operating System software or
native Cisco IOS software if the switch Supervisor Engine has a Multilayer Switch Feature Card
(MSFC). Otherwise, these switches can run only Cisco Catalyst software. The Cisco Catalyst
4500 and 4000 Series can also run Cisco Catalyst software or native Cisco IOS software,
depending on the Supervisor Engine. Generally, late-edition Supervisor Engines run native
Cisco IOS software; however, you should check the product documentation to determine the
Supervisor Engine and the operating system that is supported on your specific model.
Verifying PoE
You can use the following command to display a view of the power allocated on Cisco Catalyst
switches running the CatOS:
CatOS>(enable) show port inline power 7

Default Inline Power allocation per port: 10.000 Watts (0.23 Amps @42V)
Total inline power drawn by module 7: 75.60 Watts (1.80 Amps @42V)
Port InlinePowered PowerAllocated
Admin Oper Detected mWatt mA @42V
---- ----- ---- -------- --------- -----------
7/1 auto off no 0 0
7/2 auto on yes 6300 150
7/3 auto on yes 6300 150
7/4 auto off no 0 0
7/5 auto off no 0 0
7/6 auto off no 0 0
7/7 auto off no 0 0
You can use the following command to display a view of the power allocated on Cisco Catalyst
switches running the NativeIOS:
show power inlin e

NativeIOS#s
Available:360(w) Used:22(w) Remaining:338(w)
Interface Admin Oper Power Device Class Max

--------- ------ ---------- ------- ------------------- ----- ----
Fa0/1 auto off 0.0 n/a n/a 15.4
Fa0/8 auto on 10.3 IP Phone 7970 15.4
Fa0/10 auto on 6.3 IP Phone 7960 n/a 15.4
Fa0/14 auto on 6.3 IP Phone 7960 n/a 15.4
Table 8-2 provides a brief description of the output.
Table 8-2 Inline Power Output Descriptions
Output Description
Port Identifies the port number on the module
Inline Powered Admin Identifies the port configuration from using the set
inlinepower mod/port [auto | off] command
Oper Identifies whether the inline power is operational
Power Detected Identifies whether power is detected
Allocated
mWatt/Watts Identifies the milliwatts (CatOS) or Watts (NativeIOS)
supplied on a given port
mA @42V Identifies the milliamps at 42 V supplied on a given port (the
actual voltage is –48 V)
Data and Voice VLANs

All data devices typically reside on data VLANs in the traditional switched scenario. You might
need a separate voice VLAN when you combine the voice network into the data network.
Although you can think of it as a voice VLAN, in the future, other types of nondata devices will
reside in the voice VLAN.
NOTE The Cisco Catalyst software (CatOS) refers to this new voice VLAN as the auxiliary
VLAN for configuration purposes.
The placement of nondata devices (such as IP Phones) in a voice VLAN makes it easier for
customers to automate the process of deploying IP Phones. IP Phones will boot and reside in the
voice VLAN if you configure the switch to support them, just as data devices boot and reside in
the access (data) VLAN. The IP Phone communicates with the switch via Cisco Discovery
Protocol when it powers up. The switch provides the telephone with the appropriate VLAN ID.
NOTE Although a voice VLAN is not required, it is encouraged by Cisco to isolate voice
traffic for QoS and security purposes. It might also be impossible to put more devices on the
existing data VLAN due to address space depletion in the data subnet DHCP scope, in which
case, the voice VLAN becomes imperative.
Administrators can implement multiple VLANs on the same port by configuring trunk port. A
tagging mechanism must exist to distinguish among VLANs on the same port. 802.1Q is the IEEE
standard for tagging frames with a VLAN ID number. The IP Phone sends tagged 802.1Q frames.
The PC sends untagged frames and the switch adds the access VLAN tag before forwarding
toward the network. When the switch receives a frame from the network destined for the PC, it
removes the access VLAN tag before forwarding the frame to the PC.
There are some advantages in implementing dual VLANs:
■ This solution allows for scalability of the network from an addressing perspective. IP subnets
usually have more than 50 percent (often more than 80 percent) of their IP addresses
allocated. A separate VLAN (separate IP subnet) to carry the voice traffic allows you to
introduce a large number of new devices, such as IP Phones, into the network without
extensive modifications to the IP addressing scheme.
■ This solution allows for the logical separation of data and voice traffic, which have different
characteristics. This separation allows the network to handle these two traffic types
individually.
■ This solution allows you to connect two devices to the switch using only one physical port
and one Ethernet cable between the wiring closet and the IP Phone or PC location.
Data and Voice VLANs 153
Configuring and Verifying Dual VLANs Using the CatOS

Configure auxiliary VLAN ports in Cisco Catalyst software 5.5 and later using the set port
auxiliaryvlan command to configure the auxiliary VLAN ports:
set port auxiliaryvlan [mod/port] {vlan | untagged | dot1p | none}
Table 8-3 provides a brief description of the syntax.
Table 8-3 set port auxiliaryvlan Command Syntax Description
Syntax Description
[mod/port] Number of the module and (optional) ports
vlan Number of the VLAN; valid values are from 1 to 1000
untagged Keyword to specify that the IP Phone 7960 sends untagged packets without 802.1p
priority
dot1p Keyword to specify that the IP Phone 7960 sends packets with 802.1p priority
none Keyword to turn off auxiliary VLAN tagging
For example, if you want to configure a 6500 switch using the CatOS with a voice VLAN of 222
for all 48 ports on Module 7, you can use the command in Example 8-1.
Example 8-1 Auxiliary VLAN configuration (CatOS)

CatOS>(enable) set port auxiliaryvlan 7/1-48 222
Auxiliaryvlan 222 configuration successful.
AuxiliaryVlan AuxVlanStatus Mod/Ports
------------- ------------- ----------------------
222 active 7/1-48
You can check the status of the auxiliary VLAN on a port or module in one of two ways:
■ Use the show port auxiliaryvlan vlan-id command to show the status of that auxiliary VLAN
and the module and ports where it is active.
■ Use the show port [module[/port]] command to show the module, port, and the auxiliary
VLAN and the status of the port.
Configuring and Verifying Dual VLANs Using the NativeOS

Use the commands in Example 8-2 to configure voice and data VLANs on the single-port interface
of a switch that is running native Cisco IOS software. These commands apply the same
functionality as setting a port to use an auxiliary VLAN on a Cisco Catalyst switch that is running
Cisco Catalyst software.
Example 8-2 Voice VLAN configuration (NativeIOS)

inte rface FastE thernet0/1
NativeIOS(config)#i
switchport mode access
NativeIOS(config-if)#s
sw itchport vo ice vlan 261
switchport access vlan 262
spanni ng-tree por tfast
Table 8-4 provides a brief description of these commands.
Table 8-4 Configuring Dual VLANs Using the NativeOS Command Descriptions
Command Description
switchport mode access Configures the switchport to be an access (nontrunking) port.
switchport voice vlan Configures the switchport with the voice VLAN (261 in this example)
voice-VLAN_ID to be used for voice traffic. The range is 1 to 4094.
switchport access vlan Configures the interface as a static access port with the access VLAN
data_VLAN_ID ID (262 in this example); the range is 1 to 4094.
spanning-tree portfast Causes a port to enter the spanning-tree forwarding state immediately,
bypassing the listening and learning states. You can use PortFast on
switch ports that are connected to a single workstation or server (as
opposed to another switch or network device) to allow those devices
to connect to the network immediately.
You can verify your voice VLAN configuration on the Cisco Catalyst switches that are running
native Cisco IOS software by using the show interfaces mod/port switchport command, as
displayed in Example 8-3.
Example 8-3 Voice VLAN Verification (NativeIOS)

show interfaces fa0/4 swit chport
NativeIOS#s
Name: Fa0/4
Switchport: Enabled
Administrative Mode: static access
Operational Mode: static access
Administrative Trunking Encapsulation: negotiate
Operational Trunking Encapsulation: native
Negotiation of Trunking: Off
Access Mode VLAN: 262 (VLAN0262)
Trunking Native Mode VLAN: 1 (default)
Voice VLAN: 261 (VLAN0261)
<output truncated>
Configuring Class of Service 155
Configuring Class of Service

CoS is a data link layer marking that you can use to classify traffic as it passes through a switch.
You should ensure that voice traffic has priority as it travels throughout your network because it is
extremely sensitive to delay. Cisco IP Phones send all voice packets tagged with CoS 5 by default,
which is the highest level of CoS that Cisco recommends for user traffic.
The multi-VLAN port also receives packets from the devices (PCs and workstations) that are
connected to the PC port of the IP Phone. The attached device, if it is not in the native VLAN, can
send packets with a CoS equal to or higher than the packets that are being sent by the IP Phone,
which can cause severe voice quality problems on your IP telephony network. This can be done
only if the device is not in the access VLAN or is sharing the same VLAN as the IP Phone.
Cisco Catalyst switches have the ability to extend the boundary of trust to the IP Phone. You can
use the switch to instruct the IP Phone to accept the CoS value of frames that are arriving from
connected devices (trust) and allow the CoS to remain unchanged. Alternatively, you can choose
not to trust the attached device and set the CoS to 0 or set the CoS to a configured value that you
determine.
The Cisco Catalyst switch uses Cisco Discovery Protocol to send this configuration information
to the IP Phone. The switch sends an additional Cisco Discovery Protocol packet to the IP Phone
whenever there is a change in the CoS configuration.
The switch uses its queues, which are available on a per-port basis, to buffer incoming and
outgoing frames. The switch can use the CoS values to place the frames in the appropriate queues.
Voice frames should be placed in the priority queue for minimal delay.
NOTE Setting CoS markings represents one small piece of the big picture of QoS. For
detailed information regarding QoS design and configuration, check out Cisco QOS Exam
Certification Guide, Second Edition published by Cisco Press (ISBN: 1-58720-124-0) and the
online Cisco QoS Solution Reference Network Design Guide (SRND) at http://www.cisco.com/
go/srnd.
You can configure the switch with different QoS settings on a per-port basis. In the CatOS, use the
set port qos [mod/port] cos command to set the default value for all packets that have arrived
through an untrusted port. For example, the command set port qos 3/1 cos 3 sets the CoS value to
3 on port 3/1. The cos-value specifies the CoS value for a port; valid values are from 0 to 7. Seven
is the highest priority. The default is 0 (not trusted).
Use the set port qos [mod/ports] trust-ext {trusted | untrusted} command to either extend trust
to the PC by specifying that all traffic received through the access port passes through the phone
switch unchanged, or to not trust the port and change the CoS value to 0.
In the NativeIOS, use the switchport priority extend {cos cos-value | trust} interface
configuration command to set a port priority for the incoming frames received by the IP Phone
connected to the specified port. The cos-value is used to set the IP Phone port to override the
priority received from the PC or the attached device. Valid values for the cos-value are from 0 to
7. Seven is the highest priority. The default is 0 (not trusted). Alternatively, the trust keyword
causes the IP Phone port to trust the priority received from the PC or the attached device, that is,
to not change the CoS value. Figure 8-2 provides a visual representation of the different CoS
behaviors the IP Phone can support.
Figure 8-2 Extending QoS to the IP Phone

Phone VLAN = 200 PC VLAN = 3
CoS = 5 CoS = 7
V Access port
IP
IP Phone: Desktop PC:
IP Subnet B IP Subnet A
PC is not trusted CoS = 0 CoS = 7

CoS set to 0 (default)
IP
PC is not trusted CoS = 2 CoS = 7

CoS set to 2
IP
CoS = 7 CoS = 7
PC is trusted
IP
Summary
This chapter covered the role of the Catalyst switch platform in a Cisco IP telephony design. To
support the new voice data traversing the network, Cisco has equipped their mainline switch
platforms with three new capabilities:
■ The ability to support inline power—Allows a Cisco Catalyst switch to send power through
the Ethernet copper to a Cisco IP Phone or other inline power-compatible devices using either
the Cisco prestandard PoE or the IEEE 802.3af industry standard PoE.
■ The ability to support auxiliary/voice VLANs—Allows a Cisco Catalyst switch to support

multiple VLANs on a single port to separate voice and data traffic.
■ The ability to support CoS/QoS capabilities—Allows a Cisco Catalyst switch to mark

voice and data traffic with CoS values that other equipment can use to prioritize network
traffic.
These capabilities play a vital role in ensuring the voice network operates seamlessly with the
highest possible quality and reliability.
Review Questions
1. Which command enables inline power on port 3 of module 2 of a Cisco Catalyst 6000 Series
switch that is running Cisco Catalyst software?
a. power inline in global configuration mode
b. power inline auto in interface configuration mode
c. set port inline power default
d. set port inline power 3/2 802.3
e. set port inline power 2/3 auto
2. Cisco Catalyst switches can provide which three functions in an IP telephony deployment?
(Choose three.)
a. Convert SCCP signaling packets from the IP Phone to MGCP.
b. Power IP Phones through the same Ethernet cable that carries data.
c. Enable the classification and prioritization of voice packets at Layer 2.
d. Instruct the IP Phone to change the CoS of an incoming data frame.
e. Determine available bandwidth before placing a call over the WAN.
3. What are three requirements for configuring dual VLANs on a single port that is attached to
an IP Phone? (Choose three.)
a. Configure the interface for access and voice VLANs.
b. Tag the voice packets with a voice VLAN identifier.
c. Extend the CoS boundary to the IP Phone.
d. Ensure that the IP Phone has an internal switch.
e. Ensure that the IP Phone is inline power-capable.
4. What three methods can you use to power all PoE-capable Cisco IP Phones?
a. wall power
b. solar power
c. Cisco prestandard PoE
d. IEEE 802.3af PoE
e. Cisco Inline Power Patch Panel
5. Which of the following represent the industry standard PoE mechanism?

a. 802.3
b. 802.11
c. 802.11b
d. 802.3af
6. Which of the following commands will activate PoE on a Catalyst 3750 switch?
a. set port inlinepower mod/port auto
b. set port inlinepower mod/port on
c. power inline auto
d. power inline on
7. To support dual VLANs from a Catalyst 6500 running CatOS and a Cisco IP Phone, what
should be configured? (Choose two.)
a. an ISL trunk
b. an 802.1Q trunk
c. an auxiliary VLAN configuration
d. a voice VLAN configuration
8. You are attempting to set up a dual VLAN configuration between a Cisco IP Phone and a
CatOS-based 6500 switch. You enter the command set port auxiliaryvlan 3/1 50 to place the
IP Phone on the voice VLAN 50. What more must be done?
a. Configure an 802.1Q trunk.
b. Configure an ISL trunk.
c. Enable PoE.
d. Nothing more must be done.
9. By default, a Cisco IP Phone tags all voice packets with what CoS value?
a. 1
b. 3
c. 5
d. 7
10. How does the Cisco IP Phone mark traffic from an attached PC?
a. The Cisco IP Phone trusts the CoS marking coming from the attached PC.
b. The Cisco IP Phone does not trust the CoS marking and sets it to 0.
c. The Cisco IP Phone does not trust the CoS marking and sets it to 3.
d. The Cisco IP Phone does not support CoS markings for data traffic.
following topics:
■ Describing the role of the gateway in an IP

telephony infrastructure
■ Comparing analog and digital gateways with
respect to interfaces and the types of telephony
devices that they can connect
■ Describing the core gateway requirements that
a gateway must have to support an IP
telephony network
■ Describing the gateway communication
protocols that Cisco CallManager supports
■ Adding and configuring H.323, MGCP, and
non-IOS MGCP gateways in Cisco
CallManager and on the gateway to interface
to the IP WAN and PSTN and support analog
phones and faxes
■ Choosing the appropriate trunk type to satisfy
deployment requirements for call-control
requirements and call routing
■ Configuring non-gatekeeper-controlled trunks
in Cisco CallManager Administration to
enable calling to remote clusters across an IP
WAN
■ Describing the SIP signaling protocol and how
it is implemented in Cisco CallManager
■ Explaining how to create a SIP trunk and route
calls to a SIP endpoint
CHAPTER 9
Configuring Cisco
Gateways and Trunks
Voice gateways and trunks provide the bridge between an IP telephony network and the public
switched telephone network (PSTN), and between Cisco CallManager clusters. This chapter
prepares you for integrating Cisco gateways and trunks in a Cisco CallManager solution.
You will learn about analog and digital gateways, recommended gateway requirements, gateway
protocols, the types of trunks that Cisco CallManager supports, and how to configure gateways,
intercluster trunks, and Session Initiation Protocol (SIP) trunks.
Cisco Gateway Concepts

A gateway is a device that translates one type of signal into another type of signal. One type of
gateway is the voice gateway. A voice gateway is a router or switch that converts IP voice
packets to analog or digital signals that are understood by trunks or stations. Gateways are used
in several situations, for example, connecting to a PSTN or PBX, or connecting individual
devices such as an analog phone or fax.
NOTE This chapter provides an overview of the voice gateways that you can use with the
Cisco CallManager system and describes their basic configuration. For more information on
configuring voice gateways, refer to Authorized Self-Study Guide: Cisco Voice over IP
(CVOICE), Second Edition (ISBN: 1-58705-262-8) or Cisco Voice Gateways and
Gatekeepers (ISBN: 1-58705-258-X) from Cisco Press.
Analog and Digital Gateways

There are two types of Cisco access gateways: analog and digital. At a high level, the difference
between these two types of gateways is their capacity. Analog gateways (and analog
connections) receive one line per port. This port could be a connection to the PSTN, an end
162 Chapter 9: Configuring Cisco Gateways and Trunks
device such as a fax machine, or a link to a PBX. Regardless of the connection, analog ports only
permit one voice path per line. There are two categories of analog gateways:
■ Access analog station gateways—Access analog station gateways connect Cisco

CallManager to plain old telephone service (POTS) analog telephones, interactive voice
response (IVR) systems, fax machines, and voice-mail systems. Station gateways provide
Foreign eXchange Station (FXS) ports for connecting to analog devices such as telephones
and faxes.
■ Access analog trunk gateways—Access analog trunk gateways connect Cisco CallManager
to PSTN central office (CO) or PBX trunks. Trunk gateways provide Foreign eXchange Office
(FXO) ports for PSTN or PBX access and E&M (known by various names, primarily receive
and transmit, or ear and mouth, or earth and magneto) ports for analog trunk connection to a
legacy PBX. Analog Direct Inward Dial (DID) is also available for PSTN connectivity.
Depending on the cards you have installed into a gateway, it could fill both station (FXS) and trunk
(FXO) roles.
On the flip side, digital gateways typically allow multiple calls per port. These gateways connect
the Cisco CallManager to the PSTN or to a PBX via digital trunks, such as PRI common channel
signaling (CCS), BRI, T1 channel-associated signaling (CAS), or E1. Digital T1 PRI trunks might
also connect to certain legacy voice-mail systems. Regardless of the connection type, you will
typically use a digital gateway to provide a high-capacity connection to an alternate network type.
Core Gateway Requirements

To support modern VoIP networks, your IP telephony gateways must meet these core feature
requirements:
■ Dual tone multifrequency (DTMF) relay capabilities—DTMF signaling tones, which

contain the digits a user has dialed, must be processed. Gateways must separate DTMF digits
from the voice stream and then send the signaling in voice over IP (VoIP) signaling protocols,
such as H.323, Cisco IOS software Media Gateway Control Protocol (MGCP), SIP, and so on.
■ Supplementary services support—These services are typically basic telephony functions,

such as hold, transfer, and conferencing.
■ Cisco CallManager redundancy support—Cisco CallManager clusters provide for Cisco

CallManager redundancy. The gateways must support the ability to re-home to a secondary
Cisco CallManager in the event of a primary Cisco CallManager failure, which differs from
call survivability in the event of a Cisco CallManager or network failure.
Cisco Gateway Concepts 163
■ Call survivability in Cisco CallManager—The voice gateway preserves the Real-time

Transport Protocol (RTP) bearer stream (the voice conversation) between two IP endpoints
when the Cisco CallManager that the endpoint is registered to is no longer reachable.
Any IP telephony gateway that you select for an enterprise deployment should support these core
requirements. In addition, every IP telephony implementation has its own site-specific feature
requirements, such as analog or digital access, DID, and capacity requirements.
Gateway Communication Overview

For Cisco CallManager to reach other networks (including the PSTN), it must be able to
communicate with the gateways connected to these networks. Cisco CallManager 4.x supports
these three types of gateway communication protocols:
■ H.323—H.323 uses a peer-to-peer model. You perform most of the configuration through
Cisco IOS software on the voice gateway device. With the peer-to-peer model, Cisco
CallManager does not have control over the gateway, which limits the Cisco CallManager
feature support on H.323 gateways. For example, H.323 gateways do not support call
survivability, and only devices that support H.323 version 2 (H.323v2) can take advantage of
Cisco CallManager supplementary services, such as hold, transfer, and conference features.
However, H.323 gateways support additional Cisco IOS features outside of Cisco
CallManager that the other gateways do not, such as call admission control, fractional PRI
support, FXO caller-id support, NFAS signaling, and Cisco Survivable Remote Site
Telephony (SRST).
Examples of Cisco gateway devices that support H.323 include the Cisco VG224 Analog
Phone Gateway (FXS only), Cisco 2600, Cisco 2800, Cisco 3700, and Cisco 3800 devices.
■ MGCP—MGCP uses a client/server model, with voice-routing intelligence that resides in a
call agent (the Cisco CallManager). Because of its centralized architecture, MGCP simplifies
the configuration of voice gateways (the gateway requires no advanced dial-peer
configuration) and supports multiple (redundant) call agents in a network. MGCP gateways
provide call survivability (the gateway maintains calls during failover and fallback). If the
MGCP gateway loses contact with its Cisco CallManager, it falls back to using H.323 control
to support basic call handling of FXS, FXO, T1 CAS, and T1 and E1 PRI interfaces. In
addition, MGCP controlled gateways can support the Q Signaling (QSIG) protocol, which is
used to communicate with many PBX systems.
Examples of Cisco gateway devices that support MGCP are the Cisco VG224 (FXS only),
Cisco 2600, Cisco 2800, Cisco 3700, and Cisco 3800 devices. Examples of non-IOS
devices that support MGCP are the Cisco Catalyst 6000 WS-X6608-T1 and -E1.
■ SCCP—Skinny Client Control Protocol (SCCP, or Skinny), is a client/server protocol that

uses Cisco proprietary messages to communicate between IP devices and Cisco CallManager.
The Cisco IP Phone is an example of a device that registers and communicates with Cisco
CallManager as an SCCP client. During registration, a Cisco IP Phone receives its line and all
other configurations from Cisco CallManager. After it registers, it is notified of new incoming
calls and can make outgoing calls. SCCP is used for VoIP call signaling and enhanced features
such as message waiting indication (MWI).
Examples of Cisco devices that support SCCP are the Cisco VG224 Analog Phone
Gateway (FXS only) and the Cisco VG248 Analog Phone Gateway. The Cisco VG224
gateway is 24-port gateway for analog phones, fax machines, modems, and speakerphones
using Cisco CallManager or Cisco CallManager Express. The Cisco VG248 device is a 48-
port gateway.
NOTE SIP can also be used as a gateway control protocol. Most Cisco IOS images that
support H.323 and MGCP also support SIP. Cisco CallManager 4.x supports SIP trunks to
connect CallManager to distributed SIP networks.
Most gateway devices support multiple gateway protocols. Selecting the protocol to use depends
on site-specific requirements and your installed base of equipment. You might prefer MGCP to
H.323 because of the simpler configuration of MGCP or its support for call survivability during a
Cisco CallManager switchover from a primary to a secondary Cisco CallManager. In addition, you
might prefer H.323 to MGCP because of the interface robustness of H.323 or the ability to use it
with call admission control or SRST. The Cisco-recommended best practices direct corporations
to use MGCP unless they have a specific reason to choose another protocol.
Configuring Access Gateways

To make the Cisco CallManager aware of the gateways it can use, you must add them to the
configuration. You must add the gateways to the CallManager database based on the protocol the
CallManager uses to communicate with the gateway. After you have added the gateway to the
CallManager configuration, you can include it in your route plan design to allow your IP Phones
to dial off the local network. The full route plan design and configuration is covered in Chapters
10-13.
NOTE This section focuses on configuring the CallManager to communicate with the
gateway. You must have also configured the gateway to communicate with the CallManager and
additional networks such as a PBX or the PSTN. Although this book presents brief configuration
examples, the full gateway configuration is covered in depth in Authorized Self-Study Guide:
Cisco Voice over IP (CVOICE), Second Edition and Cisco Voice Gateways and Gatekeepers.
Configuring Access Gateways 165
H.323 Gateway Configuration

Complete these steps to add an H.323 gateway to your CallManager configuration:
Step 1 Choose Gateway from the Device menu in the Cisco CallManager
Administration window.
Step 2 Click the Add a New Gateway link and choose H.323 Gateway from the
Gateway Type menu.
Step 3 Cisco CallManager automatically populates the Device Protocol field with
H.225. Click Next.
Step 4 In the Device Name field shown in Figure 9-1, enter the IP address
(recommended to remote DNS reliance) or hostname of the Cisco router
that will be acting as the gateway.
Figure 9-1 CallManager H.323 Gateway Configuration
Step 5 Assign the gateway to a device pool.

Step 6 Click Insert when finished.
TIP The prior configuration allows you to add an H.323 gateway to the CallManager
configuration. However, many more configuration options are available. For information on all
the settings available in the Gateway Configuration window, refer to the Cisco CallManager
“Help for this Page” option (available from the CCMAdmin Help menu). This page provides a
brief explanation for each configuration field available.
Because H.323 is a peer-to-peer protocol, you must configure most of the gateway configuration
using Cisco IOS software on the gateway itself. Table 9-1 lists the Cisco IOS commands you can
use to configure an H.323 voice gateway.
Table 9-1 Cisco IOS Commands to Configure an H.323 Voice Gateway
Command Description
H323_GW(config)# gateway Enables H.323 VoIP gateway functionality. After you enable the
gateway, it attempts to discover a gatekeeper by using an H.323
gatekeeper request message.
H323_GW(config)# voice class Creates an H.323 voice class that is used to configure a TCP
h323 tag timeout duration.
H323_GW(config-class)# h225 Configures the H.225 TCP timeout duration in seconds. Possible
tcp timeout seconds values are 0 to 30. The default is 15. If you specify 0, the H.225
TCP timer is disabled.
When the duration (seconds) of the H.225 TCP is exceeded, the

voice gateway will use the next ordered dial peer (controlled via
the preference command), which points to a backup Cisco
CallManager.
H323_GW(config)# dial-peer Creates a VoIP dial peer.
voice tag voip
H323_GW(config-dial-peer)# Assigns the previous created voice class to this dial peer.
voice class h323 tag
H323_GW(config-dial-peer)# Configures the dial string that this dial peer matches.
destination-pattern dial-string
H323_GW(config-dial-peer)# Identifies the IP address to route a call to when the destination
session target ipv4:ccm-ip- pattern in the previous command is matched. The IP address is
address the address of the Cisco CallManager on an H.323 gateway.
H323_GW(config-dial-peer)# Assigns a preference to a dial peer when multiple dial peers
preference 0–10 contain the same destination pattern but different session targets.
0 is the highest, 10 is the lowest (used to configure Cisco
CallManager redundancy on H.323 gateways).
Table 9-1 Cisco IOS Commands to Configure an H.323 Voice Gateway (Continued)
Command Description
H323_GW(config-dial-peer)# Configures the gateway to use out-of-band DTMF relay. DTMF
dtmf-relay h245- relay sends DTMF tones across the signaling channel, instead of
alphanumeric as part of the voice stream. DTMF relay is needed when you are
using a low-bit-rate coder-decoder (codec) for voice
compression, because the potential exists for DTMF signal loss
or distortion.
Example 9-1 shows an H.323 gateway that has been configured for Cisco CallManager
redundancy.
Example 9-1 H.323 Gateway Configuration

!Creates VoIP dial-p eer 101
H323_GW(config)#!
dial-pe er voice 101 voip
H323_GW(config)#d
!Sends D TMF digits o ut-of-band
H323_GW(config-dial-peer)#!
dtmf-relay h2 45-alphanume ric
H323_GW(config-dial-peer)#d
!Matches all 4-digit e xtensions
destinatio n-pattern ... .
!Tar get address of Primary Ca llManager
session tar get ipv4:10.1 .1.101
H323_GW(config-dial-peer)#s
!Sel ects this as the preferre d dial-peer
preferen ce 0
H323_GW(config-dial-peer)#p
!Crea tes VoIP dia l-peer 102
H323_GW(config)#!
dial -peer voice 102 voip
H323_GW(config)#d
dtmf -relay h245-a lphanumeric
d estination-pa ttern ....
!Target address of S econdary Cal lManager
session targ et ipv4:10.1 .1.102
H323_GW(config-dial-peer)#s
!Sel ects this as the secondar y dial-peer
preferenc e 1
H323_GW(config-dial-peer)#p
Call Classification
The Call Classification feature, introduced in Cisco CallManager Release 4.1, provides the ability
to configure gateways and trunks as on the network (OnNet) or off the network (OffNet) at the
device or at the global level. As a result, a call through these devices is classified as either OnNet
or OffNet. With these classifications, Cisco CallManager provides the ability to restrict OnNet-to-
OffNet transfers and to drop an ad hoc conference when no OnNet parties remain in the
conference. This field provides an OnNet or OffNet alerting tone when the call is classified as
OnNet or OffNet, respectively.
The corresponding Call Classification field in the gateway and trunk configuration windows,
shown in Figure 9-2, marks the corresponding devices as OnNet or OffNet or Use System Default.
The default Call Classification settings are OnNet for all IP Phones and intercluster trunks and
Use System Default (which is OffNet) for other gateways.
Figure 9-2 CallManager Call Classification Gateway Configuration
In Cisco CallManager 4.X releases, Cisco defined a new, global service parameter called Call
Classification. The default value of this parameter is OffNet. When Use System Default is
selected at the device level for a particular gateway or trunk, the value of the service parameter
(defined using the Service > Service Parameters > Cisco CallManager menu option in
CCMAdmin) is used to judge the device as OnNet or OffNet. Therefore, gateways are classified
as external (OffNet) by default.
FXS ports and IP Phones are not configurable and are always treated as OnNet.
The parameters that determine whether a transfer is allowed or restricted are as follows:
■ The Call Classification setting on the gateway or trunk.
■ The Call Classification setting and Allow Device Override check box setting on the route
pattern. If the Allow Device Override check box is checked, the Call Classification setting
on the gateway or trunk takes precedence.
■ The Service parameter Block OffNet to OffNet Transfer. When this parameter is set to False,
the transfer is not restricted. When this parameter is set to True (the default) and the transfer
is initiated between two OffNet parties, the transfer is blocked.
NOTE The ability to restrict external transfers and to drop an ad hoc conference when no
OnNet parties remain on the call help to prevent toll fraud. Preventing toll fraud and many other
IP telephony security considerations are covered in detail in Chapter 22, “Preventing Toll
Fraud.”
MGCP Gateway Configuration

Complete these steps to add an MGCP gateway to the CallManager configuration:
Step 2 Click the Add a New Gateway link and choose one of the various MGCP-
capable devices from the Gateway Type menu.
NOTE Selecting a Cisco router by the actual model of router from the Gateway Type drop-
down menu automatically chooses the MGCP protocol for communication.
Step 3 Cisco CallManager automatically populates the Device Protocol field with
the MGCP protocol. Click Next.
Step 4 For the Domain Name field, shown in Figure 9-3, enter the unique
hostname (or fully qualified domain name [FQDN]) of the Cisco device
that will be acting as the gateway.
Figure 9-3 CallManager MGCP Gateway Configuration
Step 5 Assign the gateway to a Cisco CallManager group for redundancy.

Step 6 Choose the type of network modules that are used in the MGCP gateway.
Step 7 Choose the type of voice interface cards (VICs) that are used in the subunit
slots of the MGCP gateway.
Step 8 Click Insert.
Step 9 Click an endpoint identifier (for example, 1/0/0) to configure device
protocol information and add ports for the installed types of VICs.
Table 9-2 lists the IOS commands to help configure endpoint identifiers for analog devices
connected to the MGCP gateway.
Table 9-2 Cisco IOS Commands to Configure an MGCP Voice Gateway
Command Description
Router(config)# hostname Assigns a unique name to the voice gateway so that the Cisco
MGCP_GW CallManager server can identify it. This name must be unique
throughout the network.
MGCP_GW(config)# mgcp Enables MGCP on the voice gateway.
MGCP_GW(config)# mgcp Identifies the primary Cisco CallManager for the gateway.
call-agent ip-address
MGCP_GW(config)# ccm- Indicates to the gateway that the Cisco CallManager is using
manager mgcp MGCP.
MGCP_GW(config)# ccm- Specifies the secondary and tertiary Cisco CallManager servers
manager redundant-host ip- that are used for Cisco CallManager redundancy.
address1 ip-address2
MGCP_GW(config)# ccm- Specifies how the gateway behaves if the primary server
manager switchback {graceful becomes unavailable and later becomes available again. The
| immediate | schedule-time keywords and arguments are as follows:
hh:mm | uptime-delay
minutes} • graceful—Completes all outstanding calls before returning
the gateway to the control of the primary Cisco CallManager
server.
• immediate—Returns the gateway to the control of the
primary Cisco CallManager server without delay, as soon as
the network connection to the server is reestablished.
• schedule-time hh:mm—Returns the gateway to the control
of the primary Cisco CallManager server at the specified
time, where hh:mm is the time according to a 24-hour clock.
If the gateway reestablishes a network connection to the
primary server after the configured time, the switchback
occurs at the specified time on the following day.
• uptime-delay minutes—Returns the gateway to the control
of the primary Cisco CallManager server when the primary
server runs for a specified number of minutes after a network
connection is reestablished to the primary server. Valid
values are from 1 to 1440 (from 1 minute to 24 hours).
MGCP_GW(config)# mgcp Configures the gateway to use out-of-band DTMF relay for all
dtmf-relay voip codec all mode codecs. If this command is not configured, DTMF tones will
out-of-band not be regenerated correctly on remote endpoints.
MGCP_GW(config)# dial-peer Creates a POTS dial peer.
voice tag pots
continues
Table 9-2 Cisco IOS Commands to Configure an MGCP Voice Gateway (Continued)
Command Description
MGCP_GW(config-dial-peer)# Configures the dial peer to use the MGCP application. The
application MGCPAPP MGCPAPP option is case sensitive in some Cisco IOS
software releases. Unless you know that your version is not
case sensitive, always issue this command in uppercase. You
can check whether your version is case sensitive after you
configure this command by looking at the output of the show
running-config command.
MGCP_GW(config-if)# isdn Places a PRI or BRI ISDN interface under CallManager control
bind-l3 ccm-manager (known as backhauling the interface). This allows CallManager
to control all L3 (network layer) communication of the PRI/
BRI line. This command is entered under the interface
configuration mode for the PRI/BRI connection.
Example 9-2 shows an MGCP gateway that has been configured for Cisco CallManager
redundancy placing one FXS port under CallManager control.
Example 9-2 MGCP Gateway Configuration Example

!The hostname li nks the gate way in the Ca llManager co nfiguration
Router(config)#!
hostnam e MGCP_GW
Router(config)#h
!enables the protoco l
Router(config)#!
mgc p
MGCP_GW(config)#m
!Iden tifies the IP address of the primar y CallManager
Router(config)#!
mgc p call-agen t 10.1.1.1
MGCP_GW(config)#m
!Us es MGCP to communicate with the Cal lManager
Router(config)#!
cc m-manager m gcp
MGCP_GW(config)#c
!I dentifies s econdary Cal lManager
Router(config)#!
ccm-ma nager redu ndant-host 10 .1.1.2
MGCP_GW(config)#c
!All DTMF digi ts sent out- of-band
Router(config)#!
mgcp dt mf-relay vo ip codec all mode out-of- band
MGCP_GW(config)#m
!Creates POTS dial -peer 101
Router(config)#!
d ial-peer v oice 101 pot s
MGCP_GW(config)#d
!Selects the physical port the Cal lManager wil l control
Router(config-dial-peer)#!
port 1 /0/0
MGCP_GW(config-dial-peer)#p
!Plac es the port u nder CallMan ager control
MGCP_GW(config-dial-peer)#!
applic ation MGCPAP P
MGCP_GW(config-dial-peer)#a
Non-IOS MGCP Gateway Configuration

You can equip Cisco Catalyst 6500 series switches with a variety of modules, allowing them to act
as a voice gateway. If the Catalyst 6500 is running the NativeIOS software image, you can perform
the configuration as you would a typical Cisco gateway. However, the Cisco Catalyst 6500 can also
function using the CatalystOS operating system. If you are operating in this type of environment,
you will need to configure the CallManager to communicate with the 6500 as a Non-IOS MGCP
Gateway. Complete the following steps to add a non-IOS MGCP gateway to Cisco CallManager:
Step 2 Click the Add a New Gateway link and choose one of the various non-IOS
MGCP devices from the Gateway Type menu. If you have selected the
Cisco Catalyst 6000 T1 or E1 VoIP Gateway module (WS-X6608), the
Device Protocol field provides you with the option of specifying either
digital access PRI or digital access T1. After you have made your selection,
click Next.
Step 3 Add the MAC address to the MAC Address field.
Cisco CallManager associates with a non-IOS MGCP gateway (such as the
Cisco Catalyst 6608 T1 or E1 blade) through the MAC address of the port.
The show port module command from enable mode on the Cisco Catalyst
6000 is a quick way to identify and list the MAC addresses of each digital
gateway port on the Voice T1/E1 and Services (WS-X6608) module:
show port 3
Cat6000(enable)s
Port DHCP MAC-Address IP-Address Subnet-Mask
-------- ------- ----------------- --------------- ---------------
3/1 disable 00-30-b6-3e-8e-c4 172.16.10.121 255.255.255.0
3/2 disable 00-30-b6-3e-8e-c5 172.16.20.122 255.255.255.0
3/3 disable 00-30-b6-3e-8e-c6 172.16.30.123 255.255.255.0
3/4 disable 00-30-b6-3e-8e-c7 172.16.40.124 255.255.255.0
3/5 disable 00-30-b6-3e-8e-c8 172.16.1.125 255.255.255.0
3/6 disable 00-30-b6-3e-8e-c9 172.16.1.126 255.255.255.0
3/7 disable 00-30-b6-3e-8e-ca 172.16.1.127 255.255.255.0
3/8 disable 00-30-b6-3e-8e-cb 172.16.1.128 255.255.255.0
To display detailed information about a specific port on the module, use the
show port mod/port command.
Step 4 Assign the gateway to a device pool.
Step 5 Configure additional parameters as desired, and click the Insert button
when finished.
Step 6 Repeat Steps 1 through 5 for each T1 or E1 port that you want to add as a
gateway in Cisco CallManager administration.
After you add the gateway to the database, Cisco CallManager creates a configuration file in the
cluster on the Cisco TFTP server, and this file is where the T1 or E1 port downloads its
configuration details, which include an ordered list of Cisco CallManager servers.
On the gateway itself (Cisco Catalyst 6500 T1 or E1 port), it is recommended that you statically
configure the IP address, subnet mask, default gateway, and TFTP server address of each T1 and
E1 port that is used as a digital gateway. The TFTP server address should be the IP address of the
Cisco CallManager to which you want the port to register and download its configuration file. The
following command statically configures the following voice port settings: IP address, subnet
mask, VLAN ID, TFTP server IP address, and default gateway:
Cat6000 (enable) set port voice in 3/1 dhcp disable 172.16.1.121 255.255.255.0
vla n 1 tftp 17 2.16.1.5 ga teway 172.16 .1.1
NOTE When a port resets, the module has the ability to reset the adjoining port because all
eight ports on the WS-X6608 module share the same XA processor. This reset process creates
a domino effect, and all of the ports on the module reset, creating the following error message:
%SYS-4-MODHPRESET:Host process (860) mod_num/port_num got reset asynchronously
If you are not going to use a port, you should either disable the port or configure and register it
to Cisco so that it does not continually perform an asynchronous reset.
When you have successfully placed an individual 6608 T1 port under CallManager MGCP
control, you will see the following output:
AV-6509-1 (enable) show port 2/1
Port Name Status Vlan Duplex Speed Type

----- ------------------ ---------- ---------- ------ ----- ------------
2/1 notconnect 1 full 1.544 T1
Port DHCP MAC-Address IP-Address Subnet-Mask

-------- ------- ----------------- --------------- ---------------
2/1 disable 00-30-b6-3e-8e-c4 172.16.1.121 255.255.255.0
Port Call-Manager(s) DHCP-Server TFTP-Server Gateway

-------- ----------------- --------------- --------------- --------------
2/1 172.16.1.5 - 172.16.1.5 172.16.1.1
Port DNS-Server(s) Domain

-------- ----------------- ----------------------------------------------
2/1 - -
Port CallManagerState DSP-Type

-------- ---------------- --------
2/1 registered C549
Cisco Trunk Concepts

In addition to adding gateways to the CallManager configuration, trunks can also provide
connectivity to outside devices. Trunks are seen by Cisco CallManager as logical links to other
networks. These links have the ability to determine the location of an endpoint, but do not carry
voice traffic. This underscores the major difference between trunks and gateways: Gateways can
typically locate (or represent) an endpoint and carry the voice traffic to that endpoint; trunks only
Cisco Trunk Concepts 175
locate an endpoint. For example, you could create an intercluster trunk from your CallManager
cluster containing 3XXX extensions to another CallManager cluster containing 4XXX extensions.
When a user in your cluster (extension 3505) dials an extension in the other cluster (extension
4505), the local CallManager signals over the trunk to the remote CallManager. This signaling
occurs to locate the IP address of extension 4505. After this IP address has been found, the audio
path opens directly between the local extension 3505 and the remote extension 4505.
Your choices for configuring trunks in Cisco CallManager depend on whether the IP WAN uses
gatekeepers to handle call routing and on the types of call-control protocols that are used in the
call-processing environment.
Cisco CallManager Administration supports the following trunk types:
■ H.225 trunk gatekeeper-controlled—Use an H.225 gatekeeper-controlled trunk for toll

bypass or for integration with an existing H.323 environment. The H.225 gatekeeper-
controlled trunk enables Cisco CallManager to communicate with Cisco CallManager
clusters and other H.323 devices registered to the H.323 gatekeeper. The H.225 gatekeeper-
controlled trunk is not recommended in a pure Cisco CallManager environment, but it is
required in a mixed environment with Cisco CallManager and Cisco CallManager Express or
another third-party H.323 gateway. The H.225 trunk attempts to discover the other H.323
device on a call-by-call basis. If it discovers a device that understands intercluster trunk
protocol, it automatically uses that protocol. If it cannot discover the other device, Cisco
CallManager uses the standard H.225 protocol. To use this method, choose Device > Trunk
and choose H.225 Trunk (Gatekeeper Controlled).
■ Intercluster trunk gatekeeper-controlled—The intercluster gatekeeper-controlled trunk

enables Cisco CallManager to communicate with other Cisco CallManager clusters registered
to an H.323 gatekeeper. It is recommended that you use the intercluster gatekeeper-controlled
trunk only in deployments based entirely on Cisco CallManager. To use this method, choose
Device > Trunk and choose Inter-Cluster Trunk (GateKeeper Controlled) in Cisco
CallManager Administration.
TIP Because the intercluster trunk format is Cisco proprietary, Cisco recommends using
H.225 gatekeeper trunks wherever possible. The gatekeeper-controlled intercluster trunk option
remains for backward compatibility with early CallManager versions.
■ Intercluster trunk nongatekeeper-controlled—In a distributed network that has no

gatekeeper control, you must configure a separate intercluster trunk for each remote cluster
that the local Cisco CallManager can call over the IP WAN. The intercluster trunks statically
specify the IP addresses or hostnames of the remote devices. To use this method, choose
Device > Trunk and choose Inter-Cluster Trunk (Non-GateKeeper Controlled) in Cisco
NOTE Intercluster trunks (nongatekeeper-controlled) between Cisco CallManager clusters

are unidirectional. You must configure trunks to and from the cluster you are connecting for the
trunks to operate.
■ SIP trunk—Cisco CallManager Release 4.x supports a Session Initiation Protocol (SIP)
trunk for interworking with a SIP network or gateways, but it currently does not allow SIP IP
Phones to register directly with Cisco CallManager. To use this method, choose Device >
Trunk and choose SIP Trunk in Cisco CallManager Administration.
Configuring Intercluster Trunks

Although the SQL 2000 database shared between CallManagers in your cluster provides a listing
of all extensions that a user can dial within the cluster, you must manually add connections and
route patterns to remote clusters before they can be reached. The connection type that acts as the
road between CallManager clusters is the intercluster trunk. Depending on the size of your IP
telephony network, you can create trunks directly between CallManager clusters or you can create
trunks to connect to an H.323 gatekeeper.
NOTE You can create a network of enormous size using only intercluster trunks; however,
because the trunks are a full-mesh configuration, it can become increasingly difficult to manage
the larger your network gets. Intercluster trunks also lack the ability to perform call admission
control, which could result in the WAN becoming swamped with VoIP traffic.
A gatekeeper provides call admission control by using the H.225 Registration, Admission, and
Status (RAS) protocol message set that is used as a central point for call admission control,
bandwidth allocation, and dial pattern resolution (call routing). The gatekeeper provides these
services for communications between Cisco CallManager clusters and H.323 networks. Note that
the voice path is between the endpoints; no RTP audio travels through the gatekeeper.
You can configure gatekeepers and trunks in Cisco CallManager Administration to function in
either of the following ways:
■ Nongatekeeper-controlled trunks—In this case, you explicitly configure a separate

Intercluster Trunk for each remote device cluster that the local Cisco CallManager can call
over the IP WAN, as shown in Figure 9-4. You also configure the necessary dial plan details
to route calls to and from the various Intercluster Trunks. The Intercluster Trunks statically
Configuring Intercluster Trunks 177
specify the IP addresses of the remote devices. To use this method, choose Device > Trunk
and then choose Inter-Cluster Trunk (Non-Gatekeeper Controlled) in Cisco CallManager
Administration.
Figure 9-4 Nongatekeeper-Controlled Trunk Design
Cluster Cluster
1 2
IP WAN
Cluster Cluster
3 4
■ Gatekeeper-controlled trunks—In this case, a single intercluster trunk from each

CallManager cluster suffices for communicating with all remote clusters, as shown in Figure
9-5. Similarly, you need only a single H.225 trunk to communicate with multiple H.323
gatekeeper-controlled endpoints. In this configuration, the gatekeeper can dynamically
determine the appropriate IP address for the destination of each call to a remote device, and
the local Cisco CallManager uses that IP address to complete the call.
Figure 9-5 Gatekeeper-Controlled Trunk Design
Cluster Cluster
1 2
Gatekeeper
Cluster Cluster
3 4
This configuration works well in large and smaller systems. For large systems in which
many clusters exist, this configuration helps to avoid the configuration of individual
Intercluster Trunks between each cluster. To use this method, choose Device > Trunk and
choose Inter-Cluster Trunk (Gatekeeper Controlled) in Cisco CallManager
Administration.
If you configure gatekeeper-controlled trunks, Cisco CallManager automatically creates a

virtual trunk device. The IP address of this device changes dynamically to reflect the IP
address of the remote device as determined by the gatekeeper.
Complete these steps to configure a nongatekeeper-controlled intercluster trunk:
Step 1 Choose Trunk from the Device menu in the Cisco CallManager
Step 2 Click the Add a New Trunk link and choose Inter-Cluster Trunk (Non-
Gatekeeper Controlled) from the Trunk Type drop-down menu.
Step 3 Cisco CallManager populates the Device Protocol field with the
appropriate protocol. Click Next to continue.
Step 4 The Trunk Configuration window appears, as shown in Figure 9-6. Add the
device name. It does not have to be the IP address, but it must be unique
throughout the cluster.
Figure 9-6 CallManager Nongatekeeper-Controlled intercluster trunk Configuration

Step 5 At the bottom of the Trunk Configuration window, you can add the IP
addresses of up to three Cisco CallManager servers in the remote cluster.
You must add the IP address for at least one remote CallManager.
NOTE Configuring gatekeeper-controlled intercluster trunks is covered in Chapter 14,

“Implementing Multiple-Site Deployments.”
SIP and Cisco CallManager

SIP is specified in Internet Engineering Task Force (IETF) RFC 2543, published in 1999. Since
the initial publication, nine bis versions (or revisions) have been published, and in 2002, RFC
2543-bis-09 was renumbered to RFC 3261.
SIP is an ASCII-based, application-layer control protocol that can be used to establish, maintain,
and terminate multimedia sessions, such as Internet telephony calls between two or more
endpoints. SIP works in client/server relationships and in peer-to-peer relationships.
SIP is a VoIP signaling protocol (as are SCCP, MGCP, and H.323) with the following capabilities:
■ Determines the availability of the target endpoint—If a call cannot be completed because
the target endpoint is unavailable, SIP determines whether the called party is connected to a
call already or did not answer in the allotted number of rings. SIP then returns a message
indicating why the target endpoint was unavailable.
■ Establishes a session between the originating and target endpoints—If the call can be
completed, SIP establishes a session between the endpoints. SIP also supports midcall
changes, such as the addition of another endpoint to the conference or the changing of a media
characteristic or codec.
■ Handles the transfer—SIP supports the transfer of calls from one endpoint to another.
During a call transfer, SIP establishes a session between the transferee and a new endpoint
(specified by the transferring party) and terminates the session between the transferee and the
transferring party.
■ Terminates call—At the end of a call, SIP terminates the sessions among all parties.
SIP Components
A SIP network in a Cisco CallManager environment uses the following components, as shown in
Figure 9-7:
■ Cisco SIP proxy server—The proxy server works as an intermediate device that receives SIP
requests from a client and then forwards the requests on behalf of the client. Proxy servers can
provide functions such as authentication, authorization, network access control, routing,
reliable request retransmission, and security.
■ Redirect server—The redirect server provides the client with information about the next hop
or hops that a message should take, and the client then contacts the next-hop server or user
agent server directly.
■ Registrar server—The registrar server processes requests from user agent clients (UACs) for
registration of their current location. Redirect or proxy servers often contain registrar servers.
■ User agent—A combination of UAC and user agent server (UAS) that initiates and receives
calls. A UAC initiates a SIP request. A UAS is a server application that contacts the user when
it receives a SIP request. The UAS then returns a response (such as ring, busy, connected, or
no answer) on behalf of the user. Cisco CallManager can act as both a server or client (a back-
to-back user agent) and can initiate and accept calls simultaneously.
Figure 9-7 SIP Components
Location SIP Servers and Services

Registrar Redirect Database
SIP Proxy
IP
SIP User
Agents SIP User
Agents
SIP-GW
NOTE In the Cisco implementation of the Cisco SIP proxy server, all of the functions of the
Redirect and Registrar servers are implemented in a single box. The user agents are shown as
any end-user device capable of supporting the SIP protocol.
SIP uses a request-and-response method to establish communications between various

components in the network and to ultimately establish a call or session between two or more
endpoints. A single session might involve several clients and servers.
Identification of users in a SIP network is by a SIP URL that takes the form sip:user@host. The
user part is a username, telephone number, or E.164 address. The host part is either a domain name
or a numeric network address.
TIP Cisco CallManager 5.X is an appliance-based version of CallManager that implements

SIP capabilities to the end-user device using proprietary SIP add-ons to compensate for the
current IETF SIP protocol shortcomings.
CallManager SIP Integration and Configuration

Cisco CallManager Release 4.0 introduced a native SIP signaling interface in Cisco CallManager
referred to as a SIP trunk. As shown in Figure 9-8, the SIP trunk enables interoperability between
Cisco CallManager networks and SIP networks served by a SIP proxy server, and allows any
existing devices controlled by Cisco CallManager to communicate with SIP networks. Adding SIP
trunks to the CallManager configuration gives CallManager the ability to communicate with voice
devices using any of the standards-based VoIP signaling protocols available today.
Figure 9-8 Cisco CallManager SIP Trunk

CallManager
Cluster Cisco SIP
Proxy Server
V
Conf
Xcode
IP
VMail
Apps
IP
Microsoft Cisco SIP Cisco IOS
SCCP Messenger IP Phone SIP Gateway
Phones IP
SoftPhones SCCP CTI

MGCP SIP
Follow these steps to add a SIP trunk in Cisco CallManager Administration:
Step 1 Choose Device > Trunk > SIP Trunk in Cisco CallManager
Administration to display the Trunk Configuration window.
Step 2 Click Add a New Trunk and choose SIP Trunk from the Trunk Type
drop-down menu. The Device Protocol field is automatically filled in.
Step 3 Assign the SIP trunk to the device pools, location, and automated alternate
routing (AAR) group as appropriate (example shown in Figure 9-9). The
system checks the Media Termination Point Required check box by
default, and you cannot uncheck it.
Figure 9-9 Cisco CallManager SIP Trunk Configuration
Step 4 The Destination Address field indicates the IP address, fully qualified
domain name (FQDN), or Domain Name System (DNS) server address of
the proxy server. It applies to outgoing calls only; incoming calls do not use
the destination address.
Step 5 The Destination Port field indicates the port through which to send SIP
traffic to the proxy server. The default specifies port 5060, which can be
changed to any unique value from 1024 to 65,535.
Step 6 The Incoming Port field is the port that Cisco CallManager listens to for
incoming SIP traffic. The default specifies port 5060, which can be changed
to any unique value from 1024 to 65,535.
NOTE The Destination and Incoming SIP ports act as a positive/negative configuration
between the communicating devices. If the destination SIP port is set to 5061, the incoming port
on the receiving device must be 5061.
Step 7 Click Insert when finished.
Summary
This chapter covered the addition of voice gateways and trunks to the Cisco CallManager
configuration. Voice gateways have the ability to connect your Cisco CallManager cluster to
networks outside of the local VoIP system. These networks could include the PSTN or other,
remote VoIP networks managed by Cisco CallManager or some other third-party vendor.
Cisco CallManager supports direct communication with H.323, MGCP, and SCCP gateways. To
support voice networks, gateways should run a software version capable of supporting core voice
requirements: DTMF relay, supplementary services, server redundancy, and call survivability.
These voice gateways can host analog and digital trunks, allowing them to connect to traditional
voice networks. Analog trunks allow a single voice path per port, whereas digital trunks allow a
single port to support many voice paths.
In addition to supporting voice gateway configurations, Cisco CallManager also allows you to add
trunking mechanisms between voice networks. Trunks perform very similar functions to voice
gateways by providing logical paths to other voice networks. However, trunks do not carry voice
data (RTP streams). Instead, they provide connection information for devices to contact each other
directly, establishing RTP streams without the need for intermediary devices. New to CallManager
in versions 4.x is the ability to support SIP trunks, which allows the Cisco CallManager to connect
to many third-party voice systems.
Review Questions
1. Which of the following represent an interface you would find on an analog gateway?
(Choose two.)
a. FXS
b. E&L
c. T1
d. FXO
e. E1
2. Which of the following is not a core voice gateway requirement? (Choose two.)
a. DTMF Relay
b. inline power support
c. supplementary service support
d. SIP trunk support
e. call survivability
3. Which of the following devices use Skinny signaling? (Choose two.)
a. Cisco Catalyst 6500 (CatalystOS)
b. VG224
d. Cisco 2801 with a VIC-2FXS interface
4. Which of the following commands could you use to assign all four-digit extensions starting
with the number 3 (for example, 3152, 3521, and so on) to a port on an H.323 gateway?
a. destination-pattern 3
b. destination-pattern 3 ext 3
c. destination-pattern 3 ext 4
d. destination-pattern 3...
e. destination-pattern 3xxx
5. CallManager 4.1 introduced a new Call Classification feature when configuring gateways.
What does this feature accomplish?
a. It provides advanced logging functions, categorizing the higher importance calls near
the top of the log.
b. It allows or denies calls based on the class of user making the call.
c. It allows you to restrict conference calls or transfers between local devices and devices
off the network.
d. It allows you to isolate log files for each gateway in the CallManager configuration.
6. You are viewing the configuration of an MGCP router and come across the line ccm-manager
redundant-host 10.1.1.5 10.1.1.6. What does the IP address 10.1.1.5 represent?
a. the IP address of the primary Cisco CallManager
b. the IP address of the secondary Cisco CallManager
c. the IP address of the tertiary Cisco CallManager
d. the IP address of the SRST Failover Cisco CallManager
7. What protocol(s) could you use to communicate with a Cisco Catalyst 6500 switch running
the NativeIOS image? (Select all that apply.)
a. MGCP
b. H.323
c. Non-IOS MGCP
d. Skinny
8. What is the difference between configuring a gateway and a trunk in the Cisco CallManager?
a. Gateways have more security capabilities.
b. Gateways carry voice traffic; trunks do not.
c. Trunks allow more flexibility with your connections.
d. Gateways can only connect to Cisco equipment; trunks can connect to third-party
equipment.
9. What is the advantage of using an H.323 gatekeeper in a Cisco IP telephony environment?
a. Gatekeepers provide more redundancy for your connections.
b. Gatekeepers ease configuration of Intercluster Trunk connections.
c. Gatekeepers provide a more centralized security scheme for the voice network.
d. all of the above
10. What protocol does the H.323 gatekeeper use to provide call admission control?
a. H.245
b. H.225
c. H.225 RAS
d. SIP
following topics:
■ Identifying the necessary elements for

creating external route plans in Cisco
CallManager, including gateways and trunks,
route groups, route lists, and route patterns
■ Explaining how route groups are used to

prioritize and group gateways to send
external calls out of a preferred path and keep
a backup path for redundancy
■ Explaining how route lists are used to

prioritize route groups that contain different
types of gateways that connect to the IP WAN
or PSTN
■ Explaining how to create route patterns that

use wildcards to route calls to gateways or
route lists
■ Describing how Cisco CallManager analyzes

dialed digits to determine where to route calls
■ Describing how route groups, route lists, and

route patterns are used in a basic route plan to
route calls to the IP WAN or PSTN
CHAPTER 10
Route Plan Basics
To determine where to route calls, Cisco CallManager requires knowledge of the patterns of
digits that users dial to reach a particular telephone number. These patterns include access
codes, area codes, and combinations of the digits that are dialed. The route plan is the set of
configurable lists that provide Cisco CallManager with the knowledge of where to route calls.
Related to call routing is call distribution, which, simply put, is the ability to specify the order
and other factors in which Cisco CallManager extends calls to members of a group.
This chapter discusses the basic components of a route plan within Cisco CallManager. Basic
components of a route plan include route groups, route lists, and route patterns.
External Call Routing

When you place a call from a Cisco IP Phone, Cisco CallManager analyzes the dialed digits. If
the dialed number matches a directory number (DN) that is registered with the Cisco
CallManager cluster, Cisco CallManager routes the call to the destination Cisco IP Phone that
is associated with the matching DN. This type of call is an internal (or on-cluster) call. Cisco
CallManager handles the call internally without the need to route the call to an external PSTN
gateway.
IP Phones are not the only devices that can place and receive internal calls; any device that
registers a DN with Cisco CallManager can place and receive internal calls. Examples of other
devices include the Cisco IP SoftPhone/IP Communicator and analog telephones that are
attached to Media Controller Gateway Protocol (MGCP) or Skinny Client Control Protocol
(SCCP, or Skinny)–based gateways.
When a Cisco IP Phone dials a number that does not match a registered DN, it assumes that the
call is an external (or off-cluster) call. Cisco CallManager then searches its external route table
to determine where to route the call. Cisco CallManager uses the concept of route pattern and
translation pattern tables to determine where and how to route an external call. The route pattern
and translation pattern tables are very similar to the routing table that a Cisco router maintains
for routing data.
188 Chapter 10: Cisco Unified CallManager Route Plan Basics
You can create external route plans based on a three-tiered architecture that allows multiple layers
of call routing redundancy as well as digit manipulation. Route patterns match external dial
strings, in which a corresponding route list will select available paths for the outbound call based
on priority. Cisco refers to these paths as route groups, which are very similar to the trunk group
concept in traditional PBX terminology. You can think of a route pattern as a static route with
multiple paths that you can prioritize. Figure 10-1 depicts the three-tiered route plan architecture.
Figure 10-1 Cisco CallManager Route Plan Architecture
Route pattern:
• Matches dialed number for external calls Route
• Performs digit manipulation (optional) Pattern
• Points to a route list for routing
Route list:
Route
• Chooses path for call routing List
• Points to prioritized route groups
First Second
Configuration Order
Choice Choice
Route group:
Route Route
• Performs digit manipulation Group Group
• Points to the actual devices
First Second
Choice Choice
GK V V
Devices: IP WAN PSTN

• Gateways (H.323, MGCP)
• Gatekeeper
V
• ICT
(remote Cisco CallManager)
In addition to facilitating multiple prioritized paths for a given dialed number, the route plan can
also provide unique digit manipulation for each path, based on the external network requirements.
Digit manipulation involves adding or subtracting digits from the original dialed number to
accommodate user dial habits and to ensure that the external network or PSTN receives the correct
digits to place a call.
Even though the CallManager processes these route plans from the top down (a user dials the route
pattern, which directs the call to a route list, then to the preferred route group, and finally to the
device), the configuration of the route plan occurs from the bottom up (devices are added, route
groups are created from the devices, route lists are created from the route groups, and the pattern
points to the list).
Route Plan Configuration Process 189
Route Plan Configuration Process

The following is the general process for route plan configuration. You can construct a route plan
using this process:
Step 1 Add gateway devices—Create gateway devices using the Device menu.
Step 2 Build route groups from available devices—Select and place gateway
devices in an ordered list to build a route group.
Step 3 Build route lists from available route groups—Select and order route
groups into a route list.
Step 4 Build route pattern—Build a route pattern and associate it with an
available route list or gateway device.
NOTE Chapter 9, “Configuring Cisco Gateways and Trunks,” explained the process of adding
gateway devices and trunks to the Cisco CallManager configuration. This chapter discusses the
route plan configuration process shown in Steps 2 to 4.
The route pattern is the key component in a route plan. The route pattern matches an external dial
string and routes the outgoing call to the appropriate gateway. When the dialed digits match a route
pattern, Cisco CallManager routes the call to the assigned route list or gateway.
Route Group Overview and Configuration

Route groups and route lists work together to control and enhance external call routing. They also
help with implementing cost savings and redundancy, which are among the features of a Cisco IP
telephony network.
Route groups are a logical grouping of device gateways, as shown in Figure 10-2. Prioritizing
these device gateways allows you to send external calls out of a preferred gateway (possibly across
the IP WAN for toll savings) and keep a backup path for external calls (usually the public switched
telephone network [PSTN]) if the primary gateway is down or unable to route the call.
Figure 10-2 Cisco CallManager Route Group Design
IP
User DIALS EXTERNAL NUMBER
723-8912
Route
Pattern
723-xxxx
Digital GW 1
First
Choice
Route List Route Group PSTN
Second
Choice
Digital GW 2
You might encounter a scenario that requires multiple route groups, such as multiple long-distance
carriers. Each long-distance carrier offers different rates for long-distance calls on its network. You
can use route groups to prioritize the use of the cheaper carrier over the others and retain
redundancy if the cheaper carrier cannot route the call for some reason.
Use these steps to configure a route group to the Cisco CallManager configuration:
Step 1 Choose Route Group from the Route Plan > Route/Hunt menu in the
Cisco CallManager Administration window.
Step 2 Click the Add a New Route Group link. Give the new route group a name
and click Continue. It is always best to follow good naming nomenclature
(such as PSTN_RG or WAN_RG).
Step 3 Choose a gateway from the Available Devices menu to add to the route
group. Figure 10-3 demonstrates adding an H.323 PSTN gateway
(10.1.5.1) to the PSTN Connection route group for an organization. The
route group is a prioritized list of gateways. Place your highest priority
gateway at the top of the list using the up and down arrows to the right of
the Selected Devices box.
Figure 10-3 Cisco CallManager Route Group Configuration
Step 4 Click Add or Update when finished.

The method that you use to configure route groups depends on the gateway types that you plan to
include in the group. Each gateway is an entity; because the CallManager does not know the
individual ports on an H.323 gateway, you must add H.323 gateways to a route group as a whole
by device. However, because the CallManager knows everything about an MGCP gateway, you
can group them by ports, which means that individual ports on MGCP gateways can be entities in
a route group.
NOTE The device entities, whether devices in the case of H.323 gateways or ports in the case
of MGCP gateways, can be used only once in your route plan configuration. This means that you
can only assign them to a single route group. On the other hand, you can assign route groups to
multiple route lists. This key fact will play heavily in your route plan design.
In addition, the route group allows you to select the distribution algorithm you want to use. There
are two supported distribution algorithms:
■ Circular (default)—Enables a type of round-robin load balancing for the route group. The
first call will go to the first device, the second call to the second, and so on. This is the ideal
setting for load distribution across your gateways.
■ Top down—Uses a strict primary, failover model. The first device listed in the route group is
always used unless the gateway is at capacity or unreachable, at which point the second device
is used.
You should configure route groups by function. For example, all gateways to the PSTN can belong
to one route group, all gateways to long-distance carriers can belong to another (with the cheapest
carrier having priority), and all gateways across the IP WAN can belong to another route group. If
you want to control routing by gateway instead of by group of gateways or ports, you can set up
route groups so that they can contain only one gateway.
Route List Overview and Configuration

Route lists consist of an ordered list of route groups, as shown in Figure 10-4. Route lists expand
the route group concept and allow you to prioritize your route groups. Although a gateway or
group of ports on a gateway can belong to only a single route group, route groups can belong to
any number of route lists. Route groups give you more control over external call routing. In the
route plan shown in Figure 10-4, the company has connections to three IntereXchange Carriers
(IXCs). They have placed the endpoints into route groups and then prioritized the route groups in
a route list to provide least-cost-routing for toll calls.
Figure 10-4 Cisco CallManager Route List Design
IP
User DIALS NUMBER
Analog IXC
Route Pattern First Choice Gateway 1 1
Ports 1–8
Route Group 1
First
Choice Second Choice
Ports 1–4
Analog IXC
Route List
Gateway 2 2
First Choice
Second Ports 5–8
Choice
Route Group 2
Second Choice
Ports 1–8 Analog IXC
Gateway 3 3
With route lists, you can implement features such as toll bypass and PSTN fallback because within
the route list you can prioritize route groups that contain different types of gateways (IP WAN,
PSTN, and so on).
Digit manipulation is the key to making toll bypass and PSTN fallback features transparent to your
users. Digit manipulation occurs in the form of calling-party and called-party transformations.
Use calling-party transformations to manipulate caller ID information that is presented to the

called party. Use called-party transformations to actually manipulate the digits that are dialed. You
can apply calling- and called-party transformations at five different levels of the call-routing
process: at the originating device, as part of a translation pattern, as part of a route pattern, as part
of a route list, or at the terminating device. Calling- and called-party transformations that are set
at the route list level override transformations that are set at any other level.
NOTE Calling- and called-party transformations are covered in detail in Chapter 11, “Cisco
Unified CallManager Advanced Route Plans.”
Complete these steps to configure a route list:
Step 1 From the Route Plan menu in the main Cisco CallManager Administration
window, choose Route/Hunt, then Route List.
Step 2 Click the Add a New Route List link. Give the new route list a name and
description, and choose the appropriate Cisco CallManager Group. Click
Insert. It is always best to follow good naming nomenclature (such as
PHOENIX_RL or DALLAS_RL).
Step 3 Click the Add Route Group button, shown in Figure 10-5, to add the
appropriate route groups. This action displays the Route List Detail
Configuration window. You can add route groups and set calling- and
called-party transformations at this point.
Figure 10-5 Cisco CallManager Route List Configuration
Step 4 Click Insert to return to the Route List Configuration window. Click Add
Route Group again to add additional route groups.
Step 5 Use the up and down arrows to prioritize the route groups.
Route Pattern Concepts and Configuration

In the voice over IP (VoIP) world, route patterns are the equivalent of static routes. The only
difference is that route patterns point to E.164 phone numbers (which is the standard defining the
international public telecommunication numbering plan used in the PSTN and data networks)
instead of IP addresses. External route patterns can point to either an individual gateway or a route
list. Here is the call process if the route pattern points to a route list, as shown in Figure 10-6:
1. When a user dials a number, Cisco CallManager analyzes the dialed digits. If the set of digits
matches a registered DN, Cisco CallManager routes the call to the internal destination.
2. If the set of digits matches an external route pattern, Cisco CallManager then parses the route
list that is associated with that route pattern. The route list contains a prioritized list of route
groups, and the route groups contain a prioritized list of voice gateways.
3. If the preferred voice gateway (in the first route group) is unavailable to handle the call, Cisco
CallManager passes the call to the next gateway, and so on, until it either finds a gateway to
route the call to or exhausts the list of gateways in the route group.
4. If Cisco CallManager exhausts the list of gateways in the route group, it passes the call to the
preferred gateway in the next route group in the route list. This process repeats until Cisco
CallManager finds a gateway that can handle the call or until it exhausts the list of route
groups in the route list. If Cisco CallManager is unable to find a gateway that can take the call,
the call fails and the end user receives a fast busy signal.
Figure 10-6 Cisco CallManager Route Pattern Design
IP
User DIALS EXTERNAL NUMBER
723-8912
Route
Pattern
723-xxxx
Digital_GW_1
First
Choice
Second
Choice
Digital_GW_2
Complete these steps to configure a route pattern in Cisco CallManager:
Step 1 From the Route Plan menu in the Cisco CallManager Administration
window, choose Route/Hunt, then Route Pattern.
Step 2 Click the Add a New Route Pattern link.
Step 3 To create a route pattern, enter your route pattern in the Route Pattern
field, as shown in Figure 10-7, and choose a route list or gateway from the
Gateway or Route List menu.
NOTE If you point a route pattern directly to a gateway, the gateway is now “used” in the
CallManager route plan. Remember, you can only use an endpoint once in the entire route plan.
Assigning it to a route pattern keeps it from being assigned to anything else in the route plan. If
you have multiple route patterns that used the same gateway, first assign the gateway to a route
group. Then, you can create numerous route lists that use the same route group and assign those
route lists to the various route patterns.
Figure 10-7 Cisco CallManager Route Pattern Configuration
Step 4 If you configure a route pattern to route off-network calls to the PSTN
(which most route patterns do), make sure to check the Provide Outside
Dial Tone check box. This feature plays a second dial tone for users when
they dial the outside access code.
TIP When a user dials an outside access code (such as 9) on a legacy PBX, the PBX
automatically seizes a trunk connection to the PSTN. The secondary dial tone that the user hears
is sent from the PSTN carrier. When a user dials an outside access code in a CallManager
environment, the Cisco CallManager generates the secondary dial tone and does not seize a
trunk line until the user finishes dialing and CallManager performs digit analysis on the full
string.
If Cisco CallManager receives a dial string for which multiple route patterns match, Cisco
CallManager must wait for the interdigit timeout (known as the T.302 timer, which is 15 seconds,
by default) before applying the longest-match rule and deciding which route pattern to use. You
can override the interdigit timeout behavior for a specific route pattern by checking the Urgent
Priority check box. When a route pattern is marked as urgent, Cisco CallManager immediately
routes any outbound calls that match the pattern. This approach avoids the interdigit timeout issue.
However, you should use this option very carefully because it can prevent users from reaching
certain destinations if it is configured incorrectly. In North America, Urgent Priority is most often
used for the 911 and 9911 route patterns.
If your route patterns point to specific gateways and not to route lists, you can set calling-party and
called-party transformations at the gateway level. If calling- and called-party transformations are
set here and the route pattern points to a route list, Cisco CallManager overrides the gateway-level
transformation settings and instead uses the transformation settings that are configured on the
route list. Transformation settings are fully discussed in Chapter 11.
Near the bottom of the Route Pattern Configuration window, you will find the ISDN Network
Specific Facilities Information Element configuration. This feature allows you to enter the
appropriate carrier identification code (up to four digits) to route long-distance calls to specific
interexchange carriers on a route pattern-by-route pattern basis.
As of Cisco CallManager Release 4.1, you can classify a route pattern as OnNet or OffNet with
an additional option to allow the device associated with the route pattern to override the call
classification settings configured in the Route Pattern Configuration window. By checking the
Allow Device Override check box (shown in Figure 10-7), the gateway or intercluster trunk
associated with the route pattern dictates the OnNet/OffNet classification rather than the route
pattern.
Route Pattern: Commonly Used Wildcards

A route pattern is a sequence of digits and other alphanumeric characters. If a route pattern
contains digits only, it represents a single route pattern match and matches only one destination.
By including nonnumeric wildcards in a route pattern, you can allow the route pattern to represent
multiple destinations. The purpose of using wildcards is to reduce the number of route patterns
that you need to configure. For example, a single route pattern of 1xxx would match all dialed
numbers from 1000 to 1999. Table 10-1 shows the route pattern wildcards available in Cisco
CallManager.
Table 10-1 Cisco CallManager Route Pattern Wildcards
Wildcard Description
X Matches a single digit. For example, 2xxx matches any number from 2000 to
2999.
@ Matches the entire North American Numbering Plan (NANP). This one wildcard
matches all numbers that are valid on the North American PSTN. For example
9@ matches an access code of 9 followed by any number that would be
considered valid on the North American PSTN. This is the fastest way to create
an external PSTN dial plan. Note that the @ symbol is really a macro
representing over 150 individual dial strings. Because of this, using the @ symbol
extensively in your route plan can cause significant memory consumption on the
Cisco CallManager.
! Matches one or more digits. Indicates a variable-length dial string that could be
any length of digits. The CallManager can only tell if a user is done dialing by
waiting for the interdigit timeout (15 seconds by default) or if the user presses the
# key on their telephone keypad. This wildcard is typically used to encompass
international dialing plans due to their varying lengths.
. Terminates access code. The “.” is not technically a wildcard, but, rather, can
separate an access code from a number for easier digit manipulation. For
example, the route pattern 9.@ allows an administrator to easily strip the “9”
before sending the call to the PSTN.
# Terminates interdigit timeout. The “#” is also not technically a wildcard, but,
rather, a key that a user can use to cause CallManager to skip the typical 15-
second interdigit timer and process the call immediately.
[xyz] Set of matching digits. For example, [458] matches one occurrence of either 4, 5,
or 8.
[x-y] Range of digits. For example, [3-9] matches one occurrence of either 3, 4, 5, 6, 7,
8, or 9. You can use the range notation along with the set notation. For example,
[3-69] matches one occurrence of either 3, 4, 5, 6, or 9.
[^x-y] Exclusion range. If the first character after the open square bracket is a carat, the
expression matches one occurrence of any digit (including * and #) except those
specified. For example, [^1-8] matches one occurrence of 9, 0, *, or #.
<wildcard>? A question mark that follows any wildcard or bracket expression matches zero or
more occurrences of any digit that matches the previous wildcard. For example,
9[12]? matches the following dial strings: 9, 91, 92, 912, 9122, 92121, and many
others. This wildcard is rarely used.
<wildcard>+ A plus sign that follows any wildcard or bracket expression matches one or more
occurrences of any digit that matches the previous wildcard. For example, 3[1-
4]+ matches 31, 3141, 3333, and many others. This wildcard is rarely used.
TIP The most commonly used wildcards in real-world Cisco CallManager deployments are as
follows:
X—Represents a single digit
@—Represents the NANP
!—Represents one or more digits
[x-y]—Number range notation
.—Access code termination
Route Pattern Examples

Table 10-2 illustrates various examples of route pattern wildcard implementations.
Table 10-2 Route Pattern Examples
Pattern Result
1234 Matches 1234
1*1x Matches numbers between 1*10 and 1*19
12xx Matches numbers between 1200 and 1299
13[25-8]6 Matches 1326, 1356, 1366, 1376, 1386
13[^3-9]6 Matches 1306, 1316, 1326
9011!# Matches any number that begins with 9011, is followed by one or more digits,
and ends with #; 901145123# and 90117893271211# are example matches
Although these examples show four-digit extensions, route patterns are more commonly used for
external numbers. Route patterns are normally in the form of seven-digit numbers, such as 723-
xxxx, or longer.
A great example is the 9.@ route pattern. The first digit of the route pattern matches a dialed digit
“9,” which users commonly use as a code to gain outside access to the PSTN. The second digit “.”
is used to identify the first digit as an access code and all numbers afterward as the dial string. The
third digit “@” is the wildcard that is used to match the North American Numbering Plan. The
NANP encompasses a number of dial strings, as presented in Table 10-3.
Table 10-3 North American Numbering Plan @ Wildcard
Dial String Description

Service calls Service calls are in the form of three- to four-digit numbers that are
used to access telephony services such as 911, 411, 611, and so on.
Local calls Local calls are in the form of a seven-digit number ([2-9]xx-xxxx) that
is used to dial within your local calling area.
Expanded local calls Expanded local calls are in the form of a 10-digit number ([2-9]xx-[2-
9]xx-xxxx) that is used to dial expanded-area local calls.
Long-distance calls Long-distance calls are in the form of a 10-digit number ([2-9]xx-[2-
9]xx-xxxx) that is used to place the calls directly to a long-distance
carrier.
Direct-dial long-distance Direct-dial long-distance calls are in the form of an 11-digit number
calls (1-[2-9]xx-[2-9]xx-xxxx) that is used to place a long-distance call
through a local carrier.
International calls International calls are in the form of 01 1 xx xxxxxxxxx where “xx” is
the country code. The actual length of the dial string depends on which
country the user is calling.
Digit Analysis
Call-routing component behavior can be counterintuitive. When a user places a call from a device
that is registered with Cisco CallManager, Cisco CallManager must analyze each dialed digit to
determine where to route the call. In collecting dialed digits, the call-routing component goes
through the following process:
1. Cisco CallManager compares the current sequence of dialed digits against the list of all route
patterns and determines which route patterns currently match. Then, Cisco CallManager
names the set of route patterns “potentialMatches.”
2. If potentialMatches is empty, the user-dialed digit string does not currently correspond with
a destination and the calling party will hear a reorder tone.
3. If potentialMatches contains one or more members, the call-routing component determines
the closest match. The closest match is the route pattern in potentialMatches that matches the
fewest number of route patterns. For example, the dialed digit string 1001 matches both route
pattern 1xxx and 10xx. Although there are 1000 different dialed digit strings that match 1xxx,
only 100 dialed digit strings match 10xx. Therefore, 10xx is the closest match.
Digit Collection
Figure 10-8 details a call-routing example in which one route pattern matches the dialed digits
exactly. The Cisco CallManager in this example includes the route patterns shown in the figure.
Figure 10-8 Cisco CallManager Digit Collection
User dial string: 1111 Match!

1111
121X Does not match
1[23]XX Does not match
131 Does not match
13[0–4]X Does not match
13! Does not match
When the user goes off hook, Cisco CallManager begins its routing process. The current set of
collected digits is empty. Every route pattern that Cisco CallManager has configured is a potential
match at this point. As long as the potentialMatches condition holds true, Cisco CallManager must
wait for more digits.
The user now dials a 1. At this time, there are no current matches and every route pattern is still a
potential match. The user dials another 1. At this point, Cisco CallManager eliminates route
patterns 121x, 1[23]xx, 131, 13[0-4]x, and 13! as potential matches. The only route pattern left is
1111. However, because there are no current matches and the potentialMatches condition is still
true, Cisco CallManager must continue to analyze digits. This requirement is in place because the
user might continue dialing and dial a string that does not match any entries.
The user dials another 1, which does not change anything. The condition currentMatches is false,
and potentialMatches is still true. The user dials 1 again. At this point, the route pattern 1111 is a
match, and the currentMatches condition is true. Cisco CallManager removes the route pattern
1111 from the potential matches table. Because there are no more route patterns in the potential
matches table, additional dialed digits will not cause Cisco CallManager to match a different route
pattern. At this point, Cisco CallManager routes the call to the dialed destination.
Closest Match Routing

Figure 10-9 details a closest-match call-routing example. The Cisco CallManager that is used in
this example includes the route patterns shown in the figure.
Figure 10-9 Cisco CallManager Closest Match Routing
User dial string: 1111 Does not match

1211
121X Match! (10 Destinations)
1[23]XX Match! (200 Destinations)
131 Does not match
13[0–4]X Does not match
13! Does not match
The user dials the digits 12. At this point, Cisco CallManager eliminates the route patterns 1111,
131, 13[0-4]x, and 13! as potential matches. This leaves route patterns 121x and 1[23]xx as
potential matches. Because there are no current matches and the potentialMatches condition is
true, Cisco CallManager continues to analyze digits.
The user dials another 1, which does not change anything. The condition currentMatches is false,
and potentialMatches is still true. The user dials 1 again. At this point, the route patterns 121x and
1[23]xx are current matches, and Cisco CallManager removes them from the potential matches
table. Because the potential matches table does not contain additional route patterns, additional
dialed digits will not cause Cisco CallManager to match any different route patterns. Now, Cisco
CallManager must decide where to route the call based on the route patterns that are available in
the current matches table. This is where the closest-match rule is applied. The route pattern 121x
matches 10 destinations (1210 to 1219). The route pattern 1[23]XX matches 200 destinations
(1200 to 1299 and 1300 to 1399). Cisco CallManager then routes the call to the gateway or route
list that is associated with the 121x route pattern.
Interdigit Timeout
If you configure a Cisco CallManager with route patterns that contain wildcards that match
multiple digits, CallManager must often wait for the interdigit timeout to expire before routing the
call. The ! wildcard usually represents a variable-length dial string and will never be an exact
match for a group of dialed digits. If the user presses # after dialing the last digit, Cisco
CallManager does not wait for the interdigit timeout. The Cisco CallManager in this example
includes the route patterns shown in Figure 10-10.
Figure 10-10 Cisco CallManager Interdigit Timeout
User dial string: 1111 Does not match

1311<timeout>
121X Does not match
1[23]XX Match! (200 Destinations)
131 Does not match
13[0–4]X Match! (50 Destinations)
13! Match! (Infinite Destinations)
In this example, the user has dialed the string 1311. This action causes Cisco CallManager to
eliminate the route patterns 1111, 121x, and 131. Cisco CallManager places the route patterns
1[23]xx, 13[0-4]x, and 13! in the current matches table. The 13! route pattern remains in the
potential matches table. The 13! route pattern ensures that the potentialMatches condition is
always true because Cisco CallManager has no way of knowing whether the user intends to
continue dialing. For example, the user might intend to dial the number 1311555. As long as the
potentialMatches condition is true, Cisco CallManager must continue to wait for dialed digits.
In this case, the only event that allows Cisco CallManager to select a destination is an interdigit
timeout. When the interdigit timeout timer expires, Cisco CallManager knows that no more digits
are forthcoming and can now make a final routing decision. In this example, the user has dialed
1311 and then stopped dialing digits. This action has triggered an interdigit timeout and caused
Cisco CallManager to make a final decision based on the following route patterns in the current
matches table: 1[23]xx, 13[0-4]x, and 13!. Because the dial string of 1311 matches multiple route
patterns, the closest-match rule is applied.
The route pattern 1[23]xx matches 200 destinations (1200 to 1299 and 1300 to 1399). The route
pattern 13[0-4]x matches 50 destinations (1300 to 1349). The route pattern 13! matches an infinite
number of destinations. Cisco CallManager uses this pattern only if it is the only route pattern in
the current matches table. The call is routed to the gateway or route list that is associated with the
13[0-4]x route pattern.
TIP The system interdigit timeout defaults to 15 seconds. To change it, change the value that
is associated with the Cisco CallManager service parameter TimerT302_msec (accessible from
Service > Service Parameters > Cisco CallManager). This parameter defines the duration of
the interdigit timer in milliseconds (ms). The default is 15,000 ms. The lowest configurable
value for the T302 timer is 3 seconds (3000 milliseconds). Cisco recommends the value of 5
seconds (5000 milliseconds)for the T302 timer.
Simple Route Plan Example

Figure 10-11 details a simple route plan. In this scenario, the network administrator has configured
two gateways for long-distance access to the PSTN (Digital_GW1 and Digital_GW2).
Digital_GW1 connects to a carrier that offers a long-distance rate of 7 cents per minute.
Digital_GW2 connects to a carrier that offers a long-distance rate of 10 cents per minute.
Figure 10-11 Simple Route Plan Example
IP
User dials number
723-XXXX
408-555-XXXX 836-XXXX
868-XXXX
Route Route
Local_GW LEC
Pattern Pattern
Cisco Access
First Choice Digital_GW_1 $0.07 per min.
RL_PSTN RG_PSTN
Second Choice Cisco Access $0.10 per min.

Digital_GW_2
The network administrator has created a route group RG_PSTN to group these gateways and give
first priority to Digital_GW1. The route list RL_PSTN uses the route group RG_PSTN. Currently,
users need only to call long distance to one destination, which is a remote office in San Jose,
California. Therefore, the administrator has created the San Jose route pattern 408-555-XXXX.
This route pattern then associates directly with the RL_PSTN route list.
Users also need to dial off-cluster to the PSTN to reach destinations within the local calling area.
A separate gateway (Local_GW) connects to the local exchange carrier (LEC) for local PSTN
calls. The administrator has defined the route patterns 723-XXXX, 836-XXXX, and 868-XXXX
for local calls. These route patterns point directly to the Local_GW gateway for local PSTN access
through the LEC.
NOTE You can only use a device in direct Route Patterns or Route Groups in a Cisco
CallManager route plan. For example, if you point the route pattern 723-XXXX to a gateway
device, you will no longer be able to add that device to a route group. Likewise, if you add a
gateway device to a route group, you will no longer be able to point route patterns directly to the
gateway device.
Summary
This chapter covered the design and configuration of a basic Cisco CallManager route plan. A
route plan is required to route external, or off-cluster, calls in a Cisco IP telephony network. By
understanding the call-routing process of Cisco CallManager, you can design your route plan to
take advantage of cost considerations and redundancy. A route plan consists of voice gateways and
trunks, route groups, route lists, and route patterns.
Route patterns should represent all valid digit streams. Route patterns can be assigned directly to
a gateway, or they can be assigned to a route list for more flexibility, such as setting a digital access
gateway as the first choice for the least expensive route.
A route list sets the route group usage order. If a route list is used, you must also configure route
groups.
A route group sets the access gateway device usage order. This order can be used to select the least
expensive route and allows overflow from a busy or failed device to an alternate device.
The route plan configuration order is to add the gateway, add a route group for the gateway, add a
route list for the route group, and add route patterns to the route list.
Review Questions
1. When building a Cisco CallManager route plan, which of the following should you configure
first?
a. route pattern
b. route list
c. route group
d. devices/trunks
2. When configuring route groups, you are also given the opportunity to perform digit
manipulation using calling- and called-party transformations. (True/False)
3. You are configuring your Cisco CallManager system for an advanced route plan. You access
the Device > Gateway menu and verify that you have an H.323 gateway configured at 10.1.1.1
connecting to the PSTN. You then access the Route Pattern Configuration window to add the
9.@ route pattern to the system. When you attempt to associate the H.323 gateway with the
route pattern, CallManager does not list the gateway in the drop-down selection box. What is
the most likely cause of the problem?
a. The H.323 gateway must already be used in a route group.
b. CallManager can only use MGCP gateways to handle PSTN connectivity.
c. The gateway should have been configured with a public IP address that is valid on the
Internet.
d. There is already a 9.@ route pattern in the CallManager configuration.
e. You must associate the gateway with a route group because route patterns can only point
to route lists.
4. Which interface does a voice gateway provide?
a. X.25 to SS7
b. VoIP to H.323
c. VoIP to PSTN
d. PSTN to PBX
5. Which of the following contains a list of gateways to use in precedence order?
a. route group
b. route list
c. translation pattern
d. route pattern
6. Which three of the following are valid wildcards? (Choose three.)
a. *
b. !
c. .
d. @
7. What can be placed into a route list?

a. Cisco CallManager servers
b. route patterns
c. route groups
d. devices
8. You are examining a Cisco CallManager route plan and notice the route pattern 34[45]67.
Which of the following extensions does this route pattern match?
a. 344567
b. 3467
c. 344
d. 3467
e. 34567
9. You have created a route pattern of 4xxx to reach extensions in another CallManager cluster.
The calls seem to work fine; however, users complain that they hear a second dial tone after
dialing the first digit of the remote extension. What is causing this problem?
a. The Cisco CallManager assumes all route patterns reach extensions external to the VoIP
network (the PSTN) and plays an outside dial-tone automatically.
b. You need to remove the check from the Provide Outside Dial-Tone check box for the
route pattern.
c. You need to check the box for the Provide Outside Dial-Tone for the route pattern.
d. This feature can be disabled through the CallManager Service parameters.
10. You have created the route pattern 7XX!. Which of the following strings match this route
pattern? (Select all that apply.)
a. 7
b. 71
c. 712
d. 7123
following topics:
■ Configuring route filters in Cisco

CallManager Administration to reduce the
number of route patterns or restrict calling to
undesirable locations
■ Modifying route patterns to use access codes

and discard digit instructions to convert the
dialed number to a number that is supported
by a national numbering plan
■ Configuring transformation masks to

manipulate the appearance of the number of
the calling party for outgoing calls and to
manipulate called numbers for PSTN
compatibility
■ Configuring translation patterns that

manipulate dialed digits before routing a call
to enable users to include a uniform dialing
plan between offices or to enable hot line
functionality
■ Describing how to access route plan reports

to view a listing of all the Call Park numbers,
Call Pickup numbers, conference numbers
(such as Meet-Me numbers), route patterns,
and translation patterns in the system
CHAPTER 11
Advanced Route Plans
Users of any phone system often need to reach a variety of destinations that include calls to
extensions located within the same site, to a different site within the same company (sometimes
with different dialing plans), and to other companies located within the same country or a
different country. Because these calls can take different paths, such as the IP WAN or a preferred
public switched telephone network (PSTN) carrier, completing these calls often requires dialing
various access codes, numbers of digits, or prefixes. At the same time, it is often prudent to
restrict certain destinations for all users, such as 900 numbers.
To require users to understand the specific dialing patterns necessary to reach these various
destinations is impractical and inconvenient. Digit manipulation, or the ability of Cisco
CallManager to add or subtract digits to comply with a specific internal dial plan or national
numbering plan, is the key to providing transparent dialing and creating a unified dialing plan.
Implementing route filters in Cisco CallManager Administration blocks access to specified area
codes.
This chapter covers route filters, discard digits instructions (DDIs), transformation and
translation patterns, and the route plan report to view all route patterns in a Cisco IP telephony
clustered solution.
Route Filters
When creating route patterns, you can use the “@” wildcard to represent all the routes defined
in the North American Numbering Plan (NANP). Although this is a simple way to provide
PSTN access to your internal users, you might be providing far more access than you intend. As
shown in Figure 11-1, the NANP includes high-expense patterns such as international dialing
access, service numbers (such as 411), and 900 numbers. You can assign route filters to route
patterns with the @ route pattern to help reduce the danger that full access to the NANP
provides. You can accomplish this reduction by filtering what is included in the @ (or 9.@) route
pattern.
210 Chapter 11: Cisco Unified CallManager Advanced Route Plans
Figure 11-1 @ Route Pattern Without Route Filters
9 [2–9]11 311, 611, 911 SERVICE
9 [2–9]XX XXXX 7-digit dialing by OFFICE CODE
10-digit local dialing by

9 [2–9]XX [2–9]XX XXXX LOCAL AREA CODE
11-digit long-distance dialing by
9 1 [2–9]XX [2–9]XX XXXX AREA CODE
International dialing by
9 011 3[0–469] ! COUNTRY CODE
When using the 9.@ route pattern, Cisco CallManager recognizes that dialing is complete when
the user dials 1 + 10 digits (signifying long-distance dialing) or just dials 10 digits (local area
codes without the 1). If the number dialed does not begin with a 1, Cisco CallManager considers
it a local area code and assumes that dialing is complete after 10 digits.
In an area where seven digits are dialed for local numbers, Cisco CallManager cannot recognize
which office exchange codes (NXXs) to use for routing unless you specifically code them as route
patterns.
NOTE NXX is the central office (CO) exchange code, which consists of three digits that
designate a particular CO or a block of 10,000 subscriber lines. N is any digit between 2 and 9,
and X is any digit between 0 and 9.
Generally, telephone company service providers arrange many NXXs in a given area code
contiguously where you can use route pattern wildcards to assist in your configuration. Coding
these individual route patterns for NXXs can be extremely difficult. You can use a route filter to
simplify this procedure.
A route filter called seven-digit dialing is always preconfigured in Cisco CallManager. You should
assign this route filter to any 9.@ route pattern in an area that uses seven-digit dialing. This route
filter removes all local area codes. If a dialed number does not begin with a 1, then it is a seven-
digit number, and Cisco CallManager considers dialing complete after seven digits. This situation
requires you to configure local area codes specifically as separate route patterns. Doing so is
generally not an issue because the number of area codes in a geographical region is usually small.
NOTE Route filters are only used with the @ route pattern and are not necessary if you have
configured a robust dial plan (that does not use the @ route pattern).
Route Filters 211
Route Filter Tags

You can design and configure route filters using a number of predefined tags in the Cisco
CallManager CCMAdmin web utility. Table 11-1 provides a list of tags available to you when
implementing route filters.
Table 11-1 Cisco CallManager Route Filter Tags
Tag Name Example Pattern Description

AREA-CODE 1 214 555 1212 The area code in an 11-digit long-
distance call
COUNTRY-CODE 011 33 123456# The country code in an
international call
END-OF-DIALING 011 33 123456# The #, which terminates interdigit
timeout for an international call
INTERNATIONAL-ACCESS 01 1 33 123456# The initial 01 of an international
call
INTERNATIONAL-DIRECT-DIAL 01 1 33 123456# The digit that denotes the direct-
dial component of an international
call
INTERNATIONAL OPERATOR 01 0 The digit that denotes the operator
component of an international call
LOCAL-AREA-CODE 214 555 1212 The area code in a 10-digit local
call
LOCAL-DIRECT-DIAL 1 555 1212 The initial 1 that is required for
some 7-digit calls
LOCAL-OPERATOR 0 555 1212 The initial 0 that is required for
operator-assisted local calls
LONG-DISTANCE-DIRECT-DIAL 1 214 555 1212 The initial 1 that is required for
long-distance direct-dialed calls
LONG-DISTANCE-OPERATOR 0 214 555 1212 The initial 0 that is required for
operator-assisted long-distance
calls
NATIONAL-NUMBER 011 33 123456# The national number component of
an international call
OFFICE-CODE 1 214 555 1212 The office exchange code of a
North American call
SATELLITE-SERVICE 011 88141234# A specific value that is associated
with calls to the satellite country
code
SERVICE 1 411 A value that provides access to
local telephony provider services
continues
Table 11-1 Cisco CallManager Route Filter Tags (Continued)
Tag Name Example Pattern Description

SUBSCRIBER 1 214 555 1212 A particular extension that is
served by a given exchange
TRANSIT-NETWORK 101 0321 1 214 A long-distance carrier code
555 1212
TRANSIT-NETWORK-ESCAPE 101 0321 1 214 The escape sequence that is used
555 1212 for entering a long-distance carrier
code
Configuring Route Filters

Route filter configuration occurs in two major steps:
Step 1 Configure route filter with necessary tags and arguments.

Step 2 Apply the route filter to a route pattern or translation pattern.
Just as with access lists on routers, you can create route filters all day and they will never make
any difference until you have applied them. Just like an access list, the sole purpose of a route filter
is to match criteria; how you apply the route filter determines if the criteria is permitted or denied.
To configure a route filter, use the Cisco CallManager Administration window:
Step 1 Choose the Route Plan menu.

Step 2 Choose Route Filter from the menu bar.
Step 3 Click the Add a New Route Filter hyperlink.
Step 4 Choose North American Numbering Plan from the Dial Plan menu.
Step 5 Enter a name in the Route Filter Name field. The name can consist of up
to 50 alphanumeric characters, and can contain any combination of spaces,
periods (.), hyphens (-), and underscore characters (_). Each route filter
name must be unique to the route plan.
After you have accomplished these initial steps, the Route Filter Clause Configuration window
appears, as shown in Figure 11-2.
Route Filters 213
Figure 11-2 Route Filter Clause Configuration Window
From this point, you can combine your tags with operators to define match conditions. Table 11-2
describes the four operators available when configuring route filters.
Table 11-2 Cisco CallManager Route Filter Operators
Operator Description
NOT-SELECTED Do not filter calls based on the dialed digit string associated with this tag.
EXISTS Filter calls when the dialed digit string associated with this tag is found.
DOES-NOT-EXIST Filter calls when the dialed digit string associated with this tag is not
found.
== Filter calls when the dialed digit string associated with this tag matches
the specified value.
The following are examples of match conditions using route filter tags and operators:
■ A route filter that uses the tag AREA-CODE and the operator DOES-NOT-EXIST selects
all dialed digit strings that do not include an area code.
■ A route filter that uses the tag AREA-CODE, the operator = =, and the entry 515 selects all
dialed digit strings that include the 515 area code.
■ A route filter that uses the tag AREA-CODE, the operator = =, and the entry 5[2-9]X selects
all dialed digit strings that include area codes in the range of 520 through 599.
■ A route filter that uses the tag TRANSIT-NETWORK, the operator ==, and the entry 0288,
along with the tag TRANSIT-NETWORK-ESCAPE, the operator ==, and the entry 101,
selects all dialed digit strings with the carrier access code 1010288.
Applying Route Filters

After you have configured the route filter, you must apply them to a route pattern or translation
pattern to define the permit or deny action.
NOTE Translation patterns are discussed later in this chapter.
To apply the route filter you have created to a route pattern, perform the following steps:
Step 1 Choose the Route Plan > Route/Hunt > Route Pattern menu selection.
Step 2 Choose the Add a New Route Pattern hyperlink.
Step 3 Define the route pattern as @ (or 9.@) to represent the NANP.
Step 4 Choose your configured route filter from the Route Filter drop-down list,
as shown in Figure 11-3.
Route Filters 215
Figure 11-3 Applying a Route Filter to a Route Pattern
Step 5 Choose a PSTN exit gateway or route list.

Step 6 Select the Route This Pattern radio button to allow the numbers matching
the filter to route to the PSTN or select the Block This Pattern radio button
to block the numbers matching the filter from reaching the PSTN.
Practical Route Filter Example

To demonstrate route filter configuration, imagine that a company wanted to create a route filter
that kept 1-900 numbers from being made available in the CallManager route plan. The first step
would be to create a route filter that matched the area code 900, as shown in Figure 11-4.
Figure 11-4 Route Filter Matching the 900 Area Code
After the route filter has been added to the configuration, you then need to apply it to a route
pattern to define the action required. Figure 11-5 illustrates the creation of a 9.@ route pattern
(representing the NANP) with the Match 900 Numbers route filter applied. The key action rests
in the Route This Pattern or Block This Pattern radio buttons. If you select to Route This
Pattern, you would add only 900 numbers to the Cisco CallManager route plan. Unless you had
a very strange organization, this is not a desired effect. Rather, you would select the Block This
Pattern radio button to block any numbers containing the 900 area code.
CallManager allows you to configure multiple 9.@ route patterns, provided each has a unique
route filter configuration. This provides flexibility when creating your route plan. By combining
multiple 9.@ route patterns with unique route filters, you can route (or block) exactly what you
want from the CallManager route plan.
TIP You could also accomplish this same objective by creating a route pattern of
1900XXXXXXX and choosing Block this Pattern under the route pattern configuration.
Discard Digit Instructions 217
Figure 11-5 Route Pattern Configuration
Discard Digit Instructions

Discard Digit Instructions (DDIs) allow conversions of a dialed number specific to a national
numbering plan. Typically, companies use a route pattern such as 9.@ to access the PSTN.
However, only the internal IP telephony network uses the 9 access code to reach the PSTN. If the
Cisco CallManager were to keep the access code prefixed to the number to forward to the PSTN,
the call would not complete. To avoid this, you can use DDIs to strip extra digits before the call
reaches the PSTN.
In general, administrators apply DDIs to route patterns that contain the @ wildcard; however, you
can use the DDI PreDot with route patterns that use the “.” wildcard even if the route patterns do
not contain the @ wildcard. Cisco CallManager applies DDIs to the called-party transformation
masks at the route pattern, the route details of a route list, or a translation pattern. DDI identifiers,
shown in Figure 11-6, are additive. The DDI PreDot 10-10-Dialing combines the effects of each
individual identifier. Table 11-3 depicts the most commonly used DDIs in the Cisco CallManager
route plan.
Table 11-3 Cisco CallManager Digit Discard Instructions

Digit Discard If the route pattern is 9.5@... Used For
Instructions Discarded Digits
PreDot 95 1 602 555 1212 Access codes
PreAt 95 1 602 555 1212 Access codes
11D/10D@7D 95 1 602 555 1212 Toll bypass
11D@10D 95 1 602 555 1212 Toll bypass
IntlTollBypass 95 011 33 1234 # Toll bypass
10-10-Dialing 95 1010321 1 602 555 1212 Suppressing carrier selection
Trailing-# 95 1010321 011 33 1234 # PSTN compatibility
You can configure DDIs at multiple places in the CallManager route plan. One of the more common
places is at the route pattern configuration. As shown in Figure 11-6, you can find the DDIs under the
Called Party Transformations near the end of the Route Pattern Configuration window.
TIP Digit Discard Instructions applied at the route pattern level are visible to the end user. For
example, if the caller dials 914085551212 and the DDI removes the 9, the user will see the
number change to 14085551212 on the LCD display of their IP Phone. DDIs applied at the route
group level (from within the route list) are not visible to the end user. Applying DDIs at the route
group level is covered later in this chapter.
Figure 11-6 Digit Discard Instructions

As you can see, CallManager offers a variety of combinations of all the DDIs listed in Table 11-3.
Transformation Masks
Dialing transformations allow the call-routing component to modify either the calling number or
the dialed digits of a call. Transformations that modify the calling number are calling-party
transformations; transformations that modify the dialed digits are called-party transformations.
Calling-party transformation settings allow you to manipulate the appearance of the calling-party
number for outgoing calls. A common application of a calling-party transformation is to use the
company external phone number of a calling station in place of the directory number (DN) for
outgoing calls. The calling-party number is used for Calling Line Identification (CLID). During
an outgoing call, the CLID is passed to each PBX, CO, and interexchange carrier (IXC) as the call
progresses. The CLID is also delivered to the calling party when the call is completed.
NOTE Caller ID is also referred to as Automatic Number Identification (ANI) .
Called-party transformation settings allow you to manipulate the dialed digits, or called-party
number, for outgoing calls. Examples of manipulating called numbers include appending or
removing prefix digits (outgoing calls), appending area codes to calls that are dialed as seven-digit
numbers, appending area codes and office codes to interoffice calls that are dialed as four- or five-
digit extensions, and suppressing carrier access codes for outgoing calls.
When configuring calling- or called-party transformations, a transformation mask operation

allows the suppression of leading digits, the change of some digits while leaving others
unmodified, and the insertion of leading digits. A transformation mask requires two pieces of
information: the number that you want to mask and the mask itself.
In the transformation mask operation, Cisco CallManager aligns the number with the mask so that
the last character of the mask aligns with the last digit of the number. Cisco CallManager uses the
corresponding digit of the number wherever the mask contains an X. If the number is longer than
the mask, the mask removes the leading digits. Figure 11-7 demonstrates the transformation mask
logic.
Figure 11-7 Transformation Mask Logic

Directory Number 35453
First
Transformation Mask 60211XXXXX
6021135453
Second
Transformation Mask 48085XX000
Result 4808535000
As you can see from Figure 11-7, the initial number (which could be dialed or caller ID
information, depending on the type of transformation mask you choose) passes through the first
transformation mask. By right-justifying the digits, CallManager passes the number 35453
through the first transformation mask. The digits match to the right-justified X wildcards, causing
it to pass through. CallManager prepends the additional digits to the left of the wildcards. As it
passes through the second transformation mask, the numbers are again right-justified. This time,
only the “35” digits pass through the X wildcards; CallManager replaces the rest of the string with
the hard-coded digits to the left and right of the X wildcards.
Calling-Party Transformations
The example in Figure 11-8 shows the applicable settings for calling-party transformations and
the order in which Cisco CallManager processes those instructions.
Figure 11-8 Calling-Party Transformations
Directory Number 35062
External Phone
Number Mask 21471XXXXX
2147135062
Calling-Party
Transformation
Mask 40885XX000
Caller ID 4088535000
You can configure three types of calling-party transformations in the call-routing component and
on route lists:
■ Use the external phone number mask, which instructs the call-routing component to use the
external phone number of a calling station rather than its DN or the caller ID information.
Without the external phone number mask, the calling number might appear as an extension
number to the PSTN rather than a fully routable PSTN phone number. You can apply the
external phone number mask on a line-by-line basis through the DN configuration screen on
the device.
■ The calling-party transformation mask allows the suppression of leading digits, leaves other
digits unmodified, and inserts leading digits. As shown in Figure 11-8, the post-external
phone number mask caller ID information is 2147135062. This passes through the calling-
party transformation mask of 40885XX0000, which allows only the numbers “35” to pass
through the XX wildcards. The resulting caller ID information is 4088535000.
■ Prefix digits allow the prepending of specified digits to the calling number.
Cisco CallManager applies the transformations in the order that is presented in the example.
TIP Remember, the calling-party transformations transform caller ID information (the

number of the person who is calling).
Called-Party Transformations
The example in Figure 11-9 shows the applicable settings for called-party transformations and the
order in which Cisco CallManager processes those instructions.
Figure 11-9 Called-Party Transformations
Dialed
9 1010321 18085551221
Number
DDIs 10-10-Dialing
9 18085551221
Calling-Party
Transformation
XXXXXXXXXX
Mask
8085551221
Prefix Digits 8
Called Number 8085551221

You can configure the following three types of called-party transformations in the call-routing
component and on route lists:
■ DDIs allow the discarding of digits in the dialed number. Such instructions are critical for
implementing toll-bypass solutions. This need arises when Cisco CallManager must convert
the long-distance number that the calling party has dialed into a local number. This number
allows Cisco CallManager to pass the digits to the PSTN. You can also use DDIs to discard
PSTN access codes, such as 9. Figure 11-9 demonstrates the removal of the 10-10-Dialing
portions of the call to eliminate other long-distance carriers a user might choose to use.
■ The called-party transformation allows the suppression of leading digits, changes the existing
digits while leaving others unmodified, and inserts leading digits. Figure 11-9 illustrates the
use of a called-party transformation mask consisting of ten “X” wildcards. This has the effect
of stripping any preceding digit beyond the ten wildcards (in this case, CallManager strips the
9 and the 1). This might be necessary when configuring toll-bypass on the CallManager. For
example, you might have a location that can access the 808 area code without incurring toll
charges. You can configure the Cisco CallManager to route the call across the IP WAN then
out a LEC.
■ Prefix digits allow the prepending of one or more digits to the called number. Figure 11-9
illustrates prefixing an 8 to the front of the original dialed number. This might be required by
a PBX at a remote site. For example, the call is transformed using the called-party
transformation mask, then prefixed with an 8 and routed across the IP WAN. When the PBX
at the remote site received the string, it recognizes the 8 as an outside access code and routes
the call to the PSTN LEC.
Cisco CallManager applies the transformation in the order that is presented in the example.
NOTE You could accomplish the same result shown in Figure 11-9 by simply applying a
called-party transformation mask of 8XXXXXXXXXX. This is a more efficient method; the
called-party transformations shown are used to illustrate the various transformations and the
order in which the CallManager applies them.
TIP Remember, the called-party transformations transform dialed-number information (the

number that a user called).
Configuring Calling- and Called-Party Transformation Masks

You can apply calling- and called-party transformations at the route pattern, route list, or
translation pattern configuration windows. The calling-party transformation setting that is used in
route lists applies to the individual route groups that make up the list rather than to the entire route
list. The calling-party transformation settings that are assigned to the route groups in a route list
override any calling-party transformation settings that are assigned to a route pattern that is
associated with that route list.
Because you can be more specific, network administrators usually apply transformation masks at
the route list level. In this way, you can assign a different transformation mask for each route group
in the route list. Transformation masks configured at the route list level have priority over those
configured at the route pattern level because they are processed last. If you have configured a
transformation at the route pattern level, it becomes more of a “global” translation, that is, as soon
as the pattern is matched, the transformation takes effect. As the route pattern sends the call to the
route list and the prioritized route group is chosen, the transformations relating to that specific
route group apply second, transforming the already transformed number from the route pattern
into whatever you have defined.
For example, in the network illustrated in Figure 11-10, a network administrator has two route
groups created: the PSTN route group and the IP WAN route group. Both of these route groups
contain multiple gateways that connect to their respective networks. When Cisco CallManager
forwards a call to a gateway in the PSTN route group, the network administrator applies a mask
that transforms the number into an E.164-compliant phone number. However, when Cisco
CallManager uses a gateway from the IP WAN route group, Cisco CallManager leaves the number
as a four-digit extension.
Figure 11-10 Transformation Network Design
Secondary Voice Path

Prepend “1215555” and
Gatekeeper(s) send to PSTN
V
V
PSTN
IP IP IP
IP IP IP Philadelphia
(215) 555-xxxx
Arizona
IP WAN 4-Digit Internal Dialing
(480) 526-xxxx
4-Digit Internal Dialing
Primary Voice Path

Leave four digit extension
as dialed
Transformation Example
Figure 11-11 summarizes how transformations to the called-party (dialed digits) and to the
calling-party numbers are made within Cisco CallManager. In Figure 11-11, a user dials a number
to which Cisco CallManager first applies a calling-party transformation (“calling party” refers to
the person who originated the call). This action changes the caller ID number that is displayed on
the destination phone. Cisco CallManager then applies a called-party transformation to change the
number that is dialed.
Figure 11-11 Called-Party Transformations

1 2 3
User Dial
Users Route Pattern DDI
Numbers
A - 5062 A - 91234 Wildcards allow all

IP dialed numbers to match 9.1XXX Discard “9”
one route pattern.
B - 5063 B - 91324
IP
C - 5064 C - 91432 A–1234

User Dialed
IP B–1324
Numbers
C–1432
6
To: 1000
From: 5000 5 Caller ID 4
A–1000 A–5000
B–1000 “X000” B–5000 “X000”
IP C–1000 C–5000
Extension 1000 User Dialed Transform Transform

User DNs
Rings Numbers Called Number Calling Number
The two transformations are explained in the figure and, for user A specifically, in the following
steps:
1. User A has a DN of 5062. This user dials DN 91234.

2. The dialed number matches the route pattern 9.1xxx.
3. The DDIs contain instructions to discard the 9. The dialed number is now 1234.
4. The calling number 5062 now passes through the calling-number transformation mask, which
contains instructions to change the last three digits of the calling party number to 000. The
new calling number is 5000.
5. Cisco CallManager then passes the called number 1234 through the called-number
transformation mask X000, which changes the dialed number to 1000.
6. The result is a calling-party number of 5000 and a called-party number of 1000.
Translation Patterns
Cisco CallManager uses translation patterns to manipulate dialed digits before routing a call. In
some cases, the dialed number is not the number that is used by the system. In other cases, the
dialed number is not a number that is recognized by the PSTN.
Digit manipulation and translation patterns are used frequently in cross-geographical distributed
systems where, for instance, the office codes are not the same at all locations. In these situations,
a uniform dialing plan can be created, and translation patterns can be applied to accommodate the
unique office codes at each location. The following are additional examples when you can use
translation patterns:
■ Routing emergency calls to security desks and operator desks (such as sending emergency
numbers to a local security office or routing ‘0’ to the local operator extension)
■ Hot lines with a need for private line, automatic ringdown (PLAR) functionality
■ Extension mapping from the public to a private network
Translation patterns use the results of called-party transformations as a set of digits for a new
analysis attempt. The second analysis attempt might match a translation pattern. In this case, Cisco
CallManager applies the calling- and called-party transformations of the matching translation
pattern and uses the results as the input for another analysis attempt. For example, you have a
called-party transformation set up under a route pattern that transforms the dialed number 5555 to
0. You could then have a translation pattern defined to match the dialed number 0 and transform it
to an operator extension or hunt group number. To prevent routing loops, Cisco CallManager
breaks chains of translation patterns after ten iterations.
Translation Pattern Configuration

Configuration of a translation pattern is similar to configuration of a route pattern. Each pattern
has calling- and called-party transformations and wildcard notation. The difference is that when
Cisco CallManager applies the translation pattern, it starts the digit-analysis process over and
routes the call through a new path if necessary.
To configure a translation pattern, choose the Route Plan menu and choose Translation Pattern.
After you click the Add a New Translation Pattern hyperlink, the Translation Pattern
Configuration window appears, as shown in Figure 11-12. You can define the route pattern to
match (in the Translation Pattern field) and the calling- or called-party transformation settings that
you want to apply.
Translation Patterns 227
TIP Many new Cisco CallManager administrators get the translation and transformation terms
confused. Cisco CallManager allows you to create translation patterns that contain both calling-
and called-party transformation masks to modify the caller ID or dialed-number information.
Translation patterns are usually used to transform dialed digits, so the called-party
transformation masks are more frequently used.
Figure 11-12 Translation Pattern Configuration Window
Practical Use of a Translation Pattern

Figure 11-13 shows an application for translation patterns. When the Direct Inward Dial (DID)
range from the CO does not match the internal DN range, you can use a translation pattern to make
the connection.
Figure 11-13 Practical Translation Pattern Example
PSTN
V
PSTN DID Range

408-555-1xxx
IP IP IP
Employee Attendant
Phones (4111)
Internal Extensions
4XXX
In Figure 11-13, a company has a PSTN DID range of 408-555-1xxx. However, all of the internal
four-digit extensions begin with 4xxx. When the company receives an incoming call, the company
could use DDIs to remove the 555 from the beginning of the number. However, the 1xxx extension
still remains. Instead, the translation pattern could apply a 4XXX called-party transformation
mask. This mask would convert the 1xxx external DID range to a 4xxx internal range. After Cisco
CallManager applies the transformation mask, it reanalyzes the dialed number and directs it to the
correct internal extension. So, to summarize (and simplify), the following configuration would be
created:
Translation Pattern: 4085551XXX

Called-Party Transformation Mask: 4XXX
Resulting Phone Number: 4XXX
If a call came into the DID 4085551111, CallManager would convert the dialed digits to 4111.
Route Plan Report 229
Route Plan Report

The route plan report is a listing of all the Call Park numbers, Call Pickup numbers, conference
numbers (such as Meet-Me numbers), route patterns, and translation patterns in the system. The
route plan report allows you to view either a partial or full list and go directly to the associated
configuration windows. You can accomplish this by selecting a route pattern, partition, route
group, route list, Call Park number, Call Pickup number, conference number, or gateway.
The route plan report allows you to save report data into a comma-separated values (CSV) file that
you can import into other applications (such as Microsoft Excel). The CSV file contains more
detailed information than the web pages, including DNs for phones, route patterns, and translation
patterns.
To generate a route plan report, use the Route Plan > Route Plan Report selection. The Route
Plan Report window shown in Figure 11-14 appears. From here, you can either generate a route
plan report to view in the web interface or click the View in File hyperlink to download the route
plan as a CSV file.
Figure 11-14 Generating a Route Plan Report

Summary
This chapter covered the design and configuration of an advanced Cisco CallManager route plan.
The concepts started by using the route filter configurations to screen the 9.@ NANP route pattern.
By default, the @ wildcard includes numbers that might be undesirable for a corporate entity. By
using route filters, you can limit the number of patterns that are added when creating a PSTN dial-
plan using the @ wildcard.
You can also use discard digit instructions (DDIs) to manipulate the digits sent to a destination,
especially the PSTN. Most corporations have a number that is dialed (such as 9) to indicate an
outside PSTN call. You can use DDIs to strip this access code before the call leaves to the PSTN.
You can use DDIs in a number of circumstances to properly transform incoming or outgoing calls,
many of which are seen in Chapters 12-14.
You can use calling- and called-party transformations to change caller ID information (calling) or
dialed digits (called). These masks give you complete flexibility to transform this information
however you see fit. Because you can apply them at nearly every level of the CallManager route
plan, you can make differing modifications to dialed-number information depending on the
gateway the CallManager decides to use to route the call. Translation patterns can also be added
to the route plan for last resort modifications using calling- or called-party transformations.
Review Questions
You can find the solutions to these questions in Appendix A, "Answers to Review Questions."
1. What does Cisco CallManager do when dialed digits pass through a translation pattern?
a. extends the call to the destination
b. forwards the call to a route pattern
c. selects the closest match to that pattern
d. sends the transformed digits through digit analysis one more time
2. When DN 1111 calls and a calling transformation mask of 972555XXXX is applied, which
CLID is sent?
a. 1111
b. 5551111
c. 9725551111
d. 19725551111
3. What are the final digits that Cisco CallManager sends when the discard digits instruction
PreDot is applied to the 9808.5550150 pattern?
a. 98085550150
b. 5550150
c. 95550150
d. 8085550150
4. You have created a route filter called “Match 900 numbers” where you have defined a
condition, “AREA-CODE == 900”. What must you do to put this route filter into effect?
a. You need to apply the route filter to a route pattern and select the allow or deny action.
b. You need to apply the route filter to a route list and select the allow or deny action.
c. You need to create a transformation pattern.
d. Nothing needs to be done; the route filter takes effect as soon as it is inserted into the
configuration.
5. You apply the DDI 11D@10D to the route pattern 16025551212. What digit(s) are discarded?
a. 1
b. 1602
c. 1602555
d. 555
6. You are editing the properties for the 9.@ route pattern. You apply a calling-party
transformation mask of 8. What does this accomplish?
a. The dialed digits are prefixed with an 8.
b. The caller ID information is prefixed with an 8.
c. The dialed digits are transformed into an 8.
d. The caller ID information is transformed into an 8.
7. What information does a called-party transformation mask modify?
a. the long-distance code information
b. the dialed-digit information
c. the caller ID information
d. It depends where the mask is applied.
8. You apply a prefix digit of 99 and a calling-party transformation mask of 555XXXX under a
PSTN route pattern. A user at extension 5123 dials a PSTN number; what information is
displayed for the caller ID information on the external phone?
a. 555
b. 99555
c. 5555123
d. 995555123
e. 5559912
9. You have created a route pattern of 4xxx to reach extensions in another CallManager cluster.
To properly forward the number, you have added a prefix-digit transformation of 8. In
addition, you created a translation pattern of 8.xxxx that applies a PreDot DDI. What dialed-
number information does the Cisco CallManager forward?
a. The dialed digits will forward to the other side as 4xxx.
b. The dialed digits will forward to the other side as 84xxx.
c. The dialed digits will forward to the other side as xxx.
d. The CallManager will play a fast busy signal.
10. Which transformation changes caller ID information?
a. translation patterns
b. calling-party transformations
c. called-party transformations
d. route patterns
following topics:
■ Defining call-distribution components and

algorithms
■ Describing how hunting differs from

forwarding and how the Call Coverage
feature extends hunting to allow final
forwarding if hunting exhausts
■ Configuring line groups, hunt lists, and hunt

pilots to create a hunt group
■ Configuring hunting with internal and

external forwarding support specifications
for busy, no answer, and no call coverage
■ Given usage scenarios, determining the

hunting and forwarding behavior that results
CHAPTER 12
Configuring Hunt Groups and
Call Coverage
Many businesses have sales or service support departments that service inbound calls from
customers. These businesses typically need several phone lines and a method to make the lines
work together. If one representative is busy or not available, calls will be routed to other
members of the group until it is answered or forwarded to an auto-attendant or voice mail. Hunt
groups are the mechanisms that help these businesses manage inbound calls. A hunt group is a
group of telephone lines that are associated with a common number. When a call comes in to
the number associated with the hunt group, the call cycles through the group of lines until an
available line is found. This process is known as hunting.
This chapter discusses how to configure hunt groups and enable the Call Coverage feature to
ensure that the calling party will receive final forwarding treatment if hunting fails (that is, when
no hunt party answers, either because the hunt has exhausted the list of hunt numbers or because
it has timed out).
Call Distribution Components

Line groups, hunt lists, and hunt pilots work together to provide call-distribution capabilities in
Cisco CallManager Administration. Call distribution is the ability of a caller to dial a number
and have the call extended in an ordered manner to members of a group. An example of this
functionality is a company 800 number to reach a company technical support department.
A line group contains directory numbers (DNs) and designates the order in which DNs are
chosen.
A hunt list contains one or more prioritized line groups.
A hunt pilot is a number that is associated with a hunt list. The hunt pilot can be called directly
(for example, to a technical support hotline for a company), or can be reached through
forwarding (for example, a caller places a direct call to a technical support group member, and
if that member is not available, the call is forwarded to the hunt pilot number). Figure 12-1
demonstrates the proper design of the Cisco CallManager call distribution components.
236 Chapter 12: Configuring Hunt Groups and Call Coverage
Figure 12-1 Cisco CallManager Call Distribution Components

1-800-555-0111
Hunt Pilot
Hunt List
Line Group 1 Line Group 2
1000 1001 1003 1004
IP IP IP IP
Line Groups
A line group allows you to designate the distribution mechanism where DNs are chosen.
Line groups contain the following components, shown in Figure 12-2:
■ Members—Directory numbers that are in one of these states:
— Idle—Not serving any call

— Available—Serving an active call but able to accept new calls
— Busy—Unable to accept any calls
■ Distribution Algorithm—A method for distributing calls to members.
■ Ring No Answer Reversion (RNAR) timeout value—A mechanism for determining how to
handle calls that go unanswered. The RNAR is a value, in seconds, after which Cisco
CallManager will distribute a call to the next available or idle member of this line group or to
the next line group if the call is not answered and if the first hunt option, Try Next Member;
Then, Try Next Group in Hunt List, is chosen. The RNAR timeout applies at the line-group
level to all members.
■ Hunt options—The ability to roll past members who are busy, not available, or do not
answer, continuing until the call is answered or options are exhausted.
Figure 12-2 Line Group Components
Cisco CallManager distributes a call to idle or available members of a line group based on the call-
distribution algorithm and on the RNAR setting.
Call-Distribution Algorithms
Line groups contain an algorithm that controls how calls that come in on the hunt pilot are
distributed to members, as follows:
■ Top down—If you choose this distribution algorithm, Cisco CallManager distributes a call to
idle or available members starting from the first idle or available member of a line group to
the last idle or available member. Referring to Figure 12-3, a top-down distribution algorithm
would extend the next call to 1000, then to 1001 (if 1000 was unavailable), then to 1002 (if
1001 was unavailable), then to 1003 (if 1002 was unavailable). The next incoming call would
then be sent back to 1000 to start the process over again.
Figure 12-3 Call-Distribution Algorithms
Line Group
IP IP IP IP
1000 1001 1002 1003
Idle 10 min. Available Idle 1 min. Idle 5 min.
■ Circular—If you choose this distribution algorithm, Cisco CallManager distributes a call to
idle or available members starting from the (n+1)th member of a line group, where the nth
member is the last member to receive a call. Because of this, Circular distribution is
commonly referred to as round robin. If the nth member is the last member of a line group,
Cisco CallManager distributes a call starting from the top of the line group. Referring to
Figure 12-3, assume that Cisco CallManager extended the last call to 1002 (n). The next call
that comes in on the hunt pilot number would go to 1003 (n + 1).
■ Longest idle time—If you choose this distribution algorithm, Cisco CallManager distributes
a call starting from the member of a line group who has been idle longest to the member who
has been idle for the shortest time. Referring to Figure 12-3, assume that 1000 has been idle
for 10 minutes and 1003 has been idle for 5 minutes. A longest idle time distribution
mechanism would extend the call to 1000, and the next incoming call would go to 1003.
■ Broadcast—If you choose this distribution algorithm, Cisco CallManager distributes a call
to all idle or available members of a line group simultaneously.
Hunt Options
Hunt options configured in the line group apply to members in one of the three states: no answer,
busy, or not available. Not available is a state triggered by the Do Not Disturb (DND) line state,
covered later in the chapter. For a given distribution algorithm, the hunt option specifies where
CallManager should distribute a call next if a member of a line group is busy, does not answer, or
is not available. The Line Group Configuration window provides the following options for each of
the three hunt options:
■ Try Next Member, Then, Try Next Group in Hunt List (Default)—Distributes a call to idle
or available members. If all lines respond with no answer (or busy or not available), try the
next line group in a hunt list.
■ Try Next Member, But Do Not Go to Next Group—Distributes a call to idle or available
members. Stops hunting upon reaching the last member of the current line group.
■ Skip Remaining Members, and Go Directly to Next Group—Skips the remaining

members of current line group when the RNAR timeout value elapses for the first member (or
first member is busy or not available). Hunting proceeds to the next line group in a hunt list.
■ Stop Hunting—Stops hunting after trying to distribute a call to the first member of current
line group if the member does not answer (or first busy member or first unavailable member).
TIP Distributing the call to other line groups can be useful in an environment where you have
Tier 1 and Tier 2 tech support. If all the lines in Tier 1 are busy, the call can spill over to the Tier
2 group. In addition, the Tier 2 employees will see the call redirect information so they can
answer the call appropriately.
Call Distribution Scenarios: Top-Down Example

Figure 12-4 shows an example of a line group (Line Group 1) with the following setup
information:
■ Line Group 1—Contains DNs 1000, 1001, and 1002
■ Distribution algorithm—Top down
■ RNAR timeout—10 seconds. After 10 seconds, Cisco CallManager will distribute a call to
the next available or idle member of this line group or to the next line group if the call is not
answered and if the first hunt option, Try Next Member; Then, Try Next Group in Hunt List,
is chosen.
■ Hunt Options—
— RNA—Try Next Member; Then, Try Next Group in Hunt List

— Busy—Try Next Member; Then, Try Next Group in Hunt List
— Not available—Try Next Member; Then, Try Next Group in Hunt List
Figure 12-4 Call Distribution Example

1-800-555-0111
Hunt Pilot
Hunt List
Line Group 1
1000 1001 1002
IP IP IP
The call flow is as follows:
1. Caller calls hunt pilot 1-800-555-0155.

2. 1000 rings and is not answered for 10 seconds (RNAR timeout).
3. The call extends next to 1001, following the top-down algorithm and RNAR setting.
4. 1001 rings for 3 seconds.
5. 1001 is answered.
Hunting and Forwarding

Hunting differs from call forwarding, although both allow calls to be redirected. Hunting allows
Cisco CallManager to extend a call to one or more lists of numbers, where each such list can
specify a hunting order that is chosen from a configurable set of algorithms. When a call extends
to a hunt party from these lists and the party fails to answer or is busy, hunting resumes with the
next hunt party. (The next hunt party varies depending on the current hunt algorithm.) Hunting
thus ignores the Call Forward No Answer (CFNA) or Call Forward Busy (CFB) settings for the
attempted party.
Call forwarding allows detailed control as to how to extend (divert and redirect are equivalent
terms for extend) a call when a called party fails to answer or is busy and hunting is not taking
place. For example, if the CFNA setting for a line is set to a hunt pilot number, an unanswered call
to that line diverts to the hunt pilot number and thus begins a hunt.
Configuring Line Groups, Hunt Lists, and Hunt Pilots 241
Starting with Cisco CallManager Release 4.1, Cisco CallManager offers the ability to redirect a
call when hunting fails (that is, when hunting terminates without any hunt party answering, either
because the list of hunt numbers exhausts or because the hunt process times out). If used, this final
redirection constitutes a Call Forwarding action. Therefore, the Hunt Pilot Configuration window
in Cisco CallManager Administration (choose Route Plan > Route/Hunt > Hunt Pilot) includes
call forwarding configuration concepts that are similar to those found in the Directory Number
Configuration window (Forward No Answer/Forward Busy).
In Cisco CallManager Release 4.0, hunting stops either when one of the hunt parties answers the
call or when the hunt list is exhausted. When hunting stops due to exhaustion, the caller receives
a reorder tone (or an equivalent announcement).
Configuring Line Groups, Hunt Lists, and Hunt Pilots

To access the line group, hunt list, and hunt pilot configuration windows in Cisco CallManager
Administration Release 4.1, choose Route Plan > Route/Hunt, as shown in Figure 12-5.
Figure 12-5 Hunt Configuration Menu

When configuring hunting, follow this general process:
Step 1 Create the line groups, add members, and configure the distribution
algorithm and hunt options.
Step 2 Create the hunt list and add the line groups.
Step 3 Create the hunt pilot, associate the hunt list with the hunt pilot, and
configure hunt forward settings.
These steps are covered in more detail in the following topics.
Configuring Line Groups

The general process for configuring a line group follows. The DNs that will become the members
of the line group must already exist in the database before you can complete this procedure.
Step 1 Choose Route Plan > Route/Hunt > Line Group.

Step 2 Click Add a New Line Group. The Line Group Configuration window
shown in Figure 12-6 appears.
Figure 12-6 Line Group Configuration Window

Step 3 Enter a name in the Line Group Name field. The name can contain up to
50 alphanumeric characters and can contain any combination of spaces,
periods (.), hyphens (-), and underscore characters (_). Ensure that each line
group name is unique to the route plan.
Step 4 Configure the distribution algorithm, hunt options, and RNAR timeout as
desired, or leave them at their default values.
To add members to the line group, follow this procedure:
Step 1 If you need to locate a DN, choose a route partition from the Route
Partition drop-down list, enter a search string in the Directory Number
Contains field, and click Find. To find all DNs that belong to a partition,
leave the Directory Number Contains field blank and click Find.
Step 2 A list of matching DNs is displayed in the Available DN/Route Partition
pane.
Step 3 In the Available DN/Route Partition pane, select a DN to add and click
Add to Line Group to move it to the Selected DN/Route Partition pane.
Repeat this step for each member that you want to add to this line group.
Step 4 In the Selected DN/Route Partition pane, choose the order in which the
new DNs will be accessed in this line group. To change the order, click a
DN and use the up and down arrows to the right of the pane.
Step 5 Click Update to add the new DNs and to update the DN order for this line
group.
TIP A single DN can be a member of multiple Line Groups
Notice the RNA Reversion Timeout (RNAR), which defaults to 10 seconds. This is the time, in
seconds, after which Cisco CallManager will distribute a call to the next available or idle member
of a line group or to the next line group if the call is not answered and if the first hunt option, Try
Next Member; Then, Try Next Group in Hunt List, is chosen.
Configuring Hunt Lists

To add a hunt list in Cisco CallManager Administration 4.1, follow these steps:
Step 1 Choose Route Plan > Route/Hunt > Hunt List.

Step 2 Click Add a New Hunt List.
Step 3 In the Hunt List Name field, enter a name. The name can comprise up to
50 alphanumeric characters and can contain any combination of spaces,
periods (.), hyphens (-), and underscore characters (_). Ensure that each
hunt list name is unique to the route plan.
Step 4 Choose a Cisco CallManager group from the drop-down list. The group
must already exist in the database; you cannot create a new group from this
window.
Step 5 To add this hunt list, click Insert. The Hunt List Configuration window
Figure 12-7 Hunt List Configuration Window
NOTE After clicking the Insert button, a popup message reminds you that you must add at
least one line group to this hunt list for it to accept calls.
Step 6 The Hunt List Configuration window displays the newly added hunt list.
Step 7 Add at least one line group to the new hunt list. To add a line group, click
Add Line Group. The Hunt List Detail Configuration window is
displayed.
Step 8 From the Line Group drop-down list, choose a line group to add to the hunt
list.
Step 9 To add the line group, click Insert. The line group name is displayed in the
Hunt List Details list on the left side of the window.
Step 10 To add more line groups to this list, click Add Line Group and repeat Step 6 and
Step 7.
Step 11 When you are finished adding line groups to the hunt list, click Update.
Step 12 Click Reset to reset the hunt list. When the popup windows are displayed,
click OK.
Cisco CallManager accesses line groups in the order in which they are shown in the hunt list. You
change the access order of line groups by selecting a line group from the Selected Groups pane
and clicking the up or down arrow on the right side of the pane to move the line group up or down
in the list.
Configuring Hunt Pilots

Before you create a hunt pilot, ensure that the following items are configured in Cisco
CallManager:
■ Hunt list
■ Partition (unless you are using None)
■ Route filter (unless you are using None)
You can then perform the following steps to configure a hunt pilot:
Step 1 Choose Route Plan > Route/Hunt > Hunt Pilot.

Step 2 Click Add a New Hunt Pilot. The Hunt Pilot Configuration window shown
in Figure 12-8 appears.
Figure 12-8 Hunt Pilot Configuration Window
Step 3 Enter the hunt pilot number in the Hunt Pilot number field.
Step 4 Assign the hunt pilot to a hunt list using the Hunt List drop-down menu.
Step 5 (Optional) Assign the hunt pilot to a partition and configure other settings,
such as hunt forward, as desired.
NOTE If you do not configure Hunt Forward Settings for Forward Hunt No Answer and
Forward Hunt Busy, CallManager returns a reorder tone if all members assigned to the hunt pilot
are unavailable.
TIP The Use Personal Preferences check box under the Hunt Forward Settings enables the
Call Forward No Coverage settings in the Directory Number Configuration page for the original
called number that forwarded the call to this hunt pilot. If checked, CallManager ignores the
Destination and Calling Search Space settings configured under the hunt pilot.
The Hunt Pilot Configuration window also allows you to define a Maximum Hunt Timer (in
seconds), which dictates the length of time a call to the hunt pilot number will search through
members. This setting overrides the Ring No Answer Reversion (RNAR) setting configured under
the line group. For example, if you have three members of a line group configured for a RNAR
setting of 10 seconds per member, the call hunts for a maximum of 30 seconds before returning a
status of No Answer from the line group. However, if you have defined a Maximum Hunt Timer
of 25 seconds on the Hunt Pilot Configuration window (shown in Figure 12-8), the call would stop
hunting through the line group after moving through the first two members (10 seconds each) and
ringing for 5 seconds on the third member.
Summary
This chapter covered the design and configuration of a Cisco CallManager hunt group. You can
use hunt groups to define a group of DNs to search through for an incoming call. This is useful in
an extraordinarily large number of environments, from receptionist pools for incoming calls to
technical support help desk representatives.
Configuring hunt groups is very similar to configuring the CallManager route plan. The
configuration is performed from the bottom up, first starting by adding member DNs to a line
group and choosing the distribution algorithm. You can then group one or more line groups into a
hunt list, which is nothing more than an ordered list of line groups to which CallManager should
distribute calls. You will finally create the hunt pilot, which defines a trigger DN and points to the
hunt list that should receive the incoming calls.
Review Questions
1. In addition to checking the Use Personal Preferences check box in the Hunt Forward
Settings section of the Hunt Pilot Configuration window, what else must be configured to
enable final forwarding for a call that is forwarded to the hunt pilot?
a. maximum hunt timer and RNAR timers in the Line Group Configuration window
b. the Call Forward No Coverage settings for the original called number that forwarded the
call to the hunt pilot
c. the Destination and Calling Search Space fields in the Hunt Forward Settings section
of the Hunt Pilot Configuration window
d. the Forward Busy and Forward No Answer settings for the original called number that
forwarded the call to the hunt pilot
2. Assume a line group with five members: party 1 (DN 5001), party 2 (DN 5002), party 3 (DN
5003), party 4 (DN 5004), and party 5 (DN 5005). The top-down distribution algorithm is
applied to the line group. Party 3 answered the last call that came in on the hunt pilot. To
which party will Cisco CallManager extend the next call that comes into the hunt pilot?
a. 1
b. 2
c. 3
d. 4
e. 5
3. A hunt group has been configured with a maximum hunt timer of 15 seconds on the Hunt Pilot
Configuration window. However, the line group with five members has been configured with
a RNA reversion timeout value of 5 seconds. When an incoming call comes in for the hunt
group, how long will the hunting process occur before the call is forwarded?
a. 5 seconds
b. 15 seconds
c. 25 seconds
d. 40 seconds
4. Which of these components are absolutely required to configure a hunt group? (Select all that
apply.)
a. hunt pilot
b. hunt list
c. line group
d. route list
e. gateway
f. route pattern
5. If a line group member DN is marked as an available line, what state is it in?
a. It is not serving a call at this time.
b. It is serving a call but can accept a new call.
c. It is serving a call but cannot accept a new call.
d. It cannot accept any calls.
6. Which of the following call distribution algorithms rings all members of the line group at the
same time?
a. multicast
b. broadcast
c. group
d. circular
7. You are looking through a hunt group configuration in Cisco CallManager. Under the line
group configuration window, you notice that the No Answer hunt option is configured with
the Stop Hunting selection. What does this cause?
a. The CallManager will hunt to the first member in the line group and stop if the member
is unavailable.
b. The CallManager will hunt through the current line group and stop if the members are
unavailable.
c. The CallManager will hunt through all the line groups and stop if the members are
unavailable.
d. The CallManager will not hunt through the line group, but will divert the call to the call
forward settings.
8. A hunt pilot receives a call and passes it down to the hunt list, which distributes the call to
three members of a line group. The call returns from the line group with a busy status. Under
the call forwarding section of the hunt pilot, the Use Personal Preferences check box is
checked for both the No Answer and Busy fields. How does the CallManager handle the call?
a. The call is forwarded to the destination defined under the Forward Hunt Busy section of
the hunt pilot.
b. The call is forwarded to the destination defined under the Call Forward Busy section of
the calling party.
c. The call is forwarded to the destination defined under the Call Forward Busy section of
the original called party.
d. The call is forwarded to the destination defined under the Call Forward No Coverage
section of the calling party.
e. The call is forwarded to the destination defined under the Call Forward No Coverage
section of the original called party.
9. When configuring a CallManager hunt group, which option should you configure first?
a. member DNs
b. line group
c. hunt list
d. hunt pilot
10. By default, how are calls routed through a hunt group?
a. Calls match the hunt pilot DN and hunt through the members listed in the first line
group defined in the hunt list.
b. Calls match the hunt pilot DN and hunt through the members listed in all line groups
defined in the hunt list.
c. Calls match the hunt pilot DN and hunt to the first member listed in the first line group
defined in the hunt list.
d. Calls match the hunt pilot DN and forward to the defined number in the call coverage
area.
following topics:
■ Defining Class of Service as it relates to users

of a phone system
■ Defining partitions and calling search spaces

and identifying the components of each
■ Configuring partitions and calling search

spaces in Cisco CallManager to assign
calling privileges to users and devices
■ Defining time-of-day routing and identifying

its uses and components
■ Configuring time-of-day routing to control

call routing based on the time of day, day of
week, and day of year when the call was
made
CHAPTER 13
Implementing Telephony Call
Restrictions and Control
An important dial plan consideration is the assignment and enforcement of calling privileges.
You might want company executives to have different calling privileges than receptionists, or
employee phones to have different calling privileges than lobby phones. You might also want to
place further restrictions on the times that these privileges are available, for example, by
allowing international calls to be placed only during business hours to prevent unauthorized use
after hours. Partitions, calling search spaces, and time-of-day routing are the primary class of
control components that enable you to assign and enforce calling privileges.
This chapter discusses how to deploy calling restrictions and control by using Cisco
CallManager partitions, calling search spaces, and time-of-day routing.
Class of Service Overview

Class of Service (CoS) is a collection of calling permissions and restrictions that you assign to
individual users or groups of users. (The terms calling permissions and calling restrictions are
used interchangeably in this chapter.)
NOTE CoS as it applies to telephony networks holds a completely different meaning than
its application in quality of service (QoS). CoS in telephony means the implementation of
calling restrictions, whereas CoS in data networking means the application of a data link layer
marking in a data frame.
Examples of class-of-service permissions are as follows:
■ One class of users can place toll calls during normal business hours, but not at other times.
■ Lobby phone users can dial the local emergency numbers and campus extensions, but
cannot dial local, long-distance, or international numbers.
■ Receptionists can dial anywhere within the company and all local area codes, but cannot
dial long-distance or international numbers.
■ Executives can call anywhere except 900 area codes.

254 Chapter 13: Implementing Telephony Call Restrictions and Control
Cisco CallManager uses partitions, calling search spaces, and time-of-day routing to implement
class-of-service restrictions.
Partitions and Calling Search Spaces Overview

A partition is a group of directory numbers (DNs) with similar accessibility. A calling search space
defines which partitions are accessible to a particular device. A device can call only the DNs
located in the partitions that are part of its calling search space.
A partition comprises a logical grouping of directory numbers (DNs) and route patterns with
similar reachability characteristics. Items that are placed in partitions include DNs and route
patterns (or anything else that has a directory number). For simplicity, partition names usually
reflect their characteristics, such as “NYLongDistancePT,” “NY911PT,” and so on.
A calling search space is an ordered list of partitions that Cisco CallManager digit analysis looks
at before a telephone call is placed. Calling search spaces then determine the partitions that calling
devices, such as Cisco IP Phones, Cisco IP SoftPhones, and gateways, can reach when attempting
to complete a call. If a device attempts to reach a route pattern or DN that is not contained in one
of the partitions in its calling search space, it receives a fast busy signal (or a prerecorded message
played by the annunciator service).
Items that can be placed in partitions all have a dialable pattern, which includes phone lines, route
patterns, translation patterns, computer telephony integration (CTI) route group lines, CTI port
lines, voice-mail ports, and Meet-Me conference numbers. In short, you can place absolutely
anything that has a directory number into a partition. To do this, you group common numbers
together. For example, you could create an internal number partition (perhaps named
INTERNAL_PT) that contains all numbers internal to the organization, an emergency partition
(named EMERGENCY_PT) that contains route patterns representing emergency numbers such as
911, and a local PSTN partition (named LOCAL_PT) that contains route patterns representing
local PSTN numbers. By doing this, you have grouped the numbers to assign the calling
restrictions. By default, everything belongs to the <NONE> partition (or no partition assignment),
which is why there are no calling restrictions.
Conversely, you can assign a calling search space to all devices capable of dialing a call, such as
telephones, telephone lines, gateways, and applications (via their CTI route groups or voice-mail
ports). The calling search space contains nothing more than an ordered list of partitions. When you
assign a calling search space to a device, that device can reach whatever is in the partitions listed
in the calling search space. For example, you could create a calling search space named
INT_EMG_CSS that contains just the internal and emergency partitions (INTERNAL_PT and
EMERGENCY_PT mentioned in the previous paragraph). By assigning this to an IP Phone, you

will have restricted it to only dialing internal extensions and emergency numbers.
NOTE Regardless of the calling search space assigned, all devices are able to reach any
number assigned to the <NONE> partition. Because of this, Cisco recommends that you
should never leave numbers assigned to the <NONE> partition.
As shown in Figure 13-1, the DNs of lobby phones and break room phones are placed into
partition A. Partitions B, C, D, E, and F contain the route patterns to reach local numbers, long-
distance numbers, international numbers, and company extensions.
Figure 13-1 Partition and Calling Search Space Example
Local Area Codes Partition B
Long-Distance
Partition C
Partition Area Codes
A
Lobby Phones
Break Room Phones International Numbers Partition D
Company Employee
Partition E
Extensions
Local Emergency
Partition F
Numbers
The calling search space for the lobby phones includes only the partitions containing the company
extensions and the local emergency numbers. Therefore, a lobby phone located in partition A has
a calling search space containing partition A, E, and F and can dial only company destinations,
including the break room (in other words, devices belonging to the same partition) and local
emergency numbers (for example, 911 in the United States).
NOTE In addition to assigning calling search spaces to the phone or directory number, you
can assign calling search spaces to the call forwarding fields of an IP Phone (call forward busy,
call forward no answer, and so on). This allows you to place different calling restrictions
beyond normal calling when a user forwards their phone.
Calling Search Spaces Applied to Gateways and Intercluster Trunks

In addition to applying calling search spaces to the IP Phones in the network, you must also apply
them to the gateways and intercluster trunks. By default, the gateways and intercluster trunks of
the network are assigned a calling search space of <NONE>. This calling search space can only
access the directory numbers and route patterns assigned to the <NONE> partition.
TIP The default state of ALL devices in the CallManager cluster is the following:
Partition: <NONE>
Calling Search Space: <NONE>
Even though it might appear as though these mean “no assignment,” there is an
unconfigurable partition and calling search space called <NONE>. The <NONE> calling
search space only has access to the devices in the <NONE> partition. As you begin to assign
directory numbers and route patterns to partitions other than <NONE>, any devices with the
<NONE> calling search space (default) will lose access to the reassigned numbers.
The previously mentioned tip warrants special consideration when dealing with gateways and
intercluster trunks. Because these devices are assigned the default calling search space of
<NONE>, they will begin to lose access to the cluster directory numbers as you assign them to
other partitions. The directory numbers that the gateways and intercluster trunks have access to
affect inbound calls to the cluster.
For example, you could create a calling search space named INT_LOC_CSS that only allowed
access to directory numbers assigned to the Internal (INTERNAL_PT) and Local PSTN
(LOCAL_PT) partitions. If this INT_LOCAL_CSS calling search space were applied to an
Intercluster Trunk to a remote CallManager cluster, any calls coming inbound from the remote
cluster would only be able to reach the internal and local PSTN extensions. This feature is useful
when you do not have administrative control of the remote cluster but have the need to configure
call restrictions into your cluster from the remote devices.
NOTE This feature works identically for gateway devices configured in the CallManager
cluster. The calling search space applied to gateway devices affect calls coming inbound (into
the cluster) from the gateway. It does not affect calls outbound to the gateway from the devices
in the cluster.
Partition Configuration
To configure partitions, choose Route Plan > Class of Control > Partition. When the Find and
List Partition window appears, click the Add a New Partition link. The Partition Configuration
window that is shown in Figure 13-2 appears. From here, you can add a maximum of 75 partitions
at a time using the following syntax:
<partitionName>, <description>
Figure 13-2 Partition Configuration Window
NOTE Class of Control is a new submenu item under the Route Plan menu in Release 4.1
of Cisco CallManager. The Class of Control submenu includes partitions, calling search
spaces, and time-of-day routing items.
Cisco CallManager Administration requires that you enter the partition name. However, adding a
description for the partition can be useful for documentation purposes.
Using partitions allows you to group DNs with similar characteristics together. For example, the
network administrator at ABC Company must allow certain individuals to call the DNs in the 1xxx
and 2xxx ranges. Before assigning the DNs to a partition, the administrator must first configure the
partitions using the Partition Configuration window in Cisco CallManager Administration. The
administrator could add the necessary partitions using the following format:
■ DN_1XXX, Directory Numbers 1000-1999
■ DN_2XXX, Directory Numbers 2000-2999
After the administrator adds the partitions, DNs must be assigned. To do this, the administrator
must enter the configuration mode of the telephones or route patterns that have the DNs, proceed
to the Directory Number Configuration window, and choose the newly created partition from the
menu, as shown in Figure 13-3.
Figure 13-3 Assigning a Directory Number to a Partition
Calling Search Space Configuration

After configuring the partitions, you can configure the calling search spaces that contain a
prioritized list of the available partitions. Figure 13-4 shows the Calling Search Space
Configuration window in Cisco CallManager Administration (choose Route Plan > Class of
Control > Calling Search Space).
You can use the arrows between the Available Partitions and Selected Partitions panes to choose
the partitions that you want to add to the calling search space. You can reorder partitions by using
the arrows to the right of the Selected Partitions panes. To reduce call-processing time, place the
partitions with the most frequently used numbers at the top of the Selected Partitions list with the
exception of critical partitions. Emergency number partitions or partitions containing blocked
numbers are usually placed near the top of the list.
Figure 13-4 Calling Search Space Configuration Window
The calling search space gives you the ability to place any number of calling restrictions on the IP
telephony network. For example, to restrict access to the 1XXX and 2XXX partitions, the network
administrator from the ABC Company must create a calling search space that is named
“1XXX_Only_CSS” and add the DN_1XXX partition to this calling search space. The
administrator then creates another calling search space named “2XXX_Only_CSS” and adds the
DN_2XXX partition to this calling search space. Finally, the administrator creates a calling search
space named “All_DNs_CSS” and adds both the DN_1XXX and DN_2XXX partitions to this
calling search space.
After configuring the calling search spaces, the administrator assigns them to the various network
devices. For example, telephone A is assigned the 1XXX_Only_CSS calling search space, which
means that telephone A can call only the DNs that are assigned to the partition DN_1XXX. By
assigning telephone B to the 2XXX_Only_CSS calling search space, the administrator restricts
telephone B from calling any numbers outside of the DN_2XXX partition. If the administrator
assigns telephone C to the All_DNs_CSS calling search space, telephone C can call the DNs that
are assigned to both partitions DN_1XXX and DN_2XXX.
Time-of-Day Routing Overview

Time-of-day routing, introduced in Release 4.1 of Cisco CallManager, routes calls to different
locations based on the time of day when a call is made. For example, during business hours, calls
can route to a pilot number (linked to a hunt group), and after hours, calls can go directly to a
voice-messaging system. Time-of-day routing also provides the ability to route calls based on the
time-of-day or based on a specific day, such as December 25. The following are practical
applications of how you could use time-of-day calling restrictions:
■ Allow international calls only during office hours.
■ Block international calls based on local caller’s time zone.
■ Block international calls on holidays.
■ Allow international calls only during office hours of the caller’s time zone.
■ Implement a least-cost routing system by choosing the cheapest provider for a specific time
of day.
■ Route calls after hours to a home number or voice mail.
■ Any application where you want to control the calling search space based on the time of the
day.
Time-of-day settings are assigned to partitions. After the administrator has configured the time-of-
day settings and assigned them to partitions, the time-of-day feature filters the calling search space
through time-of-day settings that are defined for each partition in the calling search space.
The time-of-day feature is applied when a called number is validated. Cisco CallManager filters
the partitions contained in the calling search space of the originating device, and the called number
is validated against this filtered calling search space. Keep in mind that you can assign a calling
search space to a voice gateway. This could, as an example, allow you to reroute all incoming calls
from the PSTN to the IP telephony network based on time-of-day restrictions.
To implement time-of-day routing in Cisco CallManager, you must understand the concept of time
periods and time schedules.
Time Periods
A time period comprises a start time and end time. The available start times and end times are 15-
minute intervals on a 24-hour clock, from 00:00 to 24:00. In addition, a time period requires
definition of a repetition interval. Repetition intervals are the days of the week (for example,
Monday through Friday) or a day of the calendar year (for example, June 9).
Time-of-Day Routing Overview 261
The following are examples of four time periods:
■ Time period weekdayhrs_TP as 08:00 to 17:00 from Monday to Friday.
■ Time period weekendhrs_TP as 08:00 to 17:00 on Saturday and Sunday.
■ Time period newyears_TP as 00:00 to 24:00 on January 1.
■ Time period noofficehours_TP as no hours on Saturday and Sunday. For this time period, the
associated partition is not active on Saturday and Sunday.
Time Schedule
A time schedule consists of a group of defined time periods that the administrator associates with
a partition.
After the administrator selects a time period for association with a time schedule, the time period
remains available for association with other time schedules.
Figure 13-5 shows an example of a time period RegEmployees_TS with time periods
weekdayhrs_TP, newyears_TP and noofficehours_TP associated with it.
Figure 13-5 Time Schedule Example

Time Periods Start–End Repetition
weekdayhrs_TP 08:00 – 17:00 M–F

weekendhrs_TP 08:00 – 17:00 Sat – Sun
newyears_TP 00:00 – 24:00 January 1
noofficehours_TP Sat – Sun
Time Schedule Time Periods

weekdayhrs_TP
RegEmployees_TS newyears_TP
noofficehours_TP
Partition Time Schedule
CiscoAustin_PT RegEmployees_TP
The administrator associates time schedules with a partition. Partitions contained within the calling
search space are available based on time-of-day settings. Cisco CallManager filters the calling search
space through the time-of-day settings defined for each of the partitions in the calling search space.
Time-of-Day Routing Effect on Users

If time-of-day routing is enforced, users cannot set certain Call Forward All (CFA) numbers at
certain times. For example, the user A calling search space for forwarding includes a time-of-day-
configured partition that allows international calls from 8:00 to 17:00 (5:00 p.m.). User A wants
to configure his CFA number to an international number. User A can set this number only during
the 8:00 to 17:00 time period because, outside these hours, the system does not find the
international number in the partition that is used to validate the CFA number.
If the user sets the CFA during office hours when it is allowed, and the user receives a call outside
office hours, the caller hears a fast busy signal.
In addition, users cannot reach DNs in some partitions that are configured for time-of-day routing
and that are not active during the time of the call, depending upon the configuration of partitions.
Users will not be able to reach the route or translation patterns in partitions configured with time-
of-day routing that are not active at the time of the call.
Configuring Time-of-Day Routing

To access the time-of-day routing configuration components, choose Route Plan > Class of
Control > Time Period or Route Plan > Class of Control > Time Schedule. Figure 13-6
illustrates these options.
Figure 13-6 Time-of-Day Routing Configuration

Configuring Time-of-Day Routing 263
To configure time-of-day routing, follow these steps:
Step 1 Create the individual time periods.

Step 2 Create the time schedule and assign the time periods to it.
Step 3 Assign the time schedule to a partition.
Configuring Time Periods

To add and configure a time period, follow this procedure:
Step 1 In the menu bar, choose Route Plan > Class of Control > Time Period.
Step 2 Click Add a New Time Period.
Step 3 Enter the time period name, start time, end time, and repetition interval.
Step 4 To add the new time period, click Insert.
The message “Status: Insert completed” appears.
Step 5 To add more time periods, click Add a New Time Period and repeat this
procedure.
Figure 13-7 shows an example of a time period named OfficeHoursTP, a start time of 08:00, an
end time of 17:00, and a weekly repetition interval of every Monday through Friday.
Figure 13-7 Time Period Configuration Window

Table 13-1 provides a list of options available to you when implementing time periods.
Table 13-1 Time Period Options
Parameter Description
Time Period Name Enter a name in the Time Period Name field. The name can include up to 50
alphanumeric characters and can contain any combination of spaces, periods
(.), hyphens (-), and underscore characters (_). Ensure that each time period
name is unique to the plan.
Use concise and descriptive names for your time periods. The hours_or_days
format usually provides a sufficient level of detail and is short enough to
enable you to quickly and easily identify a time period. For example,
office_M_to_F identifies a time period for the business hours of an office
from Monday to Friday.
Start Time Time when this time period starts. The available start times are 15-minute
intervals throughout a 24-hour day. The default value is No Office Hours.
No Office Hours means that the selected partition will not be active for the
defined day of year or days of week.
To start a time period at midnight, choose the 0:00 value.

End Time Time when this time period ends. The available end times are 15-minute
intervals throughout a 24-hour day. The default value is No Office Hours. To
end a time period at midnight, choose the 24:00 value.
Repeat Every Click one of the radio buttons to set a schedule:
• Week From—If you click the Week From radio button, use the drop-
down menus next to From and Through to choose the days of the week
during which this time period applies.
Examples: Choose a From value of Mon(day) and a Through value of
Fri(day) to define a time period that applies from Monday to Friday. Choose
a From value of Sat(urday) and a Through value of Sat(urday) to define a
time period that applies only on Saturdays.
• Year On—If you click the Year On radio button, use the drop-down
menus to choose the month and day of the year on which this time period
applies.
Example: Choose the month Jan(uary) and the day 1 to define a time period
that applies yearly on New Year’s Day.
Figure 13-8 shows how to create a time period using the No Office Hours time interval and a
repetition interval with a specific day of the year defined. When the No Office Hours time interval
is selected, the associated partition is not active for the defined days of the week or day of the
year—in this example, December 25.
Figure 13-8 Creating a Christmas Time Period
Use the No Office Hours time interval and the Year On repetition interval for days, such as
Christmas, New Year’s Day, and national holidays, when for example, the company is closed,
certain departments are closed, or certain employees are on holiday.
Configuring Time Schedules

After the time periods are created, you can create the time schedule:
Step 1 In the menu bar, choose Route Plan > Class of Control > Time Schedule.
Step 2 Click Add a New Time Schedule.
Step 3 Name the time schedule in the Time Schedule Name field.
Step 4 Choose the desired time periods from the Available Time Periods pane,
shown in Figure 13-9, and use the down arrow to move them to the Selected
Time Periods pane. Move any time periods you do not want in the Selected
Time Periods pane to the Available Time Periods pane.
Figure 13-9 Creating a Time Schedule
Step 5 To add the new time schedule, click Insert (or Update if the time schedule
already exists and you are changing it).
The message “Status: Insert completed” appears.
Step 6 To add more time schedules, click Add a New Time Schedule and repeat
this procedure.
Applying Time Schedules

Associating the time schedule (collection of time periods) to a partition is the final step in
configuring time-of-day routing. A time schedule is not activated until it is assigned to a partition.
In Cisco IOS software, this process is similar to creating an access control list (ACL) that does not
become activated until it is assigned to an interface.
TIP Cisco CallManager allows you to create duplicate route patterns as long as they are
assigned to different partitions. This grants the flexibility to create a route pattern and assign
it to a partition for which you have assigned specific time-of-day restrictions for some users.
You can then create an identical route pattern and assign it to a partition without time-of-day
calling restrictions for other users. You can control user access privileges through the partition
you assign to their calling search space.
You can create a new partition from the Partition Configuration window and assign the time
schedule to it or assign the time schedule to an existing partition, as shown in Figure 13-10. The
process is the same:
Step 1 From the Time Schedule drop-down menu, choose a time schedule to
associate with this partition.
Step 2 Choose either the time zone of the originating device or any specific time
zone for a time schedule. The system checks the chosen time zone against
the time schedule when the call is placed to directory numbers in this
partition.
Figure 13-10 Assigning a Time Schedule to a Partition

Table 13-2 provides a list of available options when applying a time schedule to a partition.
Table 13-2 Time Schedule Application Options
Time From the drop-down menu, choose a time schedule to associate with this
Schedule partition. The associated time schedule specifies when the partition is available to
receive incoming calls.
The default value specifies None, which means that time-of-day routing is not in
effect and the partition remains active at all times.
In combination with the time zone value in the following field, associating a
partition with a time schedule configures the partition for time-of-day routing.
The system checks incoming calls to this partition against the specified time
schedule.
Time Zone Choose one of the following options to associate a partition with a time zone:
• Originating Device—If you choose this option, the system checks the
partition against the associated time schedule with the time zone of the calling
device.
• Specific Time Zone—If you choose this option, choose a time zone from the
drop-down menu. The system checks the partition against the associated time
schedule at the time that is specified in this time zone.
These options all specify the time zone. When there is an incoming call, the
current time on the Cisco CallManager is converted into the specific time zone
that you set. This specific time is validated against the value in the Time
Schedule field.
Time-of-Day Routing Usage Scenario

Figure 13-11 shows an example of how time-of-day routing can be configured to route calls after
office hours to a home number.
Time-of-Day Routing Usage Scenario 269
Figure 13-11 Time-of-Day Routing Usage Scenario

A A dials 2000
Cisco CallManager
B
Current Time: 10:00
Current Day: Wed. extends call to
2000/CiscoSJ
IP A dials 2000 IP
Current Time: 20:00
1000/CiscoSJ Current Day: Wed. 2000/CiscoSJ
CSS1: CiscoSJ, Cisco CallManager
CiscoOutside extends call to
2000/CiscoOutside
2000/CiscoOutside
CFA: 9725550111
(Home Number)
Partition CiscoSJ: Partition CiscoOutside:

• Time Schedule TS1 • Time Schedule TS2
• Specific Time Zone (EST) • Specific Time Zone (EST)
Time Schedule TS1: Time Period TP1 Time Schedule TS2: Time Period TP2, TP3
TP1: 08:00–17:00 Mon–Fri TP2: 17:00–24:00 Mon–Fri
TP3: 00:00–08:00 Tue–Sat
The setup is as follows:
■ Partition CiscoSJ is configured with the time schedule TS1 and the specific time zone Eastern
Standard Time (EST).
— Time period TP1 is associated with time schedule TS1.

— The TP1 start time is 08:00, the end time is 17:00, and the repetition interval is
Monday through Friday.
■ Partition CiscoOutside is configured to use time schedule TS2 and the specific time zone EST.
— Time periods TP2 and TP3 are associated with time schedule TS2.
Monday through Friday.
Tuesday through Saturday.
■ The user A calling search space is CSS1. CSS1 contains the CiscoSJ and CiscoOutside
partitions.
Call flow is as follows:
■ User A at extension 1000 calls 2000 at 10:00 a.m. Wednesday. The partition CiscoSJ is active
at that time (TP1), so the call is forwarded to 2000.
■ User A at extension 1000 calls 2000 at 20:00 (8:00 p.m.) Wednesday. The partition CiscoSJ
is not active at that time. However, the partition CiscoOutside is active then (TP2), and the
calling search space CSS1 contains a pattern to reach the user B home phone number
9725550111, which enables user B to set CFA to forward incoming calls to a home number.
Summary
This chapter covered the concepts and configuration of Cisco CallManager calling restrictions.
Class of Service (CoS) is necessary in nearly every corporate environment to implement telephony
policies for end users. Cisco CallManager implements CoS through partitions and calling search
spaces. A partition is a group of DNs with similar accessibility characteristics, whereas a calling
search space defines which partitions are accessible to a particular device (such as an IP Phone or
gateway).
To configure partitions and calling search spaces, create the partition and assign the appropriate
DNs to the partition. You must then create a calling search space containing the partitions you want
to be reachable and apply the calling search space to the affected IP Phones or gateway devices.
Time-of-day routing routes calls to different locations based on the time of day or day of the year
when a call is made. To configure time-of-day routing, create the necessary time periods, create a
time schedule containing similar time periods, and assign the time schedule and time zone to a
partition.
Review Questions
1. The Partition Configuration window in Cisco CallManager Administration contains which

two parameters associated with time-of-day routing? (Choose two.)
a. start and end times
b. repetition interval
c. time zone
d. time period
e. time schedule
2. Which time period would be configured to allow access to user A during user A’s weekday
office hours (assuming user A works from 8:00 a.m. to 5:00 p.m.)?
a. 08:00 to 17:00 Monday to Friday and the time zone of the originating device
b. 08:00 to 17:00 Monday to Friday and the time zone of user A
c. no office hours Saturday and Sunday and the time zone of the originating device
d. no office hours Saturday and Sunday and the time zone of user A
3. Which of the following items can be assigned to a partition? (Choose three.)
a. phones
b. directory numbers
c. gateways
d. route patterns
e. translation patterns
4. Describe the order of Class of Service configuration.
a. DNs are placed in partitions, partitions are placed in calling search spaces, and calling
search spaces are assigned to the calling device.
b. DNs are placed in calling search spaces, calling search spaces are placed in partitions,
and partitions are assigned to the calling device.
c. DNs are placed in calling search spaces, calling search spaces are placed in partitions,
and calling search spaces are assigned to the calling device.
d. IP Phones and gateways are placed in partitions, partitions are placed in calling search
spaces, and both the partition and calling search spaces are assigned to the calling
device.
5. To which of the following devices can you assign a calling search space? (Choose three.)
a. gateways
b. IP Phones
c. directory numbers (phone lines)
d. route patterns
6. How does CallManager process a call from a device configured with a calling search space?
a. It processes the call by looking through the partitions in the calling search space from
the bottom of the list to the top.
b. It processes the call by looking through the partitions in the calling search space from
the top of the list to the bottom.
c. It processes the call by first looking through the partitions in the calling search space of
the called device followed by the partitions in the calling search space of the calling
device.
d. It processes the call by first looking through the partitions in the calling search space of
the calling device followed by the partitions in the calling search space of the called
device.
7. When configuring time-of-day routing, which of the following should be created first?
a. time schedules
b. time agendas
c. time routes
d. time periods
8. You want to implement time-of-day routing restrictions. One group of users should only be
able to call PSTN numbers from 8:00 a.m. to 5:00 p.m. Another group of users should be able
to call the PSTN anytime. How could this be configured?
a. You must create two sets of PSTN route patterns. One set should be added to the parti-
tion restricted by the 8:00 a.m. to 5:00 p.m. time schedule. The other set should be
added to a partition not restricted by a time schedule. A calling search space containing
the restricted partition is then assigned to the users limited to 8:00 a.m. to 5:00 p.m.
PSTN calling and a calling search space containing the unrestricted partition is assigned
to the users able to reach the PSTN anytime.
b. You must create one set of PSTN route patterns. This set is added to a partition restricted
by the 8:00 a.m. to 5:00 p.m. time schedule and a partition not restricted by a time
schedule. A calling search space containing the restricted partition is then assigned to
the users limited to 8:00 a.m. to 5:00 p.m. PSTN calling and a calling search space con-
taining the unrestricted partition is assigned to the users able to reach the PSTN any-
time.
c. You must create one set of PSTN route patterns. This set is added to a partition restricted
by the 8:00 a.m. to 5:00 p.m. time schedule. A calling search space containing the
restricted partition is then assigned to the users limited to 8:00 a.m. to 5:00 p.m. PSTN
calling and a calling search space that does not contain the restricted PSTN partition is
assigned to the users able to reach the PSTN anytime.
d. This functionality is not currently supported in Cisco CallManager 4.x versions.
9. How is time-of-day routing applied to an end user?

a. Restrictions are applied to the user’s partition.
b. Restrictions are applied directly to the user’s DN.
c. Restrictions are applied to a partition contained in a user’s calling search space.
d. Restrictions are applied to a user’s calling search space.
10. You want to create a time period that matches the Christmas holiday every year. What time
period selection criteria would you use to accomplish this?
a. Repeat Every Day On…
b. Repeat Every Week From…
c. Repeat Every Year From…
d. Repeat Every Year On…
following topics:
■ Describing why call admission control is important

to maintain voice QoS across an IP WAN
■ Describing how the locations feature in Cisco
CallManager provides call admission control for
centralized call-processing environments
■ Configuring locations-based call admission control
in a centralized call-processing deployment to limit
the number of active calls and prevent
oversubscribing the bandwidth on the IP WAN links
■ Explaining what AAR is, how it works, and the
requirements for configuring Cisco CallManager for
AAR
■ Describing how a gatekeeper can reduce the number
of intercluster trunks that are required in a distributed
call-processing environment
■ Identifying the communication procedures between
an H.323 gatekeeper and H.323 endpoint, including
discovery, registration, admission, and bandwidth
requests
■ Configuring gatekeeper-based call admission control
in a distributed call-processing deployment to limit
the number of active calls and prevent
oversubscribing the bandwidth on the IP WAN links
■ Describing how SRST provides Cisco CallManager
failover capabilities
■ Configuring SRST on a supported gateway and in
Cisco CallManager so that the SRST router assumes
call-processing duties should the WAN link fail
CHAPTER 14
Implementing Multiple-Site
Deployments
Implementing multiple-site IP telephony deployments over an IP WAN requires additional

planning to ensure the quality and availability of voice calls. When an IP WAN connects IP
telephony clusters, a mechanism must exist to control the audio quality and video quality of calls
over the IP WAN link by limiting the number of calls that are allowed on that link at the same
time. Call admission control is the mechanism that ensures that voice calls do not oversubscribe
the IP WAN bandwidth and affect voice quality.
When the priority queue of IP WAN bandwidth is consumed, a mechanism must exist to
automatically reroute calls over the public switched telephone network (PSTN) without
requiring the caller to hang up and redial the called party. Automated Alternate Routing (AAR)
is a Cisco CallManager feature that automatically reroutes calls through the PSTN or other
networks when priority bandwidth is insufficient in a centralized call-processing deployment.
NOTE Referring to the priority queue of IP WAN bandwidth does not necessarily
encompass the entire amount of WAN bandwidth available. Rather, it encompasses the
amount of bandwidth you have set aside for high-priority traffic (including voice).
If connectivity with Cisco CallManager is lost, Cisco IP Phones become unusable for the
duration of the failure. Cisco Survivable Remote Site Telephony (SRST) overcomes this
problem and ensures that the Cisco IP Phones offer continuous service by providing call
handling support for Cisco IP Phones directly from the Cisco SRST router.
This chapter describes the operation and configuration of call admission control, AAR, and
SRST.
Call Admission Control

Call admission control (CAC) provides you with mechanisms to control the volume of calls
between two endpoints. Controlling the number of calls, or the amount of bandwidth that is
required between two endpoints, is key to maintaining quality of service (QoS) for all existing
and new calls. As shown in Figure 14-1, a WAN might only have enough bandwidth provisioned
276 Chapter 14: Implementing Multiple-Site Deployments
to support two calls. If the CallManager were to send a third call over the IP WAN, all calls
experience poor quality.
Figure 14-1 Call Admission Control Example

Cisco CallManager Cisco CallManager
IP IP
Call No. 1
Call No. 2
IP WAN
IP IP
Call No. 3
IP IP
You can provision the network to carry a specific amount of real-time traffic. Any traffic that
exceeds the provisioned bandwidth will subject all real-time traffic to delay, jitter, and possibly
packet loss. The coder-decoder (codec) used for the call determines the bandwidth calculations
used with call admission control.
Using Cisco CallManager, the following two types of call admission are possible:
■ Locations-based call admission control—The locations feature of Cisco CallManager

provides a simplified call admission control scheme for centralized call-processing systems.
A centralized system uses a single Cisco CallManager cluster to control all of the locations.
■ Gatekeeper-based call admission control—A gatekeeper device provides call admission

control for distributed call-processing systems. In a distributed system, each site contains its
own call-processing capability.
Locations-Based Call Admission Control

Cisco CallManager provides a simple locations-based call admission control mechanism for hub-
and-spoke topologies, used primarily for centralized call processing.
The locations feature in Cisco CallManager lets you specify the maximum amount of audio
bandwidth (for audio calls) and video bandwidth (for video calls) that is available for calls to and
from each location. This specification limits the number of active calls and limits oversubscription
of the priority bandwidth on the IP WAN links.
NOTE Video calls are discussed in further detail in Chapter 28, “Introducing IP Video
Telephony.”
For the purpose of calculating bandwidth for call admission control, Cisco CallManager assumes
that each call stream consumes the following amount of bandwidth:
■ A G.711 call uses 80 kbps.
■ A Global System for Mobile Communication (GSM) call uses 29 kbps.
■ A wideband call uses 272 kbps.
Locations work in conjunction with regions to define the characteristics of a network link.
Locations define the amount of available bandwidth for the link, and regions define the type of
compression (G.711, G.723, or G.729) that is used on the link.
For example, in Figure 14-2, three locations are specified: San Jose (unlimited bandwidth), New
York (256 kbps), and Dallas (64 kbps). Cisco CallManager continues to admit new calls to a link
as long as sufficient bandwidth is still available. Thus, if the link to the New York location in this
example has 256 kbps of available bandwidth, that link can support three G.711 calls at 80 kbps
each and ten G.723 or G.729 calls at 24 kbps each. If any additional calls are received that would
exceed the bandwidth limit, the system rejects them, the calling party receives a reorder tone, and
a text message is displayed on the phone.
Figure 14-2 Location-Based Call Admission Control Example
IP
Cisco V IP
CallManager
Cluster Location Bandwidth (kbps)
San Jose Unlimited New York (Remote)
New York 256

Dallas 64
IP
IP WAN
V V IP
IP IP Dallas (Remote)
San Jose (Main)
Configuring Location-Based Call Admission Control

Configuring location-based call admission control in Cisco CallManager is a relatively simple
process. Before you can accurately configure locations, you must ensure you have configured all
the devices at remote locations to use a common codec when calling across the WAN. You can
accomplish this through the region configuration assigned to the device pool of the remote phones.
After you have accurately assigned the codec, follow these steps to configure locations in Cisco
CallManager:
Step 1 Choose System > Location to configure a separate location for each IP
WAN link to which you want to apply call admission control. As shown in
Figure 14-3, allocate the maximum available bandwidth for calls across the
link to that location.
Step 2 In the individual device configuration windows, configure the telephones
and other devices and assign each of them to the appropriate device pool
and location, as shown in Figure 14-4.
Figure 14-3 Configuring Locations
Figure 14-4 Assigning Devices to a Location

AAR Overview
Automated Alternate Routing (AAR) is a mechanism used in conjunction with location-based call
admission control. AAR allows calls to be rerouted through the PSTN by using an alternate
number when Cisco CallManager blocks a call due to insufficient location bandwidth. With AAR,
the caller does not need to hang up and redial the called party. If AAR is not configured, the user
gets a reorder tone and the IP Phone displays “Not enough bandwidth.”
AAR applies to centralized call-processing deployments. For instance, if a telephone in a company

headquarters calls a telephone in branch B (shown in Figure 14-5) and the available bandwidth for
the WAN link between the branches is insufficient (as computed by the locations mechanism),
AAR can reroute the call through the PSTN using the following process:
1. Phone A (DN = 2345) dials On-Net to phone B (DN = 1234).

2. Call admission control denies the call.
3. The external phone number mask 215-555-XXXX is added. The result is a fully qualified
number, 215-555-1234.
NOTE When working with the external phone number mask, it is important to keep straight
which mask the CallManager will use when invoking AAR. When AAR redirects a call across
the PSTN, it will always use the destination IP Phone’s external phone number mask to
transform the called number. The source IP Phone’s external phone number mask will be used
to transform the caller ID information.
4. The prefix used to access the PSTN gateway is 9, and 1 is added to the string for numbering
plan compatibility. These values are configured in the AAR group.
5. 9-1-215-555-1234 is the new alternate dial string.
6. The CallManager passes the new dial string through the route plan and sends the call across
the PSTN.
AAR is transparent to users. It is configured so that users dial only the on-net (for example, four-
digit) directory number (DN) of the called phone and no additional user input is required to reach
the destination through the alternate network (such as the PSTN).
Figure 14-5 AAR Scenario
AAR:
1-215-555-1234 IP
PSTN V IP
Branch A
Phone B
1234
IP
WAN
V CAC V IP
Branch B
IP IP IP
Headquarters Phone A
2345
Keep the following requirements and caveats in mind when applying AAR:
■ AAR requires call admission control based on locations and regions; AAR will function only
if the WAN bandwidth is consumed.
■ The IP Phone that originates a call should be in the same location as the gateway that routes
the call to the PSTN.
■ The called IP Phone and the gateway that terminates the call from the PSTN should be in the
same location.
■ If a call originates from a device in one location and terminates to a non-IP Phone in another
location, AAR will not function. A voice-mail port on another vendor device also does not use
AAR.
■ If a call originates from a Cisco computer telephony integration (CTI) route point, AAR will
not function.
■ AAR does not work with SRST, which means that AAR will not function in the case of a
WAN failure.
Configuring AAR
Configuring Cisco CallManager for AAR requires the following steps:
Step 1 Configure the locations. Use locations to implement call admission control
in a centralized call-processing system. Cisco CallManager requires that
the locations be arranged in a hub-and-spoke topology.
Step 2 Configure the regions. Use regions to specify the type of compression and
the amount of bandwidth used within a region and between regions.
Step 3 Assign devices to a region and location. Use device pools to define sets of
common characteristics—in this case, the region setting—for devices in a
specific location.
Step 4 Enable AAR clusterwide. Ensure that the Automated Alternate Routing
Enable service parameter is set to True.
Step 5 Configure AAR groups. For each AAR group, enter the prefix digits that are
used for AAR within the AAR group, in addition to the prefix digits used
for AAR between a given AAR group and other AAR groups. Devices such
as gateways, telephones (by means of DNs), and trunks associate with AAR
groups.
Step 6 Configure the calling search space for AAR. Cisco CallManager uses the
AAR calling search space to reroute calls when there is no available
bandwidth on the WAN.
Step 7 Configure the endpoints as follows:
• Assign devices to the AAR group.
• Assign the external line mask to the DN.
• Assign devices to the AAR calling search space.
Steps 1 through 3 involve the location configuration, which was discussed earlier in this chapter.
The AAR configuration steps starting with Step 4 are covered in detail in the following subtopics.
Enabling AAR in the CallManager Cluster

To access the settings for AAR throughout the cluster, open the Service > Service Parameters
window for the Cisco CallManager service. You can use the Microsoft Internet Explorer Find
feature to quickly locate the Automated Alternate Routing Enable service parameter, shown in
Figure 14-6. Ensure this value is set to True. The default value for this service parameter is False.
Figure 14-6 Automated Alternate Routing Service Parameters
In addition, you can access the Service Parameters > CallManager > Clusterwide Parameters
(Device–Phone) window (shown in Figure 14-7) to modify the two fields used to configure the
messages that appear on the end user’s IP Phone if there is no WAN bandwidth available or if AAR
reroutes the call over the WAN.
Figure 14-7 Automated Alternate Routing End-User Notification Messages
AAR Group Configuration

If a user dials an internal number that goes to another location and the WAN is busy, the call should
be rerouted over the PSTN. To place a call over the PSTN, Cisco CallManager needs the fully
qualified number, not the internal DN, to reroute the call.
The destination number might require a prefix for an Off-Net access code (for example, 9) to be
routed properly by the dial plan of the origination branch. Furthermore, if the point of origin is
located in a different Numbering Plan Area (NPA, or area code), the network might require a prefix
of 1 as part of the dialed string. Cisco CallManager uses the internal DN, the external phone
number mask of the called DN, and the prefix to determine the alternate number for routing the
call over the PSTN.
When configuring AAR, place the DNs in AAR groups. As shown in Figure 14-8, for each pair of
AAR groups, you can configure prefix digits to add to the DNs for calls between the two groups,
including prefix digits for calls originating and terminating within the same AAR group. For
example, if a user in the California_AAR group dials a user in the Arizona_AAR group (shown in
Figure 14-8) and the IP WAN connection bandwidth was oversubscribed, the CallManager would
redirect the call over the PSTN connection, prefixing the digits “91” to the DN (which is also
combined with the external phone number mask, discussed in the upcoming section).
Figure 14-8 Automated Alternate Routing Group Configuration Window
As a general rule, place DNs in the same AAR group if they share all of the following
characteristics:
■ A common off-net access code (for example, 9)
■ A common PSTN dialing structure for interarea calls (for example, 1-NPA-Nxx-xxxx in North
America)
■ A common external phone number mask format
For example, assume that both the San Francisco and New York sites share all of the preceding
characteristics. The DNs for San Francisco and New York can be placed into a single AAR group,
and the group can be configured such that AAR calls placed within this AAR group are prefixed
with 91. For phone A in San Francisco to reach phone B in New York (at 212-555-1212), the AAR
group configuration prefixes 91 to the dialed string, yielding a completed string of 91-212-555-1212.
Complete these steps to add an AAR group:
Step 1 From the menu bar, choose Route Plan > AAR Group.
Step 2 Click Add a New AAR Group.
Step 3 Enter a name in the AAR Group Name field.
Step 4 Click Continue.
Step 5 In the Prefix Digits Within field for the group being added, enter the prefix
digits to use for AAR within this AAR group.
Step 6 In the Prefix Digits Between the group being added and Other AAR Groups
area, complete the following fields:
• Prefix Digits (From)—Enter the prefix digits to use for AAR when
routing a call from this group to a device that belongs to another AAR
group.
NOTE Prefix digits that are entered in this field for the originating AAR
group are also added in the Prefix Digits (To) field of the AAR destination
group.
• Prefix Digits (To)—Enter the prefix digits to use for AAR when you are
routing a call to this group from a device that belongs to another AAR
group.
NOTE Prefix digits entered in this field for the destination AAR group are
also added in the Prefix Digits (From) field of the AAR originating group.
Step 7 Click Insert to add this AAR group.
Configuring Endpoints for AAR

Most of the setup process for AAR occurs when configuring the endpoints and gateways for AAR.
This process can be completed in three steps:
Step 1 Configure the DN to use the AAR group, as shown in Figure 14-9. An AAR
group setting of None specifies that no rerouting of blocked calls will be
attempted.
Figure 14-9 Configuring DNs for AAR Groups
Step 2 Assign the external phone number mask to the DN.

The rerouting of calls requires using a destination DN that is routable
through the alternate network (for example, the PSTN). AAR uses the
dialed digits to establish the on-cluster destination of the call and then
combines them with the external phone number mask of the called party.
The combination of these two elements must yield a fully qualified number
that is routable by the alternate network.
The destination number might require a prefix for an off-net access code
(for example, 9) to be routed properly by the origination branch dial plan.
Furthermore, if the point of origin is located in a different NPA, a prefix of
1 might be required as part of the dialed string. You should add these prefix
digits in the AAR Group Configuration window rather than the external
phone number mask.
For example, assume that phone A in San Francisco (DN = 2345) dials an
on-net DN (1234) configured on phone B in New York. If locations-based
call admission control denies the call, AAR retrieves the external phone
number mask of the New York phone (212-555-XXXX) and uses it to
derive the fully qualified number (212-555-1234) that is routable on the
PSTN.
The PSTN routing of a call from San Francisco to New York requires a 1 as
a prefix to the phone number. It is recommended that this prefix not be
included as part of the external phone number mask because it would be
displayed as part of the calling line ldentification (CLID, or caller ID) for
any calls made by the phones to an off-net destination. Instead, it is
recommended that the 1 be added as part of the AAR group configuration.
Configure the external phone number mask on the IP Phone DN by
choosing Device > Phone, and then selecting the phone and the line you
want to configure. Figure 14-10 illustrates this configuration.
Figure 14-10 Configuring DNs for AAR Groups
Indicate the phone number (or mask) that is used to send caller ID
information when a call is placed from this line. A maximum of 30 numeric
and X characters can be added. The Xs represent the DN and must appear
at the end of the pattern. For example, if you specify a mask of
972813XXXX, an external call from extension 1234 displays a caller ID
number of 9728131234.
TIP Remember, the external phone number mask should always be the number the DN you are
configuring through the PSTN. If your organization does not use Direct Inward Dial (DID)
ranges, you should configure the external phone number mask as the location’s general number,
allowing a receptionist to redirect the call to the internal extension.
Step 3 Configure the phone (or gateway) to use the AAR calling search space.
With AAR, the AAR calling search space was introduced. With this special
parameter, the caller rights can be adjusted for this special event when AAR
is used. Cisco CallManager looks at the AAR calling search space to
determine whether the device has the right to call over the PSTN. For
example, if the originating calling device can reach only internal phones
with the configured calling search space, then it is possible with the AAR
calling search space to allow calls from that device to be rerouted using the
PSTN. In this case, the AAR calling search space grants additional
permissions to route the call when the IP WAN is down.
The AAR calling search space specifies the collection of route partitions
that are searched to determine how to route a collected (originating)
number that is blocked because of insufficient bandwidth. You can assign
the AAR calling search space to the phone from the Phone Configuration
window, as shown in Figure 14-11.
Figure 14-11 Configuring AAR Calling Search Space
Gatekeeper-Based Call Admission Control

Gatekeeper call admission control reduces configuration overhead by eliminating the need to
configure separate individual intercluster trunks between clusters. A gatekeeper can determine the
IP addresses of devices that are registered with it, or you can enter the IP addresses explicitly. It
uses these IP addresses to refer calls between endpoints (an endpoint could be an entire Cisco
CallManager cluster). For example, a call could originate from one CallManager cluster where the
route plan indicates to forward the call to the gatekeeper. The gatekeeper responds with the real IP
address of the CallManager in another cluster that manages the extension the call is attempting to
reach.
If you choose the gatekeeper method of call admission control, you will need to set up an
Intercluster Trunk (gatekeeper-controlled) or H.225 trunk (gatekeeper-controlled). These trunks
are configured to point to the gatekeeper device and are included in the dial plan for your network.
When you configure gatekeeper-controlled trunks, Cisco CallManager automatically creates a

virtual trunk device. The IP address of this device changes dynamically to reflect the IP address of
the remote device as determined by the gatekeeper. For example, if the gatekeeper forwards the
call to another CallManager cluster, the endpoint IP address for the virtual trunk device points to
the CallManager server in the remote cluster.
NOTE Much of the H.323 gatekeeper configuration is covered in Cisco Voice Gateways and
Gatekeepers (ISBN: 1-58705-258-x) from Cisco Press. The gatekeeper configuration discussed
in this book is directly related to the Cisco CallManager integration.
Gatekeeper Communication
For a Cisco CallManager to use an H.323 gatekeeper, a new communication exchange occurs
between the gatekeeper and the Cisco CallManager. The first process that an endpoint must go
through is gatekeeper discovery. An endpoint achieves gatekeeper discovery either manually or
through autodiscovery.
Autodiscovery uses multicast to discover the gatekeeper. A gatekeeper request (GRQ) is multicast,
and any gatekeeper that can accept a registration returns a gatekeeper confirmation (GCF). If a
gatekeeper cannot accept a registration, it returns a gatekeeper reject (GRJ). This exchange is
illustrated in Figure 14-12.
Figure 14-12 Gatekeeper Discovery

GRQ (1)
GCF/GRJ (2)
H.323 Endpoint/Gateway Gatekeeper
NOTE Although the exchange shown in Figure 14-12 is supported on most H.323 endpoints
and gateways (as it is defined in the H.323 standard), Cisco CallManager does not support
gatekeeper discovery messages. Because the gatekeeper can play such an integral role in the
CallManager route plan, H.323 gatekeepers must be manually configured rather than
dynamically discovered.
After the discovery process has completed, the Cisco CallManager must register with the
gatekeeper. A Cisco IOS gatekeeper supports two types of registration:
■ Full registration
■ Lightweight registration
An endpoint must always use a full registration during the initial registration process. An endpoint
can use lightweight registration to maintain registration. Should an endpoint become unregistered
with the gatekeeper, a full registration is required.
Full registration requires the endpoint to register any E.164 address, H.323 identifier, device type,
and other possible parameters each time that it registers. This procedure involves additional
processing load on a gatekeeper every time an endpoint registers or renews its registration. The
registration request (RRQ) includes the time between renewal of registrations or Time to Live
(TTL), and the gatekeeper can replace this value or return this value unchanged.
NOTE You can make the returned TTL value configurable with Cisco IOS Software Release
12.1.5T and later.
The lower the TTL value, the higher the load on the gatekeeper processing the registration. The
impact of a higher value is that it takes longer for the gatekeeper to become aware that an endpoint
has lost connectivity. Use 30 to 300 seconds, depending on design requirements.
When the endpoint sends a full RRQ to the gatekeeper, the gatekeeper responds with a registration
confirmation (RCF) to accept or a registration rejection (RRJ) to refuse, as illustrated in Figure 14-
13. The gatekeeper can refuse the registration for many reasons, such as duplicate E.164 or H323
identifiers or ambiguous information.
Figure 14-13 Gatekeeper Registration and Unregistration

Cisco CallManager Gatekeeper
RRQ (1)
Registration
RCF/RRJ (2)
URQ (1)
Gatekeeper-Initiated
Unregistration Request
UCF (2)
URQ (1)
Endpoint-Initiated
Unregistration Request UCF/URJ (2)
An endpoint registration has a finite life. Before the TTL expires, the endpoint is required to renew
its registration by sending an RRQ. If the TTL expires and the gatekeeper has not received an RRQ
from the endpoint, the endpoint becomes unregistered.
Lightweight registration reduces the processing load on the gatekeeper during registration
renewal. The gatekeeper receives an RRQ with the keepalive bit set and the minimum required
information from the endpoint. If the gatekeeper accepts the renewal, the gatekeeper returns an
RCF to the endpoint and resets the TTL timer. If the gatekeeper rejects the renewal with an RRJ,
the endpoint becomes unregistered.
If the endpoint is unregistered, the endpoint must start the gatekeeper discovery and registration
process again.
At any time, an endpoint or a gatekeeper can cancel a registration with an unregister request
(URQ), normally used during configuration changes.
An endpoint or gatekeeper sends an unregister confirmation (UCF) in response to a URQ. If an

unregistered endpoint sends a URQ to a gatekeeper, the gatekeeper responds with an unregister
reject (URJ) to indicate the error. Cisco IOS gatekeepers, Cisco IOS gateways, and Cisco
CallManager support lightweight registration.
After the Cisco CallManagers from multiple clusters register with the gatekeeper, the gatekeeper
device can perform admission and bandwidth control between the clusters. Telephony endpoints
(Cisco IP Phones or Cisco IP SoftPhones) send an admission request (ARQ) to the gatekeeper to
initiate a call, as shown in Figure 14-14. The ARQ contains an H.323 identifier or the E.164
address of a destination or called party that it wants to reach. Also contained within the ARQ
message are the call bandwidth (not including the header overhead), the source E.164 address, and
the H.323 identifier of the calling party.
The gatekeeper will respond to the ARQ with either an admission confirmation (ACF) or an
admission reject (ARJ). The gatekeeper sends the ACF if the requested bandwidth is available and
the called endpoint is found. The ACF contains the IP address of the endpoint. On receipt of an
ACF from the gatekeeper, the endpoint sends a setup message directly to the other endpoint, using
the IP address returned in the ACF. If bandwidth is unavailable or if the called endpoint is not
registered, the gatekeeper sends an ARJ.
When an endpoint terminates a call, the endpoint is required to indicate the termination to the
gatekeeper and return the used bandwidth. The endpoint sends a disengage request (DRQ) to the
gatekeeper to indicate that the call is complete. The gatekeeper responds with a disengage
confirmation (DCF) and returns the previously used bandwidth to the pool.
Figure 14-14 Gatekeeper Admission Control

CallManager 1 Gatekeeper CallManager 2
ARQ (1)
ACF/ARJ (2)
Setup (3)
Call Proceeding (4)
ARQ (5)
ACF/ARJ (6)
Alerting (7)
Connect (8)
H.323 RAS Messages Call Signaling
The gatekeeper can also clear the call by sending a DRQ to the endpoint, forcing the endpoint to
clear the call with the other endpoint and return a DCF.
If during the duration of the call the bandwidth requirement changes, because of a changing codec
or additional media channels opening or closing, the endpoint can request or release the bandwidth
by sending a bandwidth request (BRQ). The gatekeeper responds with a bandwidth confirmation
(BCF) if the bandwidth is available or with a bandwidth reject (BRJ) if the bandwidth is not
available, as shown in Figure 14-15.
Figure 14-15 Gatekeeper Disengage Bandwidth Control

Cisco CallManager Gatekeeper
DRQ (1)
Disengage
DCF(2)
BRQ (1)
Bandwidth
BCF/BRJ (2)
Gatekeeper-Based Call Admission Control Configuration

The general recommendation is to use separate Cisco routers as dedicated gatekeepers in your
network in a number appropriate for your topology. Cisco also supports clustering the gatekeeper
routers using H.323 clustering with the Gatekeeper Update Protocol (GUP). You can configure a
gatekeeper with the appropriate Cisco IOS feature set, such as the Enterprise Plus/H323 MCM
feature set.
TIP Other feature sets of the IOS have begun adding gatekeeper support. The best way to find
out what feature set you should download is to use the Cisco feature navigator utility available
at http://www.cisco.com/go/fn.
The following are the four general steps for configuring call admission control using gatekeepers
and trunks:
Step 1 On the gatekeeper device, configure the appropriate zones and bandwidth
allocations for the various Cisco CallManager nodes that will route calls to it.
Example 14-1 shows a sample Cisco IOS gatekeeper simple configuration.
Example 14-1 Sample H.323 Gatekeeper Basic Configuration

Router(config)# gatekeeper
Router(config-gk)# zone local SJGK1 cisco.com
Router(config-gk)# zone prefix SJGK1 408*
Router(config-gk)# gw-type-prefix 1#* default-technology
Router(config-gk)# bandwidth total zone SJGK1 512
Router(config-gk)# bandwidth session zone SJGK1 256
Table 14-1 describes these commands.

Table 14-1 Cisco IOS Gatekeeper Commands
Command Description
Gatekeeper Enters gatekeeper configuration mode.
zone local SJGK1 Specifies the zone name controlled by the gatekeeper. In this
cisco.com case, cisco.com is the zone controlled by gatekeeper SJGK1.
zone prefix SJGK1 408* Associates the digits 408 with the local zone SJGK1.
gw-type-prefix 1#* Configures a technology prefix in the gatekeeper. Cisco
default-technology gatekeepers use technology prefixes to route calls when there are
no E.164 addresses registered (by a gateway) that match the
called number. With the default technology prefix option, the
Cisco gatekeeper assigns a default gateway or gateways for
routing unresolved call addresses. This assignment is based on
the registered technology prefix of the gateways.
continues
Table 14-1 Cisco IOS Gatekeeper Commands (Continued)
Command Description
bandwidth total zone Specifies the maximum bandwidth available in zone SJGK1 as
SJGK1 512 512 kbps.
bandwidth session zone Specifies the maximum bandwidth allowed for a session in zone
SJGK1 256 SJGK1 as 256 kbps.
Step 2 Configure gatekeeper settings in Cisco CallManager Administration. Make

sure that the hostname or IP address is set the same way on each Cisco
CallManager. You can register multiple gatekeepers per Cisco CallManager
cluster.
You can add gatekeepers to the CallManager configuration by accessing the
Device > Gatekeeper Configuration window. When you click on Add a
New Gatekeepeer, the configuration window shown in Figure 14-16
appears. Table 14-2 describes the configuration options.
Figure 14-16 Configuring a Gatekeeper Reference in Cisco CallManager

Table 14-2 Cisco CallManager Gatekeeper Configuration Options
Field Description
Host Name/IP Enter the IP address or hostname of the gatekeeper in this required field (IP
Address* address preferred).
Registration Do not change this value unless a Cisco Technical Assistance Center (TAC)
Request Time engineer instructs you to do so. The default value specifies 60 seconds.
to Live
The Registration Request Time to Live field indicates the time that the
gatekeeper considers a registration request (RRQ) valid. The system must send a
keepalive RRQ to the gatekeeper before the RRQ TTL expires.
Cisco CallManager sends an RRQ to the gatekeeper to register and subsequently

to maintain a connection with the gatekeeper. The gatekeeper may confirm
(RCF) or deny (RRJ) the request.
Registration Do not change this value unless a Cisco TAC engineer instructs you to do so.
Retry Timeout* Enter the time in seconds. The default value specifies 300 seconds.
The Registration Retry Timeout field indicates the time that Cisco
CallManager waits before retrying gatekeeper registration after a failed
registration attempt.
* Indicates required item
Step 3 Configure the appropriate intercluster trunks or H.225 trunks to specify

gatekeeper information. The H.225 trunk allows connectivity to H.323
Cisco IOS gateways. The intercluster trunk provides specific Cisco
functionality for calls between Cisco CallManager clusters.
After you have added the necessary gatekeeper(s) to the CallManager
configuration, you can add trunks connecting to the gatekeeper device.
Cisco CallManager supports two types of gatekeeper-controlled trunks:
• H.225 trunk (Gatekeeper Controlled)—In an H.323 network that uses
gatekeepers, use an H.225 trunk with gatekeeper control to configure a
connection to a gatekeeper for access to other Cisco CallManager
clusters and to H.323 devices (which could connect to third-party
networks). An H.225 trunk can communicate with any H.323
gatekeeper-controlled endpoint. When you configure an H.323 gateway
with gatekeeper control in Cisco CallManager Administration, use an
H.225 trunk. To use this method, choose Device > Trunk and then
choose H.225 Trunk (Gatekeeper Controlled).
• Inter-Cluster Trunk (Gatekeeper Controlled)—In a distributed call-
processing network with gatekeepers, use an Intercluster Trunk with
gatekeeper control to configure connections between clusters of Cisco
CallManager systems. Gatekeepers provide call admission control and

address resolution for intercluster calls. A single Intercluster Trunk can
communicate with all remote clusters. To use this method, choose
Device > Trunk and then choose Inter-Cluster Trunk (Gatekeeper
Controlled) in Cisco CallManager Administration.
You can add gatekeeper-controlled H.225 and intercluster trunks to the
Cisco CallManager configuration using the same process as adding non-
gatekeeper controlled trunks, which was discussed in Chapter 10, “Cisco
Unified CallManager Route Plan Basics.” The trunk should point to the
gatekeeper device rather than the remote cluster and be configured for the
appropriate default technology prefix and local zone name that you have
configured on the gatekeeper device, as shown in Figure 14-17.
Figure 14-17 Configuring a Gatekeeper-Controlled Trunk
NOTE The two different types of gatekeeper-controlled trunks have caused much confusion
with CallManager administrators. The Intercluster Trunk is a Cisco proprietary trunking method
that allowed communication between Cisco CallManager clusters. This type of trunk is only
necessary when communicating with clusters running Cisco CallManager 3.2 or earlier. The
H.225 trunk is an industry standard trunking method used to communicate with any type of
H.323-compliant device, including CallManager clusters since version 3.3 and Cisco
CallManager Express platforms.
Step 4 Configure a route pattern to route calls to each gatekeeper-controlled trunk.

After you have added the gatekeeper-controlled trunks to the configuration,
you can include them in the Cisco CallManager route plan just like any
other device.
Survivable Remote Site Telephony

In a centralized deployment model, if a WAN connection fails, remote IP Phones cannot make
calls. This weakness can be a serious problem when it comes to emergency calls.
A simple way of solving this problem is to provide limited call-processing capabilities in the
remote office router. Cisco IP Phone enhancements grant the ability to rehome to the call-
processing functions in the local router upon WAN failure detection. This solution is Cisco
Survivable Remote Site Telephony (SRST).
The Cisco SRST feature provides call-handling support on the gateway router for attached Cisco
IP Phones when a Cisco CallManager or WAN link fails. On restoration of the Cisco CallManager
or WAN link, the Cisco CallManager resumes the call-handling capabilities for the IP Phones. The
implementation of this feature is transparent to the end user, with the exception of the notification
that a user receives on the IP Phones that it is in fallback mode.
The SRST-enabled router supports the following:
■ IP Phone-to-IP Phone calls (at the local site)
■ IP Phone-to-PSTN calls
■ Direct Inward Dialing (DID)
■ Multiple lines per IP Phone
■ Multiple line appearance across IP Phones
■ Call hold and pickup on a shared line
■ Transfer of local calls
■ Caller ID information
■ Speed dialing
■ Last-number redial
■ Call transfer without consultation (local to router only)
■ Call hold and resume
■ Multiple extensions (up to six extensions per IP Phone on Cisco 7960 IP Phones)
■ Multiple line appearances, with up to 24 appearances per system
■ Distinctive ringing
■ Extension class of service (CoS)
■ Full interworking with Cisco gatekeeper functionality
■ Voice-mail support (only to local answering machine)

■ Billing support using call detail records (CDRs)
■ Dual-line mode
■ Music on Hold
SRST goes into effect when Cisco IP Phones detect that they are no longer receiving responses to
their keepalive packets from the Cisco CallManager, as shown in Figure 14-18. The Cisco IP
Phones can be configured to register with the router as a backup call-processing source. The SRST
software in the router automatically activates and builds a local database of all Cisco IP Phones
attached to it, up to the stated maximum. The SRST router now performs all local call setup and
processing, call maintenance, and call termination.
Figure 14-18 SRST Failover Process
Central Site Call

Manager (Cisco
CallManager)
PSTN
KA to Cisco
CallManager
IP
Real-Time V
Transport V
Protocol Branch Router
WAN
KA to Cisco
Headquarters
CallManager
IP Keepalive (TCP)
Branch Office
When the WAN link resumes, the IP Phones detect keepalive responses from the central Cisco
CallManager and revert to the central Cisco CallManager for primary call setup and processing.
As IP Phones rehome to the Cisco CallManager, the SRST router purges its call-processing
database and reverts to standby mode. Calls in progress continue without interruption. IP Phones
in use during WAN link recovery rehome to the Cisco CallManager after the calls terminate.
SRST Router Configuration

The most common application of SRST is to maintain basic IP telephony functionality for the IP
Phones at remote branch offices when the WAN link fails or the Cisco CallManager at
headquarters is no longer available. The remote branch router activates SRST functions and takes
over communication with the IP Phones. IP Phones are able to call each other and make off-net
calls to the PSTN. In addition, the most basic telephone functions, such as hold and transfer, are
still available.
There is one global command (with a series of subcommands) to configure for SRST, the call-
manager-fallback command. Example 14-2 shows a sample SRST configuration.
Example 14-2 SRST Sample Configuration

callmanager-fall back
Router(config)#c
access- code fxo 9
Router(config-cm-fallback)#a
defaul t-destination 4312
Router(config-cm-fallback)#d
dialplan- pattern 1 34 5555….
Router(config-cm-fallback)#d
ip sou rce-address 10.1.1.2 port 2000
Router(config-cm-fallback)#i
ke epalive 30
Router(config-cm-fallback)#k
max-eph ones 24
Router(config-cm-fallback)#m
max-dn 48
Router(config-cm-fallback)#m
transfer- pattern 525.. .
Router(config-cm-fallback)#t
voicemail 40 01
Router(config-cm-fallback)#v
Table 14-3 describes the function of these commands.
Table 14-3 Cisco SRST Command Descriptions
Command Description
access-code Defines access codes for outgoing calls.
default-destination Defines or disables the default destination DN for ringing on unknown
called number (typically used for incoming PSTN calls).
dialplan-pattern Defines an E.164 telephone number prefix. This command defines the
number of digits sent from the provider for incoming calls and dictates
how many digits should be forwarded to the phones. In the sample
syntax, the provider is sending ten-digit DID numbers and the internal
office is using four-digit extensions.
ip source-address Defines an IP address and port for the SRST router. This command
dictates which internal IP address should expect to receive Skinny
messages when the CallManager connection is lost.
keepalive Defines the keepalive timeout period to unregister IP Phones.
max-dn Specifies the maximum DNs supported on the IP Keyswitch.
max-ephones Specifies the maximum number of IP Phones.
transfer-pattern Defines valid call transfer destinations off the VoIP network. By default,
SRST allows transfers only between devices registered to the same
SRST router.
voicemail Sets the voice-mail access number called when the Messages button is
pressed.
SRST Reference Configuration

After you input the necessary Cisco IOS SRST gateway configuration, you must then configure
the Cisco CallManager to recognize the gateway as an SRST reference. The CallManager will then
deliver the SRST reference to the IP Phone through the configuration file on the TFTP server.
NOTE If the IP Phones at a remote location use their default gateway as the SRST router, it is
not necessary to configure SRST references. In the IP Phone’s device pool configuration, you
can select the Use Default Gateway option to point the IP Phones to the correct SRST device.
To configure SRST references, choose the System menu in the Cisco CallManager Administration
utility and then choose SRST. When the Find and List SRST References window appears, click
the Add a New SRST Reference link in the upper-right area of the window. A window similar to
the one shown in Figure 14-19 should appear. To create a valid SRST reference, you must enter
values in the following fields you input:
■ SRST Reference Name—This value is a logical name that you can use when referencing the
SRST gateway. It does not need to match the name assigned to the gateway.
■ IP Address—This value is the IP address that the Cisco IP Phone should use when contacting
the SRST gateway. The IP Phone itself must be able to reach this IP address.
■ Port—This value is the port number that the IP Phone should use when contacting the SRST
reference. By default, this uses TCP port you input 2000.
Figure 14-19 SRST Reference you input Configuration Window

Summary 301
The Is SRST Secure? check box and the SRST Certificate Provider Port field are used for
configuring the Cisco CallManager portion of a secure SRST-enabled gateway connection. Secure
SRST allows Cisco CallManager to authenticate with a secure SRST-enabled gateway and add the
SRST-enabled gateway certificate to the Cisco CallManager database. The TFTP server adds the
SRST certificate to the phone cnf.xml file and sends the file to the phone. A secure phone then uses
a secure connection to interact with the SRST-enabled gateway. Phone security is discussed in
Chapters 26, “IPT Authentication and Encryption Fundamentals” and 27, “Configuring IPT
Authentication and Encryption.”
After you create the input SRST reference in Cisco CallManager, you must assign the SRST
reference to the Cisco IP Phone. Cisco CallManager creates this assignment through the device
pool. In the Device Pool Configuration window, click the SRST Reference drop-down arrow to
choose the SRST reference that the IP Phone should use. If you want the Cisco IP Phone to use its
default gateway as the SRST reference, you can choose the Use Default Gateway option from the
menu. Using this option can simplify your SRST configuration.
NOTE Each router platform supports a different maximum number of IP Phones. Up-to-date
information on these numbers can be gathered from the SRST administration guide available on
the Cisco website you input.
Summary
This chapter covered the concepts and configuration of Cisco CallManager in a multiple-site
deployment. Call admission control (CAC) mechanisms are necessary in this type of deployment
to control the volume of calls between two endpoints, which is key to maintaining QoS of all calls.
Cisco CallManager supports two models of CAC: location-based CAC for centralized call-
processing environments and gatekeeper-based CAC for distributed call-processing environments.
Locations-based CAC is configured on the Cisco CallManager and allocates a specific audio and
video bandwidth amount to each location. IP Phones are then assigned to their respective locations
on a per-device basis.
Automated Alternate Routing (AAR) is a mechanism used with location-based CAC to

automatically reroute calls to the PSTN (or other network) when there is insufficient bandwidth
on the primary voice path. Configuring AAR entails configuring AAR groups and assigning the
external phone number mask to the DN assigned to each AAR-capable device.
In a distributed call-control environment, multiple Cisco CallManager clusters can exist. This
design requires an independent authority that can control the WAN bandwidth and call routing
between clusters. The H.323 gatekeeper is used to satisfy this requirement. Gatekeepers are
configured in the IOS software and added to the Cisco CallManager configuration to support
gatekeeper-controlled trunks. These trunks are then included in the Cisco CallManager route plan
to route outside of the cluster.
The final topic discussed in this chapter addresses the failover capabilities of remote sites in a
centralized CallManager environment. Survivable Remote Site Telephony (SRST) provides call-
handling support for IP Phones when the Cisco CallManager or WAN link fails. SRST is
configured on an IOS-based router and added to the Cisco CallManager configuration as an SRST
reference. The SRST reference is assigned to end devices through their device pool.
Review Questions
1. Which is a benefit associated with gatekeeper call admission control?

a. no requirement to configure Cisco IOS commands on the gatekeeper
b. no requirement to configure an Intercluster Trunk between clusters
c. provides emergency call-handling capability if the WAN link fails
d. provides call admission control over the intranet in addition to the WAN
2. Without call admission control, what happens to existing calls when the next call
oversubscribes the WAN?
a. All calls are dropped.
b. Voice quality on all calls degrades.
c. Voice quality on the last call degrades.
d. Nothing happens if the router is configured for QoS.
3. Which two characteristics are associated with the distributed call-control environment?
(Choose two.)
a. Intercluster Trunk to each Cisco CallManager cluster
b. gatekeeper-controlled trunks
c. maximum bandwidth determined on the Cisco CallManager
d. zones and bandwidth allocation configured on the gatekeeper
4. How do the Cisco IP Phones in a branch site know how to register to a gateway running
SRST?
a. The IP Phones stop receiving keepalive packets from the SRST gateway.
b. The IP address is configured on the gateway and passed to the IP Phone.
c. The gateway IP address is specified in the SRST Reference Configuration window and
assigned to a device pool.
d. The device pool contains the IP address reference to the SRST gateway.
5. Using the Cisco CallManager locations feature, how many simultaneous audio calls can be
placed over a given WAN link with 128 kbps of available bandwidth using a G.729 codec?
a. two
b. three
c. four
d. five
6. AAR derives the alternate dial string used to route the call over the PSTN from which three
of the following? (Choose three.)
a. DN
b. external phone number mask
c. AAR calling search space
d. MAC address
e. PSTN prefix
f. AAR group
7. Which command is used to initiate SRST capabilities on an IOS-based router?
a. telephony-service
b. call-fallback
c. call-manager-fallback
d. ccm-communication
8. You are working in a location-based CallManager deployment. Phone A (DN=3423) dials on-
net to Phone B (DN=3420). The location-based CAC denies the call and AAR initiates. Which
external phone number mask will be used by AAR to complete the call?
a. Phone A
b. Phone B
c. the gateway at the calling site
d. the gateway at the called site
9. What is the default state of AAR in a CallManager cluster?
a. AAR is enabled, but must be configured by an administrator.
b. AAR is enabled and preconfigured, provided the administrator has entered the correct
external phone number masks under the applicable DNs.
c. AAR is disabled until a feature license is applied.
d. AAR is disabled.
10. You have AAR configured in a cluster to divert calls to the PSTN should the IP WAN become
unusable. To test the configuration, you disconnect the WAN and place a call to an alternate
site. The CallManager returns a reorder tone. You are sure the digit transformation settings are
configured correctly; what is the most likely cause of this problem?
a. AAR does not function in a distributed call-control environment.
b. The AAR calling search space does not provide the necessary privileges to complete the
call.
c. The standard calling search space does not provide the necessary privileges to complete
the call.
d. AAR does not function during a WAN failure.
Part IV: VoIP Features
Chapter 15 Media Resources
Chapter 16 Configuring User Features, Part 1
Chapter 17 Configuring User Features, Part 2
Chapter 18 Configuring Cisco Unified CallManager Attendant Console
Chapter 19 Configuring Cisco IP Manager Assistant

following topics:
■ Activating all necessary Cisco CallManager

services that are used in media resources
■ Configuring conference bridge resources to

enable ad hoc and Meet-Me conferencing
between IP Phones
■ Describing the services that MTP resources

provide
■ Identifying the Cisco CallManager resources

that are required for the annunciator feature
■ Configuring transcoder resources in Cisco

CallManager and on a Cisco access gateway
to provide codec conversion
■ Configuring audio sources for MoH and

assigning user and network MoH to IP
Phones
■ Allocating media resources to devices using

MRGs and MRGLs
CHAPTER 15
Media Resources
Conferencing, music on hold (MoH), and informational tones and spoken word progress tones
are important business phone services. Media resources provide these and other services in a
Cisco CallManager IP telephony deployment. This chapter describes how to configure and
allocate media resources to include conferencing, Media Termination Points (MTPs), the
annunciator, transcoders, and MoH within a Cisco CallManager solution.
Introduction to Media Resources

A media resource is a software-based or hardware-based entity that performs media processing
functions on the voice streams to which it is connected. Media-processing functions include
mixing multiple streams to create mixed, unicast output streams (conferencing), passing the
stream from one connection to another (Media Termination Point), converting the data stream
from one compression type to another (transcoding), streaming voice progress tones
(annunciation), and music on hold.
Every Cisco CallManager contains a software component called a Media Resource Manager
(MRM) that communicates with MRMs on other Cisco CallManager servers. The MRM locates
a media resource to connect the media streams and to complete the feature. The MRM manages
the following media resource types:
■ Conference bridge
■ MTP
■ Annunciator
■ Transcoder
■ MoH server
NOTE To act as a Media Resource Manager, you must activate the Voice Media Streaming
Application service on the Cisco CallManager server. This service is discussed in detail later
in this chapter.
310 Chapter 15: Media Resources
Your Cisco CallManager servers can also participate in the media-processing functions in the
cluster. After Cisco CallManager is installed, you must activate these two Cisco CallManager
services:
■ Cisco IP Voice Media Streaming Application—The Cisco IP Voice Media Streaming

Application provides voice media streaming functionality for Cisco CallManager for use with
MTP, the annunciator, conferencing, and MoH. The Cisco IP Voice Media Streaming
Application relays messages from Cisco CallManager to the IP voice media streaming driver.
The driver handles the Real-Time Transport Protocol (RTP) streaming.
■ Cisco MoH Audio Translator—The Cisco MoH Audio Translator service converts audio
source files, which can be in .WAV or .MP3 format, into the appropriate MoH source file for
various coder-decoders (codecs) so that the MoH feature can use them.
To activate the required services in Cisco CallManager, access the administration web page and
choose Application > Cisco CallManager Serviceability. In the Serviceability window, choose
Tools > Service Activation. You can activate the services on any server that you choose.
Conference Bridge Resources

A conference bridge is a resource that joins multiple participants into a single call. Depending on
the conference bridge you are using, each conference call is limited to a certain number of
participants and specific codecs. There is a one-to-one correspondence between media streams
that are connected to a conference and participants who are connected. The conference bridge
mixes the streams together and creates a unique output stream for each connected party. The
output stream for a given party is the composite of the streams from all connected parties minus
its own input stream.
Cisco CallManager supports both hardware and software conference bridges. Hardware and
software conference bridges can be active at the same time.
Hardware-enabled conferencing provides the ability to support voice conferences using hardware
resources. Digital signal processors (DSPs) convert multiple voice over IP (VoIP) packets into
streams that are mixed into a single conference call stream. The DSPs support both Meet-Me and
ad hoc conferences. Hardware conference devices can provide a mixing of multiple codecs on the
same conference call for G.711, G.729, G.723, Global System for Mobile Communication (GSM)
Full Rate (FR), and GSM Enhanced Full Rate (EFR). Not all hardware resources support all of
these codecs, but they all support at least G.711 and G.729, which are the two codecs supported
by all modern Cisco IP Phones.
Software conferences use the resources of Cisco CallManager. A software conference bridge is
capable of mixing G.711 audio streams and Cisco Wideband audio streams. Any combination of
Wideband or G.711 a-law and mu-law streams can be connected to the same conference.
Conference bridges enable both ad hoc and Meet-Me voice conferencing:
■ Ad hoc conference—A user, known as the conference controller, adds participants to a

conference by calling the new participant and pressing the Confrn softkey. Alternatively, the
conference controller can press the Select softkey and then press the Join softkey to make it
an ad hoc conference. Up to 15 established calls can be added to the ad hoc conference (16
total). Only the conference controller can add participants to an ad hoc conference. An ad hoc
conference can continue if the conference controller hangs up, but new participants cannot be
added. When only two participants remain in conference, the conference terminates and the
two remaining participants are reconnected directly as a point-to-point call. This action saves
conference resources.
■ Meet-Me conference—A user, known as the conference controller, presses the MeetMe
button or softkey and establishes the conference. The administrator must configure a directory
number (DN) or range of DNs in Cisco CallManager Administration. The conference
controller provides the DN to the participants, and at the appointed time, participants dial the
DN to join the conference. Participants can leave the conference by hanging up. If the
conference controller hangs up, the conference continues if there are least two participants on
the bridge.
NOTE The Meet-Me conference functionality included with Cisco CallManager is very
basic and does not provide scheduling and/or security capabilities. The Cisco Meeting Place
or Meeting Place Express software can be used for this functionality.
Conference Bridge Hardware

Table 15-1 identifies the conference bridge types that exist in Cisco CallManager 4.1
Administration. For each conference bridge type, the table lists the supported product or
application, the supported codecs, and the maximum number of participants.
Table 15-1 Conference Bridge Hardware
Cisco CallManager Conferences Codecs Max. Participants

Resource Type Resource per Conference
Cisco Conference WS-SVC-CMM G.711 mu-law, G.711 8 per conference, 64
Bridge a-law, G.729 annex A conferences per port
and annex B, and adapter
G.723.1
continues
Table 15-1 Conference Bridge Hardware (Continued)
Cisco CallManager Conferences Codecs Max. Participants

Resource Type Resource per Conference
Cisco Conference WS-X6608-T1 WS- G.711, G.723, G.729, 1 CFB–32 users to 10
Bridge Hardware X6608-E1 GSM FR, GSM EFR CFB–3 users
For G.711 or G.729a:

256 streams total per
module, 32 streams
maximum per port
Cisco Conference Cisco IP Voice Media G.711, Cisco 64–Ad Hoc
Bridge Software Streaming App. Wideband
128–Meet-Me
Cisco IOS Conference NM-HDV G.711, G.729 6
Bridge
Cisco IOS Enhanced NM-HD G.711, G.729, GSM 8
Conference Bridge FR, GSM EFR
NM-HDV2
G.711, G.729, or
WS-SVC-CMM-ACT G723
Cisco Video IP/VC-35xx Numerous audio and Varies by platform
Conference Bridge video coding schemes
(IP/VC-35xx)
Brief descriptions of the conference resources follow:
■ WS-SVC-CMM—This conference bridge type, when configured with the ACT port adapter,
supports the Cisco Catalyst 6500 Series and Cisco 7600 Series Communication Media
Module (CMM). This conference bridge type supports up to eight parties per conference and
up to 64 conferences per port adapter. This conference bridge type supports the following
codecs: G.711 mu-law, G.711 a-law, G.729 annex A and annex B, and G.723.1. This
conference bridge type supports ad hoc conferencing.
■ WS-X6608-T1 and WS-X6608-E1—The WS-X6608-T1 and WS-X6608-E1 provide eight

digital T1 or E1 interfaces for public switched telephone network (PSTN) and PBX gateway
access, transcoding, and conference bridging. Each one of these eight T1/E1 ports can be used
for either WAN/PSTN connectivity or to provision the DSP resources for media resource
support, but not both at the same time.
■ Cisco IP Voice Media Streaming Application—The Cisco IP Voice Media Streaming

Application provides voice media streaming functionality for the Cisco CallManager for use
with MTP, conferencing, and MoH. The Cisco IP Voice Media Streaming Application relays
messages from the Cisco CallManager to the IP voice media streaming driver. Supporting
services using the Cisco CallManager are limited to G.711 only (with the exception of
streaming MoH) because the CallManager is not capable of performing transcoding
functions.
■ NM-HDV—The NM-HDV network modules (NMs) provide conferencing, transcoding,

voice termination, and a voice WAN interface card (VWIC) slot, and include the NM-HDV-
2E1-60, NM-HDV-2T1-48, NM-HDV-FARM-C36, NM-HDV-FARM-C54, and NM-HDV-
FARM-C90. The NM-HDV modules are first generation DSP farms. Much of this
functionality has been merged into the NM-HD and NM-HDV2 modules.
■ NM-HD—The NM-HD modules include the NM-HD-1V, NM-HD-2V, and NM-HD-2VE.

The NM-HD-1V has one video interface card (VIC) slot and supports one analog or BRI VIC.
The NM-HD-2V has two VIC slots and supports two analog or BRI VICs (or any
combination). The NM-HD-2VE has two VIC slots and supports two analog or BRI VICs or
two digital VWICs (or any combination).
■ NM-HDV2—The NM-HDV2 supports up to eight voice channels of analog or BRI voice.

The NM-HDV2 supports a VIC/VWIC slot that can be fitted with either digital or analog or
BRI voice/WAN interface cards, and supports up to 60 channels of digital voice or four
channels of analog voice. The NM-HDV2-1T1/E1 adds support for one built-in T1 or E1 port
and supports up to 90 channels of digital voice or 30 channels of digital voice and four
channels of analog voice. The NM-HDV2-2T1/E1 adds support for two built-in T1 or E1
ports and supports a maximum of 120 channels of digital voice or 60 channels of digital voice
and four channels of analog voice.
■ NM-HD-2VE—The NM-HD-2VE supports analog, BRI, T1, E1, voice, and data WANs and
up to 48 channels (G.711 codec).
■ IP/VC-3500 Series—The Cisco IP/VC 3500 Series Video Multipoint Conference Unit is a
dual multimedia bridge that provides videoconferencing. The videoconference bridge
provides audio and videoconferencing functions for video Cisco IP Phones, H.323 endpoints,
and audio-only Cisco IP Phones with the VT Advantage package. The Cisco IP/VC 3500
Series includes the Cisco IP/VC 3511, the Cisco IP/VC 3521 BRI Videoconferencing
Gateway, and the IP/VC 3526 PRI Videoconferencing Gateway. For large or centralized
videoconferencing deployments, the scalable, multifunctional IP/VC 3540 Series
Videoconferencing System offers the combination of MCUs, a videoconferencing gateway, a
rate matching module, and a data collaboration server. Each product in the series supports a
number of audio and video codecs (H.261, H.263, H.263++, and H.264); these vary by
platform.
Table 15-2 provides additional detail about audio conference bridges.

Table 15-2 Conference Bridge Hardware
Conference Number of Participants and Bridges Resource

Type in Cisco
CallManager
Hardware For G.711 or G.729a: WS-X6608-T1
• 256 streams total per module and 32 streams maximum WS-X6608-E1

per port.
• Bridges can range from 10 bridges with 3 participants to 1
bridge with 32 participants.
For all GSM:
• 192 streams per module and 24 streams per port

CallManager IP Voice Media Streaming Application on a standalone Software
Software (Cisco server:
IP Voice Media
Streaming • 128 streams.
Application) • Ad hoc conference bridges can range from 42 bridges
with 3 participants to 2 bridges with 64 participants.
• A Meet-Me conference is 1 bridge with up to 128
participants.
IP Voice Media Streaming Application coresident with Cisco
CallManager:
• 48 streams.
• Bridges can range from 16 bridges with 3 participants to 1
bridge with 48 participants.
Cisco IOS • 3 bridges per PVDM-12 NM-HDV
software • 3, 6, 9, or 15 bridges per NM
• Maximum of 6 participants per bridge
For information on the PVDM2-8, PVDM2-16, PVDM2-32,
PVDM2-48, and PVDM2-64, refer to the “Hardware
Resources for MTP, Conferencing, and Transcoding” section
of the Cisco IP Telephony Solution Reference Network
Design (SRND) for Release 4.1 at: http://www.cisco.com/go/
srnd
Enhanced Cisco The total number of conference sessions is limited by the NM-HD
IOS software capacity of the entire system, the Cisco CallManager, and
the codec. Further details can be found in the “Cisco NM-HDV2
Enhanced Conferencing and Transcoding for Voice Gateway
Routers” data sheet at: http://www.cisco.com/en/US/
products/ps5855/
products_data_sheet0900aecd801b97a6.html
NOTE The hardware and capacities described in Table 15-2 are accurate at the time of this
writing; however, Cisco is continually adding new hardware and making improvements on
existing hardware. To ensure accuracy when provisioning your network, refer to the latest
CallManager SRND documentation available at http://www.cisco.com/go/srnd.
Conference Bridge Hardware CallManager Configuration

You must add hardware-based conference bridges to the Cisco CallManager configuration for the
cluster to utilize them. The process of adding conference bridge resources differs, based on the
type of conference resources you are adding. As an example, the following steps demonstrate the
addition of a WS-X6608 as a DSP farm resource to the Cisco CallManager configuration:
Step 1 Choose Service > Media Resource > Conference Bridge and then choose
Add a New Conference Bridge.
Step 2 Choose Cisco Conference Bridge Hardware in the Conference Bridge
Type field. The Conference Bridge Configuration window shown in Figure
15-1 appears.
Figure 15-1 Cisco Conference Bridge Hardware Configuration
Step 3 Add the MAC address of the WS-X6608 port that will be used for
conferencing.
On the WS-X6608, the port must be pointed toward the Cisco TFTP server to obtain the
configuration file and a list of the Cisco CallManager servers within the Cisco IP telephony
network to use for registration. Use the set port voice interface command on Cisco Catalyst
switches running the Catalyst operating system to point the port toward the Cisco TFTP server
(usually the publisher server).
The following example disables Dynamic Host Configuration Protocol (DHCP) on port 3/1 of a
6608 port, assigns an IP address, subnet mask, VLAN number, and default gateway, and points the
port to the Cisco TFTP server (in bold):
set port voice interface 3/1 dhcp disable 172.16.10.201 255.255.255.0 vlan 15 tftp
172.16.10.5 gateway 172.16.10.1
NOTE Any Cisco CallManager servers in the cluster that have the Voice Media Streaming
Application service activated will automatically be listed and used as a software-based
conference resource. If you do not want a CallManager server to participate in conference
bridge processing (to save resources), you can remove it as a conference bridge server by
deleting the CallManager server in the Conference Bridge Configuration window.
Media Termination Point Resources

Media Termination Points (MTPs) are used to extend supplementary services to H.323 endpoints
that do not support the H.323 Version 2 (H.323v2) OpenLogicalChannel and
CloseLogicalChannel request features with the EmptyCapabilitiesSet feature. Supplementary
services are features such as:
■ Call Hold
■ Call Transfer
■ Call Park
■ Conferencing
An MTP is an entity that accepts two full-duplex G.711 streams. It bridges the media streams
together and allows them to be set up and torn down independently. Simply put, the MTP allows
a call to be held (maintained) when a user places it on hold due to one of the services mentioned
previously. The streaming data received from the input stream on one connection is passed to the
output stream on the other connection, and vice versa. In addition, the MTP transcodes G.711 a-
law to G.711 mu-law, and vice versa, and adjusts packet sizes as required by the two connections.
When needed, an MTP is allocated and connected into a call on behalf of an H.323 endpoint. After
being inserted, the media streams are connected between the MTP and the H.323 device, and these
connections are present for the duration of the call. The media streams connected to the other side
Media Termination Point Resources 317
of the MTP can be connected and disconnected as needed to implement features such as hold,
transfer, and so forth.
Cisco CallManager requires an RFC 2833 dual-tone multifrequency (DTMF)-compliant MTP

device to make Session Initiation Protocol (SIP) calls. The current standard for SIP uses in-band
payload types to indicate DTMF tones. Cisco components such as Skinny-based IP Phones
support only out-of-band payload types. Thus, an RFC 2833-compliant MTP device monitors for
payload type and acts as a translator between in-band and out-of-band payload types. With the
MTP device, any service that requires a media change (such as call hold) happens transparently.
Table 15-3 provides additional detail about MTP resources.
Table 15-3 Media Termination Point Resources
MTP Type Limitations

Cisco IOS Software This type supports Cisco 2600XM, 2691, 2811, 2821, 2851, 3660, 3725,
Enhanced Media 3745, 3825, and 3845 access routers and the following MTP cases:
Termination Point
For a software-only implementation that does not use DSP but has the same
packetization time for devices that support G.711-to-G.711 or G.729-to-
G.729 codecs, this implementation can support up to 500 sessions per
gateway.
For a hardware-only implementation with DSP for devices that use G.711
codec only, 200 sessions can occur per NM-HDV2 and 48 sessions can
occur per NM-HD.
Cisco IOS Software Enhanced Media Termination Point does not support
RFC 2833 (DTMF relay).
This type can support Network Address Translation (NAT) in a service

provider environment to hide the private address.
In Cisco CallManager Administration, ensure that you enter the same MTP
name that exists in the gateway command-line interface (CLI).
Cisco CallManager A single MTP provides a default of 48 MTP (user-configurable) resources,
Media Termination depending on the speed of the network and the network interface card
Point Software (NIC). For example, a 100-Mbps network card/NIC can support 48 MTP
(Voice Media resources.
Streaming
Application) For a 10-Mbps network card/NIC, approximately 24 MTP resources can be
provided; however, the exact number of MTP resources that are available
depends on the resources that are being consumed by other applications on
that server, the speed of the processor, network loading, and various other
factors.
The Cisco IP Voice Media Streaming Application supports RFC 2833.

MTP Configuration
By default, any Cisco CallManager software-based MTP resources are automatically added to the
cluster when you have activated the Voice Media Streaming Application on a server. If you do not
want a CallManager server to participate in MTP processing (which saves resources), delete the
server from under the MTP Configuration window. Hardware-based MTP resources (known as
Cisco IOS Enhanced Software Media Termination Points) must manually be added to the cluster.
To configure or add an MTP, choose Service > Media Resource > Media Termination Point
from the Cisco CallManager Administration console and click Add a New Media Termination
Point. The Media Termination Point Configuration window shown in Figure 15-2 appears.
Figure 15-2 Media Termination Point Configuration Window
From here, you can choose to add a software- or hardware-based conference bridge to the
CallManager cluster configuration. Table 15-4 describes the configuration options available from
this window.
Table 15-4 Media Termination Point Configuration Options
Field Description
Media Termination Choose the MTP type, either Cisco IOS Enhanced Software Media
Point Type Termination or Cisco Media Termination Point Software.
Host Server Choose the server to run MTP (for software-based MTPs only).
Annunciator Resources 319
Table 15-4 Media Termination Point Configuration Options (Continued)
Field Description
Media Termination Enter an MTP name, up to 15 alphanumeric characters.
Point Name
Description Enter a description for MTP.
Device Pool Choose your preferred device pool or choose Default.
Annunciator Resources
An annunciator uses the Cisco IP Voice Media Streaming Application service and enables Cisco
CallManager to play prerecorded announcements (.wav files) and tones to Cisco IP Phones,
gateways, and other configurable devices. The annunciator enables Cisco CallManager to alert
callers as to why a call has failed. The annunciator can also play tones for some transferred calls
and some conferences.
In conjunction with Cisco CallManager, the annunciator device provides multiple one-way, RTP
stream connections to devices such as Cisco IP Phones and gateways. For example, a user at
extension 1001 dials extension 2503, an invalid number. The system cannot complete the call. The
annunciator device plays a one-way RTP stream to extension 1001: “Your call cannot be
completed as dialed. Please consult your directory and call again or ask your operator for
assistance. This is a recording.”
The annunciator plays the announcement or tone to support the following conditions:
■ Announcement—For devices that are configured for Cisco CallManager Multilevel

Precedence Preemption (MLPP).
■ Barge tone—Before a participant joins an ad hoc conference.
■ Ringback tone—When you transfer a call over the PSTN through a Cisco IOS gateway, over
an H.323 Intercluster Trunk, or to the SIP client from an SCCP phone, the annunciator plays
the tone because the gateway cannot play the tone when the call is active.
To add an annunciator to the Cisco CallManager, you must activate the Cisco IP Voice Media
Streaming Application service on the server where you want the annunciator to exist in the cluster.
Table 15-5 shows sample annunciator announcements.

Table 15-5 Sample Annunciator Announcements
Condition Announcement
An equal- or higher- “Equal- or higher-precedence calls have prevented the completion of
precedence call is in your call. Please hang up and try again. This is a recording.”
progress.
A precedence access “Precedence access limitation has prevented the completion of your call.
limitation exists. Please hang up and try again. This is a recording.”
A service interruption “A service disruption has prevented the completion of your call. In case
occurred. of emergency, call your operator. This is a recording.”
For a single annunciator, Cisco CallManager sets the default maximum to 48 simultaneous
streams, as indicated in the annunciator service parameters (accessed under the service parameters
for the Voice Media Streaming Application). It is recommended that you not exceed 48
annunciator streams on a coresident server where the Cisco CallManager and Cisco IP Voice
Media Streaming Application services run. If the server has only 10-Mbps connectivity, lower the
setting to 24 simultaneous streams.
If the annunciator runs on a standalone server where the Cisco CallManager service does not run,
the annunciator can support up to 255 simultaneous announcement streams. If the standalone
server has dual CPUs and a high-performance disk system (as in the case of the 7845 MCS server),
the annunciator can support up to 400 simultaneous announcement streams. You can add multiple
standalone servers to support the required number of streams.
Each annunciator can support G.711 a-law, G.711 mu-law, wideband, and G.729 codec formats.
A separate .wav file exists for each codec that is supported.
Annunciator Configuration
Because the annunciator service requires access to a large amount of audio files, only the Cisco
CallManager can act as an annunciator in the cluster. There are no hardware-based resources that
exist. When you activate the Cisco IP Voice Media Streaming Application service in Cisco
CallManager Serviceability, Cisco CallManager automatically adds the annunciator device to the
server configuration.
The annunciator configuration is almost identical to the MTP configuration. Minimally, you must
select a host server, enter the annunciator name, and assign the annunciator to a device pool, as
Figure 15-3 Annunciator Configuration Window
Each annunciator registers with only one Cisco CallManager at a time. The system might have
multiple annunciators, depending on your configuration, each of which can register with different
Cisco CallManager servers, depending on the device pool to which you have assigned the
annunciator service. Each annunciator belongs to a device pool. The device pool associates the
secondary (backup) Cisco CallManager and the region settings.
When you update or configure the annunciator service parameters, the changes automatically
occur when the annunciator becomes idle, when no active announcements are played.
Transcoder Resources
A transcoder device takes the output stream of one codec and converts the voice streams to another
compression type. For example, a transcoder can take an output stream from a G.711 codec and
convert it to a G.729 stream. Depending on the hardware resources you are using, transcoders for
Cisco CallManager convert between G.711, G.723, G.729, and GSM codecs. A transcoder device
provides additional capabilities and can be used to enable supplementary services for H.323
endpoints.
Figure 15-4 shows a transcoder device (XCODE) enabling communication between two different
codecs and providing MTP services for H.323 endpoints.
Figure 15-4 Transcoder Operation

XCODE Voice Messaging
IP
G.729 G.711
The Cisco CallManager invokes a transcoder on behalf of endpoint devices when the two devices
use different voice codecs and would normally not be able to communicate. When inserted into a
call, the transcoder converts the data streams between the two incompatible codecs to enable
communications between them. The transcoder remains invisible to either the user or the
endpoints that are involved in a call. For example, a user could be communicating across the IP
WAN using the G.729 codec to a legacy voice-mail server supporting only G.711. Cisco
CallManager can invoke transcoding resources to convert between the codecs and allow
communication to occur.
NOTE Because of the amount of resources necessary to perform transcoding, Cisco

CallManager servers cannot be used as transcoding resources.
Table 15-6 shows the currently available transcoding resources and the resource capabilities of
each.
Table 15-6 Transcoder Resource Capabilities
Resource From Codec To Codec

WS-X6608-T1, G.723. G.729a, GSM FR, GSM EFR G.711 a-law or mu-law
WS-X6608-E1
G.711 a-law or mu-law G.723. G.729a, GSM FR, GSM EFR
NM-HDV and G.729, G.729a, G.729b, G.729ab G.711 a-law or mu-law
NM-HDV-
G.711 mu-law G.729, G.729a, G.729b, G.729ab
FARM
NM-HD and G.729, G.729a, G.729b, G.729ab, G.711 a-law or mu-law
NM-HDV2 GSM FR, GSM EFR
G.711 a-law or mu-law G.729, G.729a, G.729b, G.729ab,
GSM FR, GSM EFR
WS-SVC-CMM This type provides transcoding between any combination of the following
codecs:
• G.711 a-law and G.711 mu-law
• G.729 annex A and annex B
• G.723.1
• GSM (FR)
NOTE Cisco is continually adding transcoding-capable devices to their product line. For
the latest information on transcoding resources, check the Cisco website.
Transcoder Configuration
To configure or add a new transcoder, choose Service > Media Resource > Transcoder, click
Add a New Transcoder (shown in Figure 15-5), and then follow this general procedure:
Step 1 Choose the appropriate transcoder type: Cisco Media Termination Point
Hardware, Cisco IOS Media Termination Point, Cisco IOS Enhanced
Media Termination Point, or Cisco Media Termination Point (WS-SVC-
CMM).
Step 2 For Cisco Media Termination Point Hardware or Cisco Media Termination
Point (WS-SVC-CMM), enter a MAC address, which must be 12
characters.
Step 3 For Cisco Media Termination Point (WS-SVC-CMM) transcoders, choose
a subunit from the drop-down list (not shown in the figure).
Step 4 Choose a device pool. For more detailed information on the chosen device
pool, click View Details.
Step 5 Enter any special load information into the Special Load Information
field or leave it blank to use the default. Valid characters include letters,
numbers, dashes, dots (periods), and underscores.
Step 6 For Cisco Media Termination Point (WS-SVC-CMM) transcoders, choose
a maximum capacity from the drop-down list (not shown in the figure).
Figure 15-5 Transcoder Configuration Window
NOTE The preceding procedure shows the CallManager transcoder configuration for the
WS-SVC-CMM module. The configuration fields for each type of transcoding resource are
unique.
Music on Hold Resources

The integrated Music on Hold (MoH) feature provides On-Net and Off-Net users with on hold
music from a streaming source. The MoH feature has two hold types:
■ User hold—A user presses the Hold button or Hold softkey.
■ Network hold—A user activates the transfer, conference, or Call Park feature, which
automatically activates the hold feature before performing the operation.
MoH is customizable so that it plays specific recordings, based on the DN that is used to place the
caller on hold or the line number that the caller has dialed. Recorded audio or a live audio stream
can be configured as audio sources.
The MoH feature requires the use of a server that is part of a Cisco CallManager cluster. You can
configure the MoH server in either of the following ways:
■ Coresident deployment—In a coresident deployment, the MoH feature runs on any server
(either publisher or subscriber) in the cluster that is also running the Cisco CallManager
software. Because MoH shares server resources with Cisco CallManager in a coresident
configuration, this type of configuration drastically reduces the number of simultaneous
streams that an MoH server can send.
■ Standalone deployment—A standalone deployment places the MoH feature on a dedicated

server within the Cisco CallManager cluster. The sole function of this dedicated server is to
send MoH streams to devices within the network. A standalone deployment allows for the
maximum number of streams from a single MoH server.
Table 15-7 provides details on the number of MoH sessions and codecs that are supported for each
server platform.
Table 15-7 Transcoder Resource Capabilities
Server Platform Codecs Supported MoH Sessions Supported

Cisco MCS 7815 G.711 (a-law and mu-law) Coresident server:
20 MoH sessions
Cisco MCS 782x (all models) G.729a
Standalone MoH server: 50 MoH
Cisco MCS 7830 (all models) Wideband audio sessions
Cisco SPE-310
Hewlett-Packard DL320
IBM xSeries 33x (all models)

Cisco MCS 7835 (all models) G.711 (a-law and mu-law) Coresident server:
20 MoH sessions
Cisco MCS 7845 (all models) G.729a
Standalone MoH server:
Hewlett-Packard DL380 Wideband audio 250 MoH sessions
IBM xSeries 34x (all models)
The maximum session limits apply to unicast, multicast, or simultaneous unicast and multicast
sessions. The limits represent the recommended maximum sessions that a platform can support.
Creating Audio Source Files

Figure 15-6 shows the interactions among the audio translator, default MoH TFTP server, and
MoH server. The large boxes in the figure represent cluster components that might reside on a
single server or on three separate servers. This figure also shows how the MoH server processes
the added audio source file.
Figure 15-6 Creating Audio Source Files
New files are 2 Audio MOH Master

automatically 3 Default MOH
detected and
Translator Storage
TFTP Server
processed. Service Directory
Cisco CallManager 4 TFTP Path

Audio Source Input Directory Administration copies Directory
C:\Program Files\Cisco\MOH\ audio source files when
DropMOHAudioSourceFilesHere they are mapped.
Start 1 5
Hard-Coded MOH
Administrator copies audio MOH
Server Audio
source into this directory. Server
Source Directory
IP
Kernel-Mode
DirectShow
RTP Streaming
6 Filters
Driver
IP IP
In creating an audio source, the following sequence takes place, as shown in the figure:
Step 1 The network administrator drops the audio files into the C:\Program
Files\Cisco\MoH\DropMoHAudioSourceFilesHere directory path.
Standard .wav and MP3 files are valid input.
NOTE It takes approximately 30 seconds to convert a 3-MB MP3 file.
Step 2 Cisco CallManager automatically detects and translates the files.

Step 3 The output and source files are moved into the default MoH TFTP server
holding directory. This holding directory is the same as the default
TFTPMoHFilePath with \MoH appended.
NOTE It is not recommended that the audio translator service be used during production
hours because the service will consume 100 percent of the CPU.
Step 4 The network administrator assigns an audio source number to the audio
source file. The corresponding audio source files are then copied to a
directory that is one level higher in the directory structure (which is the
default TFTP path) to make them available to the MoH servers.
Step 5 The MoH servers download the needed audio source files and store them in
the hard-coded directory C:\Program Files\Cisco\MoH.
Step 6 The MoH server then streams the files using DirectShow and the kernel-
mode RTP driver as needed or requested by Cisco CallManager.
Configuring MoH
The Cisco CallManager MoH Configuration is a six-step process:
Step 1 Configure the audio translator.

Step 2 Configure the MoH server.
Step 3 Add and configure audio source files.
Step 4 Set MoH servicewide settings.
Step 5 Find and configure the fixed audio source.
Step 6 Assign audio source IDs.
Step 1: Configure the Audio Translator

An MoH audio translator service converts administrator-supplied audio sources to the proper
format for the MoH server to use. The audio translator uses two parameters, an input directory and
an output directory. You can configure the input directory, which defaults to:
C:\Program Files\Cisco\MoH\DropMoHAudioSourceFilesHere, on a per-service basis.
The output directory, a clusterwide parameter, contains a Universal Naming Convention (UNC)
name to a shared directory on the default MoH TFTP directory. For whichever directory is
specified, append \MoH.
To display the Service Parameters Configuration window, choose Service > Service Parameters
in Cisco CallManager Administration, choose the server that will provide MoH (typically, the
publisher if not a standalone server), and choose Cisco MoH Audio Translator for the service.
Figure 15-7 illustrates this configuration.
Figure 15-7 Cisco MoH Audio Translator Service Configuration
After you configure the audio translator service, you can drop your MoH audio source files (in
.wav or .mp3 format) into the drop directory you specify. As soon as you copy the files into the
directory, the CallManager’s audio translator service immediately goes to work converting the
files into the various audio codecs compatible with the VoIP network.
CAUTION Converting the audio source files into audio codec format can cause massive
processor utilization on the CallManager server, potentially keeping it at 100 percent
utilization for a short time (depending on the size and number of audio source files you copy).
Cisco highly recommends doing this conversion during network off-hours.
TIP Be sure to copy the MoH audio source files into the drop directory. If you move the files
(by dragging and dropping them from a source on the same hard disk partition, such as the
desktop), the files retain their original permissions rather than inheriting the permissions of
the parent directory. If the permissions are not set correctly, the CallManager will not be able
to convert the audio files.
Step 2: Configure the MoH Server

Figure 15-8 shows the MoH Server Configuration window. The MoH server can be configured for
unicast or multicast. Multicast MoH conserves system resources. Multicast allows multiple users
to use the same audio source stream to provide MoH. Multicast audio sources associate with an
IP address. To use multicast audio for MoH, your network must be configured to support multicast
traffic. In addition, the audio source is always playing. Users placed on hold will typically join the
audio source midstream; if you have announcements that play at the beginning of the audio stream,
they might be missed.
Figure 15-8 Cisco MoH Server Configuration
Unicast MoH, the system default, uses a separate source stream for each user or connection. Users
connect to a specific device or stream. If your network supports multicast traffic, you can deploy
MoH services much more efficiently. Cisco recommends using a locally scoped multicast address
(239.x.x.x) to ensure MoH multicast traffic does not stream over the IP WAN unnecessarily.
Choose Service > Media Resource > Music On Hold Server to display the Music On Hold
(MoH) Server Configuration window.
Step 3: Add and Configure Audio Source Files

The next step in configuring MoH is to add and configure the audio source files that the
CallManager has converted (by adding .wav or .mp3 files to the directory shown in Step 1) to audio
source IDs. Choose Service > Media Resource > Music On Hold Audio Source to display the
Music On Hold (MoH) Audio Source Configuration window, shown in Figure 15-9.
Figure 15-9 Configuring Audio Source Files
To add audio source files, click the Add New MoH Audio Source link, choose an MoH Audio
Stream Number unique in the cluster (numbers range from 1 to 50), assign the MoH Audio Source
File to the number, and click Insert. The Play Continuously (Repeat) check box should always
be checked to ensure music loops when the end of the file is reached. If multicast capabilities are
necessary, you must check the Allow Multicasting check box. If the Play Continuously (Repeat)
and Allow Multicasting check boxes are both unchecked, the audio file stops playing when it
reaches the end and the network administrator must stop and start the server to reset the MoH
server.
The MoH Audio Source File Status window shows the conversion status and indicates whether the
audio file translated correctly or had errors.
Step 4: Set MoH Servicewide Settings

To set the MoH servicewide settings, open the Service Parameters Configuration window,
choose an MoH server, and choose the Cisco IP Voice Media Streaming App option. The Service
Parameters Configuration window for MoH appears, as shown in Figure 15-10.
Figure 15-10 Voice Media Streaming Application Service Parameters
The Supported MoH Codecs field is set to the codecs that are supported by the MoH servers in
the cluster. This field defaults to G.711 mu-law during the installation. You can enable multiple
codecs by pressing the Ctrl key while selecting the codecs.
The Default TFTPMOH IP Address field is set to the IP address or computer name of the default
MoH TFTP server.
NOTE Compressed codecs (such as G.729) are designed to handle Latin-based speech
patterns. Music on hold audio does not translate well through codecs of this nature and the
degradation in music quality is very perceptible.
Step 5: Find and Configure the Fixed Audio Source

One of your options for MoH is a fixed music source (attached via a sound card). This could be an
external CD player, jukebox, iPod, or any device that can connect through the Line In or
Microphone port of your sound card. You will need the exact name (case sensitive) of the fixed
audio source device to complete the fixed audio source configuration in Cisco CallManager.
NOTE CallManager MCS servers do not ship with sound cards. A list of supported sound
cards can be found in the CallManager SRND available at http://www.cisco.com/go/srnd.
To find the name of the fixed audio source, open Control Panel and choose Sounds and
Multimedia. Choose the Audio tab. You can use any sound-recording device name that appears in
the Preferred Device menu, shown in Figure 15-11. In addition, be sure to open the Recording
Control window, and click the Volume button in the Sound Recoding area. Verify that the Line In,
Microphone, or CD Audio check box is checked and the volume has been adjusted to an
appropriate level (this might require some testing in your IPT environment).
Figure 15-11 Fixed Audio Source Windows Sound Properties

To allow Cisco CallManager to use the fixed audio source, you must link the CallManager
configuration with the Windows sound device. After you have located the sound-recording device
name that appears in the Windows Control Panel, you must open the Audio Source File
Configuration window (by choosing Service > Media Resource > Music On Hold Audio
Source) and select the Fixed Audio Source (this will always be audio source ID 51). The window
Figure 15-12 Fixed Audio Source CallManager Configuration
The MoH Fixed Audio Source Device name is case sensitive and must be entered exactly as it
appears in the Sounds and Multimedia Properties window, including any spaces or symbols that
appear in the name. If the Allow Multicasting check box is checked, and the G.729 codec is
enabled, 5 to 7 percent of the CPU resources will be consumed. This setting is global for all MoH
servers. If the fixed audio source does not exist on a server, it cannot be used.
The Fixed Audio Source option affects all of the MoH servers that have an MoH fixed audio source
device by the selected name.
NOTE The FCC considers music on hold a type of broadcasting. Copyrighted music must
be licensed for prerecorded and live audio sources. Failure to comply with this law can result
in significant fines.
Step 6: Assigning Audio Source IDs

For your users to use the MoH sources you have configured, you must assign the audio source IDs
to the IPT devices. An audio source ID represents an audio source on the MoH server. The audio
source can be either a file on a disk or a fixed device from which a source stream obtains the
streaming data. Each audio source (represented by an audio source ID) can stream in unicast and
multicast mode.
The device that activates the hold will determine which audio source ID the caller will listen to.
You can assign the MoH audio source IDs to a group of devices by assigning them through the
device pool (choose System > Device Pool), through the individual IP Phones (choose Device >
Phone), or for specific directory numbers (choose Device > Phone > Directory Number). If you
assign the audio sources at both the device pool and IP Phone levels, the IP Phone settings override
the device pool. Likewise, the directory number assignments overrule both the IP Phone and
device pool settings. Figure 15-13 illustrates assigning the audio source IDs at the IP Phone level.
Figure 15-13 Assigning Audio Source IDs to an IP Phone

Media Resource Management

Media Resource Management (MGM) is an integral component of Cisco CallManager. The MRM
system controls and manages the media resources within a cluster, allowing all Cisco CallManager
servers within the cluster to share media resources.
The MRM enhances Cisco CallManager features by making it easier for Cisco CallManager to
deploy transcoder, annunciator, conferencing, MTP, and MoH resources. MRM distribution
throughout the Cisco CallManager cluster uses these resources to their full potential, which makes
the Cisco CallManager cluster more efficient and more economical.
The reasons that resources are shared include the following:
■ To enable hardware and software devices to coexist within a Cisco CallManager
■ To enable Cisco CallManager to share and access the resources that are available in the cluster
■ To enable Cisco CallManager to perform load distribution within a group of similar resources
■ To enable Cisco CallManager to allocate resources based on administrative preference
Media Resource Design

Cisco CallManager Media Resource Groups (MRGs) and Media Resource Group Lists (MRGLs)
provide a way to manage resources within a cluster.
Figure 15-14 shows the hierarchical ordering of media resources and how MRGs and MRGLs are
similar to route groups and route lists. When a device needs a media resource, it searches its own
MRGL first. If a media resource is not available, the device searches the default list, which
includes all of the media resources that have not been assigned to an MRG. After a resource is
assigned to an MRG, it is removed from the default list.
Figure 15-14 Media Resource Design

User Needs
Media
Media Resource IP
Resource
Manager IP
Assigned to Device
Media
Resource
Group List
First Choice Second Choice
Media Media
Resource Resource
Group Group
Media Resource Media Resource Media Resource Media Resource

1 2 3 1
NOTE By default, CallManager assigns all media resources in the cluster to the default
media resource group. If you decide to keep this default, you will have no control over what
media resources are used in your IPT environment.
Media Resource Groups

MRGs define logical groupings of media resources. Cisco CallManager provides a default list of
media resources. The default list of media resources includes all of the media resources that have
not been assigned to an MRG. If media resources have been configured but no MRGs have been
defined, all media resources are on the default list, and all media resources are available to all
Cisco CallManager servers within a given cluster.
Figure 15-15 shows how media resources are allocated to devices when they are listed in an MRG.
The default media resources shown for a Cisco CallManager include MOH1, MTP1, XCODE1,
XCODE2, and XCODE3. The MRM distributes the load evenly among the transcoder resources
in its default MRG for calls that require a transcoder resource.
Figure 15-15 Default Media Resource Group Behavior

Default MRG
XCODE1
XCODE2
XCODE3
Call 1 Call 2 Call 3 Call 4 Call 5 Call 6 Call 7
IP IP IP IP IP IP IP
XCODE1 XCODE2 XCODE3 XCODE1 XCODE2 XCODE3 XCODE1
This is the allocation order for incoming calls that require a transcoder resource:
■ Call 1 uses XCODE1.
Media Resource Group Configuration

Configuring an MRG is similar to configuring a route group. Choose Service > Media Resource
> Media Resource Group to display a Media Resource Group Configuration window, like that
shown in Figure 15-16. Enter a name and description for the MRG and then add the media
resources by moving them from the Available Media Resource section to the Selected Media
Resource section. To allow for a more flexible configuration, Cisco CallManager will allow you
to assign media resources to multiple MRGs.
Figure 15-16 Media Resource Group Configuration Window
Media Resource Group List Configuration

After you have added MRGs to the CallManager configuration, they can be grouped together and
assigned to the IP Phones through MRGLs. Configuring an MRGL is similar to configuring a route
list. Choose Service > Media Resource > Media Resource Group List to display the Media
Resource Group List Configuration window, shown in Figure 15-17. Enter a name for the MRGL
and choose the MRGs you want to add to the MRGL. Keep in mind that this is an ordered list. The
resources in the MRGs at the top of the list have the most priority.
Figure 15-17 Media Resource Group List Configuration Window
Assigning Media Resource Group Lists

There are two levels at which MRGLs can be assigned to devices. The higher-priority MRGL level
is configured at the device. For example, a Cisco IP Phone is configured in the Phone
Configuration window in Cisco CallManager Administration. The lower-priority level is an
optional parameter of the device pool. If an MRGL is not configured at the device level, it uses the
MRGL that is configured at the device pool level first, and then, if there are no resources available,
it tries to use resources in the default list. If a device does have an MRGL that is configured at the
device level, that MRGL is used first.
The last MRGL is the default MRGL. A media resource that is not assigned to an MRG is
automatically assigned to the default MRGL. The default MRGL is always searched and it is the
last resort if no resources are available in the device-based MRGL and the device pool MRGL or
if no MRGLs are configured at any level.
NOTE If all media resources have been placed in MRGs, there will be no resources
remaining in the default MRG pool. If a device does not have a particular resource made
available in its MRGL, access to that resource is restricted. Whatever services the missing
resource(s) support will be unavailable on the device.
MRGL Design Strategy: Grouping Resources by Type

Figure 15-18 shows how conference resources are allocated when resources are grouped by type
and the software conference resource group is listed after the hardware conference resource group
in the MRGL.
The media resources are assigned to three MRGs:
■ Hardware MRG—XCODE1, XCODE2, HW-CONF1, and HW-CONF2
■ Software MRG—MTP1, MTP2, SW-CONF1, and SW-CONF2
■ MOH MRG—MOH1 and MOH2
Figure 15-18 Grouping Resources by Type
Resource_List
Hardware MRG
XCODE1 B
1 XCODE2 RTP
HW-CONF1
HW-CONF2 IP IP
A
Software MRG
MTP1
2 MTP2 IP
SW-CONF1
C
SW-CONF2
MOH MRG
3 MOH1
MOH2
An MRGL called Resource_List was created and the MRGs assigned to it in this order: Hardware
MRG, Software MRG, and MoH MRG.
In this arrangement, when a conference is needed, Cisco CallManager allocates the hardware
conference resources first. The software conference resources are not used until all of the hardware
conference resources have been exhausted.
MRGL Design Strategy: Grouping Resources by Location

Figure 15-19 shows media resources that are grouped by location. Devices use the media resources
in their location before using the media resources at the central site (hub).
Figure 15-19 Grouping Resources by Location
Dallas_List SanJose_List
Dallas_MRG SanJose_MRG
XCODE1 XCODE2
1 HW-CONF1 1 HW-CONF2
MOH2 MOH3
Hub_MRG Hub_MRG
MTP1 IP IP MTP1
MTP2 Dallas San Jose MTP2
2 MOH1 2 MOH1
SW-CONF1 SW-CONF1
SW-CONF2 SW-CONF2
SanJose_MRG Dallas_MRG
XCODE2 XCODE1
3 HW-CONF2 3 HW-CONF1
MOH3 MOH2
This example is for multiple-site WAN deployments that use centralized call processing. All Cisco
CallManager and software resources are located at the central site. For devices at the Dallas and
San Jose locations, it is more efficient to use media resources that reside physically at the location
than to use a resource across the WAN.
Media resources are assigned to these three MRGs:
■ Hub_MRG—MTP1, MTP2, MOH1, SW-CONF1, and SW-CONF2
■ Dallas_MRG—XCODE1, HW-CONF1, and MOH2
■ SanJose_MRG—XCODE2, HW-CONF2, and MOH3
In this example, the network administrator has created a Dallas_List MRGL and assigned the
MRGs so that the resources are available in this order: local hardware resources first
(Dallas_MRG), software resources second (Hub_MRG), and distant hardware resources third
(SanJose_MRG).
The network administrator has also created a SanJose_List MRGL and assigned the MRGs so that
the resources are available in this order: local hardware resources first (SanJose_MRG), software
resources second (Hub_MRG), and distant hardware resources third (Dallas_MRG).
Lastly, the administrator has assigned an IP Phone in Dallas to use the Dallas_List MRGL and an
IP Phone in San Jose to use the SanJose_List MRGL. With this arrangement, the IP Phone in
Dallas will use the Dallas_List resources before using the central site or the SanJose_List
resources.
Summary
This chapter covered the concepts and configuration of Cisco CallManager media resources and
media resource management. Media resources provide services such as transcoding,
conferencing, music on hold (MoH), and media termination. Conference bridges resources can be
software or hardware solutions that allow ad hoc and Meet-Me conferences. Media termination
resources provide supplementary services to H.323 endpoints that do not support the H.323v2
OpenLogicalChannel and CloseLogicalChannel request features. Annunciator resources play
prerecorded announcements and tones. Transcoder resources convert an output stream from one
compression type to another, allowing devices using different codecs to communicate.
By default, all media resources are added to a default Media Resource Group (MRG) that cannot
be modified or deleted. As you create new MRGs and add media resources to them, those media
resources are removed from the default MRG. MRGs are organized into an ordered list in a Media
Resource Group List (MRGL), which is assigned to a device pool or devices to grant access to a
specific set of media resources in the cluster.
Review Questions
1. Which of these descriptions best describes MRGLs?

a. an ordered list of media resources
b. an ordered list of media gateways
c. an ordered list of Media Resource Groups
d. an ordered list of media servers
2. The Media Resource Manager manages which two resource types? (Choose two.)
a. call dispatcher service
b. Voice Media Streaming Application
c. transcoder
d. multiplexer
3. Which of the following is needed to add a 6608 hardware conference bridge into a Cisco
CallManager?
a. MAC address
b. IP address
c. port address
d. Meet-Me number
e. directory number
4. Annunciator resources require that which service be activated?
a. Cisco MoH Audio Translator
b. Cisco IP Voice Media Streaming Application
c. Cisco Messaging Interface
d. Cisco CDR Insert
e. Cisco TFTP
5. A user is having a conversation with her friend. A manager walks by and the user quickly
presses the conference button to make it look as if she were initiating a business-related
conference call. What music is the user’s friend hearing while the conference call is being
initiated?
a. whatever music is defined for the user hold option on the user’s device pool or IP Phone
b. whatever music is defined for the network hold option on the user’s device pool or IP
Phone
c. The user hears no music because conference calls do not have MoH features.
d. It depends on the route plan.
6. What is the default MoH streaming setting on a Cisco CallManager server?
a. unicast
b. multicast
c. broadcast
d. None, the MoH server must be configured before MoH functionality exists.
7. A user has an MRGL assigned at both the device pool and IP Phone level. When the user
presses the transfer button on their phone, the MGM searches for a resource based on which
criteria?
a. the MTP resource listed first in the first MRG of the device pool MRGL
b. the MTP resource listed first in the first MRG of the IP Phone MRGL
c. the MTP resource listed first in the last MRG of the device pool MRGL
d. the MTP resource listed first in the last MRG of the IP Phone MRGL
8. You have not created any MRGs or MRGLs in your Cisco CallManager cluster configuration.
How does the MGM manage the media resources that are available?
a. The media resources are unavailable to users until they have been assigned to an MRG.
b. The media resources are assigned to the default MRG, which cannot be modified or
deleted. The MRM will use a round-robin load-balancing system for these resources.
c. The media resources are assigned to the default MRG, which cannot be modified or
deleted. The MRM will use hardware-based resources first and software-based
resources second.
d. The media resources are assigned to the default MRG. The default MRG must then be
added to an MRGL, which is assigned to the end devices.
9. Which of the following media resource functions can the Cisco CallManager server NOT
perform?
a. annunciator
b. conference bridge
c. Media Termination Point
d. Music on Hold
e. transcoding
10. Which audio source ID is always reserved for the fixed audio source in the Cisco CallManager
cluster?
a. 1
b. 21
c. 51
d. 99
following topics:
■ Describing core Cisco IP Phone features of

Cisco CallManager, including hold, redial,
transfer, speed dialing and abbreviated
dialing, and Auto Answer
■ Describing enhanced IP phone features of

Cisco CallManager, such as multiple calls per
line appearance, Direct Transfer, Call Join,
and Immediate Divert
■ Defining standard and nonstandard softkey

templates and creating nonstandard softkey
templates and assigning them to Cisco IP
Phones
■ Configuring Cisco CallManager to enable

Call Park, Call Pickup, and Cisco Call Back

Barge and Privacy on a shared line

users to subscribe to IP phone Services from
their Cisco IP Phones
CHAPTER 16
Configuring User
Features, Part 1
Administrators need to have a working knowledge of the various options that are available for
Cisco IP Phones to ensure that all of the desired features and functions are available to users and
that they are properly configured. This chapter, the first of two on features, discusses many of
the Cisco IP Phone features that are available to users in a Cisco IP telephony solution. The
chapter includes a discussion of core and enhanced IP phone features: Call Park, Call Pickup,
Cisco Call Back, Barge, Privacy, and Cisco IP Phone Services. The chapter explains the purpose
of each feature and describes how to configure and use the feature.
Basic IP Phone Features

Cisco CallManager software extends enterprise telephony features and capabilities to packet
telephony network devices, such as Cisco IP Phones, media-processing devices, voice over IP
(VoIP) gateways, and multimedia applications. Four basic IP phone features do not require
configuration in Cisco CallManager and are activated when the user presses a softkey on the
IP phone:
■ Hold—Places an active call on hold. Hold requires no configuration, unless you want to use
music on hold (MoH). When you put a call on hold, the call remains active even though you
and the other party cannot hear one another. You can answer other calls while a call is on
hold. Engaging the hold feature generates music or a beeping tone (called “tone on hold”).
■ Redial—Redials the last number dialed. To redial the most recently dialed number, press
the Redial softkey. Doing so without lifting the handset activates the speakerphone or
headset. To redial a number from a line other than your primary line, select the desired line
button and then press Redial.
■ Transfer—Transfers an active call to another directory number (DN) through use of the
Transf softkey. This can be a blind transfer, where the call is cut over directly to another
extension, or a consultative transfer, where a temporary ad hoc conference call is created to
allow all parties to speak before a transfer occurs.
■ Call Waiting—Lets users receive a second incoming call on the same line without
disconnecting the first call. When the second call arrives, the user receives a brief call
waiting indicator tone.
348 Chapter 16: Configuring User Features, Part 1
Speed Dial and Abbreviated Dial Configuration

Speed dialing provides quick access to frequently dialed numbers. Abbreviated dialing, introduced
in Cisco CallManager Release 4.0, extends speed dial functionality by enabling a user to configure
up to 99 speed dial entries on a telephone. When a user lifts the handset, the AbbrDial softkey
appears. The user can access any speed dial entry by entering the appropriate index, either one or
two digits, followed by pressing the AbbrDial softkey.
You can configure speed dials or abbreviated dials in Cisco CallManager Administration. You can
access this configuration by clicking Device > Phone, selecting the phone you want to configure,
and clicking the Add/Update Speed Dials link in the upper-right corner. A window similar to that
shown in Figure 16-1 opens. Configure abbreviated dials just as you would configure speed dials.
Users (or administrators) can configure speed dials from the User Options web page (https://
<server IP address>/ccmuser).
Figure 16-1 Configuring IP Phone Speed Dials

Basic IP Phone Features 349
Auto Answer Configuration

Auto Answer is a feature that causes the speakerphone or headset to go off hook automatically
when an incoming call is received. You can program this feature on a telephone-by-telephone
basis. To configure Auto Answer, choose the device that you want to enable, and then choose Auto
Answer under the Directory Number Settings. A window appears similar to that shown in Figure
16-2. You can choose Auto Answer Off, Auto Answer with Headset, or Auto Answer with
Speakerphone. Using the Auto Answer with Headset feature works well for support desks because
it delays the call by one second and plays a tone in the ear of the user with the headset to notify
them of an incoming call.
Figure 16-2 Configuring Auto Answer
Call Forward and Configurable Call Forward Display Configuration

Call forwarding allows an administrator or user to configure a Cisco IP Phone so that all calls that
are destined for that IP phone ring at another telephone or go directly to voice mail. To access the
call forwarding options, choose Device > Phone, then select a phone and select a line to display
the call forwarding parameters shown in Figure 16-3.
Figure 16-3 Configuring Call Forwarding
The call forwarding options are as follows:
■ Forward All—The settings in this row of fields specify the forwarding treatment for calls to
this DN if the DN is set to forward all calls. Specify the following values:
— Voice Mail—Check this check box to use settings in the Voice Mail Profile
Configuration window. When this check box is checked, Cisco CallManager ignores
the settings in the Coverage/Destination and Calling Search Space fields.
— Coverage/Destination—This setting indicates the DN to which all calls are
forwarded. Use any dialable phone number, including an outside destination.
— Calling Search Space—This setting allows you to specify different calling
restrictions for call forwarding. Otherwise, the same calling restrictions applied in
the IP phone calling search space also apply to call forwarding.
■ Other call forwarding settings—Starting with Cisco CallManager Release 4.1, the
administrator can specify different call forwarding treatment based on whether the caller is
external or internal for no answer, busy, and no coverage conditions.
NOTE Call Forward Busy, Call Forward No Answer, and Call Forward No Coverage for
external and internal calls and the interaction of the No Coverage option is covered in detail in
Chapter 12, “Configuring Hunt Groups and Call Coverage.”
■ Configurable call forwarding display—Starting with Cisco CallManager Release 4.0, the
administrator can configure call forwarding information display options to the original dialed
number, to the redirected dialed number, or to both (these settings are not shown in Figure 16-3).
You can enable or disable the caller name or caller number and present this information to the
display of the forwarded party. The display option is configured for each line appearance.
Softkey Templates
Softkeys extend the functions of nearly all Cisco IP Phones (with the exception of the 7902). As
shown in Figure 16-4, softkeys are buttons along the side and bottom of the IP phone liquid crystal
display (LCD) that point to functions and feature options on the LCD screen. Softkeys change
depending on the status of the phone.
Figure 16-4 Softkeys Available on 7940, 7960, and 7970 Cisco IP Phones
Softkeys
Cisco CallManager provides softkey templates for administrator convenience. Softkey templates
group softkeys that are used for common call-processing functions and applications. You can use
these default softkey templates to provide standard softkey definitions for devices, or you can
create custom templates.
Cisco CallManager, starting with Release 4.0, includes these five standard softkey templates:
■ Standard IPMA Assistant
■ Standard User
■ Standard Feature
■ Standard IPMA Manager
■ Standard IPMA Shared Mode Manager
You cannot delete or modify these standard templates. However, you can create custom
(nonstandard) templates to meet the needs of your organization.
Adding Softkey Templates

To create a nonstandard softkey template, you must first copy a standard template and make the
modifications desired to this copy. Choose Device > Device Settings > Softkey Templates to
access softkey templates. The Find and List Softkey Templates window shown in Figure 16-5
appears. Choose the standard template and click the Copy button. The Softkey Template
Configuration window then displays the softkey template name, description, and applications that
are associated with the template. You must rename the template with a new descriptive name. After
you have entered a unique name, click the Insert button. The standard template is copied, and
when you choose Back to Find/List Softkey Templates, the new softkey template is displayed.
After the nonstandard template is made, application softkeys can be added or removed from the
template.
Figure 16-5 Standard Softkey Template Listing

Modifying Softkey Availability and Positioning

You can add or delete softkeys or modify softkey positions in a nonstandard softkey template to
customize the appearance of the softkeys on Cisco IP Phones. In the Softkey Templates field,
choose the template in which you want to add or delete softkeys or modify the softkey positions.
In the upper-right corner of the window, click the Configure Softkey Layout link. As shown in
Figure 16-6, the Softkey Layout Configuration window is displayed with call states on the left and
the Selected Softkeys pane on the right. You can select the softkeys that you want displayed for
each call state.
Figure 16-6 Modifying Softkey Availability and Positioning
To add softkeys, select a softkey from the Unselected Softkeys pane and click the right arrow to
move it to the Selected Softkeys pane. If the number of softkeys exceeds 16, an error message is
displayed that states that you must remove some of the softkeys before continuing. To delete
softkeys, select a softkey from the Selected Softkeys pane and click the left arrow to move it to the
Unselected Softkeys pane. Click Update after completing either procedure.
To modify softkey positions, use the up and down arrows to rearrange the positions of the selected
softkeys (the top position corresponds to the leftmost softkey on the IP phone). To save the
modifications that you have made to the template, click Update.
After making modifications to softkey templates, you must restart the devices that are using the
template.
Assigning Softkey Templates to Devices

You can assign softkey templates to devices in several ways. The template can be assigned in the
Device Pool Configuration window (System > Device Pool), through a user device profile (Device
> Device Settings > Device Profile), or on the device itself (Device > Phone).
For example, you might have a (standard or nonstandard) softkey template that you assign to a
device pool to configure the majority of phones in your cluster, a different softkey template that
you use for a feature that requires a device profile (for example, Extension Mobility), and a
different softkey template that includes the Barge, Privacy, and Immediate Divert softkeys that you
assign to a device belonging to a manager.
Figure 16-7 illustrates the Accounting User Softkeys template being assigned to a device pool.
Figure 16-7 Assigning Softkey Templates to Devices

Deleting Softkey Templates

Standard templates cannot be deleted. Only nonstandard templates can be deleted. If you want to
delete a nonstandard softkey template, the template cannot be in use by any device in the Cisco
CallManager system. If the softkey template is assigned to a device pool, user profile, or Cisco IP
Phone, you will receive an error message stating that the template is in use when you attempt to
delete it. You must remove the template from all devices before the template can be deleted.
To determine whether a softkey template is in use, click the Dependency Records link, located in
the upper-right corner of the Softkey Template Configuration window. By default, CallManager
disables the dependency records function because of its high CPU consumption on CallManager
servers.
NOTE Dependency records functionality causes high CPU usage when it is used. After it has
been enabled, this task executes at below-normal priority and might take time to complete
because of the dial plan size and complexity, CPU speed, and CPU requirements of other
applications.
To enable dependency records, follow these steps:
Step 1 Choose System > Enterprise Parameters.

Step 2 Scroll to the CCMAdmin Parameters area of the window.
Step 3 From the Enable Dependency Records drop-down menu, choose True.
A message box displays a message about the consequences of enabling the
dependency records. Read the information carefully before clicking OK.
Step 4 Click OK. The field displays True.
Step 6 Close the browser that you are using; then, reopen the browser. This action
makes the parameter take effect for the entire system.
Enhanced IP Phone Features

The following features are considered fairly new features to the IP Telephony environment. They
are supported in the CallManager 4.x versions and later.
Multiple Calls per Line Appearance

Cisco CallManager Release 4.0 and later enable multiple calls to exist on the same line. This
feature eliminates the need to create multiple instances of the same directory number in different
partitions to allow users to share a line and still be able to receive and place multiple calls out of
the same line. Cisco CallManager will now support up to 200 active calls on a single line and one
connected call per telephone at any time.
Two configuration settings enable multiple line appearances and are configured from the Directory
Number Configuration window, as shown in Figure 16-8:
■ Maximum Number of Calls—This setting configures the maximum number of calls

inbound or outbound per line appearance. You can configure up to a potential of 200 calls for
a line on a device, with the limiting factor being the total number of calls that are configured
on the device. As you configure the number of calls for one line, the number of calls that are
available for another line decrease. The default specifies 4.
■ Busy Trigger—This setting, which works in conjunction with Maximum Number of Calls
and Call Forward Busy (CFB), determines the maximum number of calls to be presented at
the line. If the maximum number of calls is set to 50 and the busy trigger is set to 40, then
incoming call 41 is rejected with a busy cause code (and will be forwarded if CFB is set). If
this line is shared, all the lines must be busy before incoming calls will be rejected.
Figure 16-8 Multiple Line Appearance Configuration

In Figure 16-8, the maximum number of calls is set to 4 and the busy trigger is set to 2. With this
configuration, four calls can be active at a given time. If two calls are active, and a third call comes
in, it is forwarded to the Call Forward Busy destination. The users, however, can place up to two
more outbound calls before the line is at capacity. Keep in mind that multiple lines are needed to
use various other features such as call transfer and conference.
Direct Transfer
Direct Transfer, which was introduced in Cisco CallManager 4.0, joins two established calls
(defined as a call in the hold or connected state) into one call and drops the feature initiator from
the call. Direct Transfer does not initiate a consultation call and does not put the active call on hold.
To implement Direct Transfer, the Direct Transfer initiator selects two calls at the Cisco IP Phone
and presses the DirTrfr softkey, shown in Figure 16-9. The two calls are joined immediately and
directly, and no conference resources are inserted. The initiating user is not included in the call
after the transaction is complete, and the call is released from the IP phone of the initiator.
Figure 16-9 Direct Transfer
The following steps illustrate the user sequence of events in a typical Direct Transfer call:
Step 1 Sam and Kim are participating in an active call.

Step 2 Sam asks Kim to transfer him to Mary.
Step 3 Kim puts Sam on hold.
Step 4 Kim calls Mary.
Step 5 Kim touches the line that Sam is using (Cisco IP Phone 7970 only) or uses
the Select softkey to select that line.
Step 6 Kim presses the DirTrfr key, and Sam and Mary are immediately
connected.
Call Join
Introduced in Cisco CallManager Release 4.0, the Call Join feature enables a user to link up to 15
established calls (for a total of 16) in a conference. Call Join does not create a consultation call
(does not put the active call on hold).
To implement Call Join, the user chooses an active or held call and, using either the rocker key
(Cisco IP Phones 7960 and 7940) and the Select softkey or the touch screen (Cisco IP Phone
7970), selects the appropriate line and then presses the Call Join softkey so that the selected calls
and the join initiator are joined in an ad hoc conference. Figure 16-10 illustrates this process. The
initiator can leave the Call Join session at any time, and the conference stays active. Only the
initiator of the Call Join session can add participants to the conference or drop them from it.
Figure 16-10 Call Join
Immediate Divert to Voice Mail

The Immediate Divert feature is a feature that was introduced in Cisco CallManager Release 4.0.
This feature allows you to immediately divert a call to a voice-messaging system. When the call
is diverted, the line becomes available to make or receive new calls.
Immediate Divert supports an incoming call in the call-offering (ring in), call-on-hold, or call-
active state. Immediate Divert supports an outgoing call in the call-on-hold or call-active state.
You can access the Immediate Divert feature by using the iDivert softkey, shown in Figure 16-11.
This softkey can be applied to any Cisco IP Phone that can accept softkeys. Configure this softkey
by using the Softkey Template Configuration window (discussed in the following section) of Cisco
Figure 16-11 Immediate Divert to Voice Mail
Immediate Divert requires the following components:
■ Cisco CallManager Release 4.0 or later
■ Cisco IP Phones (models 7905, 7912, 7920, 7940, 7960, or 7970)
Call Park
The Call Park feature allows you to put a call on hold so that it can be retrieved from another
telephone in the Cisco CallManager cluster.
For example, the ABC Department Store, which has an overhead paging system, is using the Call
Park feature. A call for an employee on the floor comes in to a cashier desk. The cashier can park
the call and announce the Call Park code on the overhead paging system, and the employee on the
floor can pick up the call by using the Call Park code on a nearby telephone.
If you are on an active call on your telephone, you can park the call to a Call Park extension by
pressing the Park softkey. The Cisco IP Phone will display the Call Park number that it assigned
to your active call on the IP phone (the number will display on screen for 10 seconds, by default).
Someone (or you) on another phone in your system can then dial the Call Park extension to retrieve
the call.
Before the Call Park feature functions in your IPT network, a Call Park number or range must be
configured for the cluster. When you invoke the Call Park feature, it is assigned a Call Park code.
A user uses this code to pick up the call from another Cisco IP Phone on the same Cisco
CallManager to which the original IP Phone is registered. When you assign the Call Park number
or range to a partition, you can limit access to the Call Park feature based on the device calling
search space. You should ensure that the Call Park number or range is unique throughout the Cisco
CallManager cluster.
You can configure the Call Park feature by choosing Feature > Call Park. Select the Add a New
Call Park Number hyperlink in the upper-right corner of the screen. The Call Park Configuration
window shown in Figure 16-12 appears. Configure the Call Park number or range (CallManager
accepts the same wildcards as you can use in a route pattern) and choose Insert.
Figure 16-12 Configuring Call Park
Call Pickup and Group Call Pickup

The purpose of Call Pickup is to enable a group of users who are seated near each other to cover
incoming calls as a group. When a member of the group receives a call and is not available to
answer it, any other member of the group can pick up the call from their own phone.
Three types of Call Pickup exist:
■ Call Pickup—Enables users to pick up incoming calls on any telephone within their own
group. When the users press the Call Pickup button or PickUp softkey, Cisco CallManager
automatically dials the appropriate Call Pickup number.
■ Group Call Pickup—Enables users to pick up incoming calls on any telephone within their
own group or in another group. Users press the Group Call Pickup button or GpickUp
softkey and dial the appropriate group number for Call Pickup. (The reason that users
manually enter a number with Group Call Pickup but not with Call Pickup is because more
than one group can exist and Cisco CallManager needs to know which one to dial. With Call
Pickup, in effect, there is only one number corresponding to one group.)
■ Other Group Call Pickup—Allows users to pick up incoming calls in a group that is
associated with their own group. This type of Call Pickup is covered in the next subtopic.
Other Group Call Pickup

Cisco CallManager Release 4.1(3) added support for Other Group Call Pickup. Other Group Call
Pickup allows users to pick up incoming calls in a group that is associated with their own group.
Cisco CallManager automatically searches for the incoming call in the associated groups to make
the call connection when the user activates this feature from a Cisco IP Phone. Use the softkey
OPickup for this type of Call Pickup.
When more than one associated group exists, the priority of answering calls for the associated
group goes from the first associated group to the last associated group. For example, groups A, B,
and C associate with group X, and the priority of answering calls goes to group A, then B, and then C.
Usually, within the same group, the longest alerting call (longest ringing time) is picked up first if
multiple incoming calls occur in that group. For Other Group Call Pickup, priority takes
precedence over the ringing time if multiple associated pickup groups are configured.
Both the idle and off-hook call states make the three softkeys Pickup, GPickup, and OPickup
available.
Call Pickup Configuration

To configure Call Pickup, you must first add and configure the Call Pickup number and then assign
the Call Pickup number to the desired DNs. Follow this procedure to add a Call Pickup number
and group in Cisco CallManager Administration:
Step 1 Choose Feature > Call Pickup.

Step 2 In the upper-right corner of the window, click the Add a New Call Pickup
Number link.
Step 3 The Pickup Group Configuration window opens, as shown in Figure 16-13.
Figure 16-13 Configuring Call Pickup
Step 4 Enter a unique pickup group name and unique pickup group number.
Step 5 Assign the pickup group to a route partition as desired.
Step 6 Click Insert to save the new Call Pickup Group number in the database.
Step 7 Access the directory number you want to assign to the Call Pickup Group
(Device > Phone > Select DN) and assign the Call Pickup Group, as shown
in Figure 16-14.
AutoCall Pickup
You can automate Call Pickup, Group Call Pickup, and Other Group Call Pickup by setting the
service parameter Auto Call Pickup Enabled to True (available in CallManager 4.1(3)). (Choose
Service > Service Parameters, and choose the publisher server and Cisco CallManager for the
service.)
When this parameter is enabled, Cisco CallManager automatically connects users to the incoming
call in their own pickup group, in another pickup group, or in a pickup group that is associated
with their own group after users press the appropriate softkey on the phone. This action reduces
keystrokes.
Figure 16-14 Assigning DNs to a Call Pickup Group
Automated Call Pickup connects the user to an incoming call in the user’s own group. When the
user presses the Pickup softkey on the phone, Cisco CallManager locates the incoming call in the
group and completes the call connection. If automation is not enabled, the user must press the
softkeys Pickup and Answer to make the call connection. You can also use this feature to connect
the user to an incoming call in another pickup group. The user presses the GPickup softkey on the
phone, and then dials the DN of the other pickup group. Upon receiving the DN, Cisco
CallManager completes the call connection. If automation is not enabled, the user must press the
softkeys GPickup and Answer and dial the DN of the other pickup group to make the call
connection.
Automated Other Group Call Pickup connects the user to an incoming call in a group that is
associated with the user’s own group. The user presses the OPickup softkey on the phone. The
Cisco CallManager automatically searches for the incoming call in the associated groups in the
sequence that the administrator entered in the Pickup Group Configuration window and completes
the call connection after the call is found. If automation is not enabled, the user must press the
softkeys OPickup and Answer to make the call connection.
Cisco Call Back

The Cisco Call Back feature allows you to receive call-back notification on your Cisco IP Phone
when a called-party line becomes available. To receive call-back notification, a user presses the
CallBack softkey upon receiving a busy or ringback tone. You can activate call-back notification
on a line on a Cisco IP Phone within the same Cisco CallManager cluster as your telephone. You
cannot activate call-back notification if the called party has forwarded all calls to another extension
(Call Forward All [CFA] feature).
For example, IP phone user A calls IP phone user B in the same cluster. If IP phone B is busy or
there is no answer, IP phone user A activates the Cisco Call Back feature through the CallBack
softkey. When IP phone B becomes available, IP Phone A receives an audible alert and visual
notification that the DN is available. Cisco CallManager remembers the dialed number, so IP
phone user A can then press the Dial softkey to reach IP phone user B.
NOTE The Cisco Call Back feature requires Cisco CallManager Release 3.3 or later and a
Cisco IP Phone that supports softkeys.
To configure Cisco Call Back, create a new softkey template (or modify one of your existing
templates). Next, configure the softkey layout by choosing the On Hook call state and the
CallBack option, as shown in Figure 16-15. Then, choose Ring Out, include the CallBack option
by making sure that it is at the top of the list, and click Update.
Figure 16-15 Configuring Call Back

Barge and Privacy

Using the Barge feature, users can add themselves to an existing call on a shared line. Two types
of Barge are available in Cisco CallManager Release 4.0:
■ Barge using built-in conference; Barge softkey—Barge uses the built-in conference
capability of the target IP phone. Barge also uses the Standard User or Standard Feature
softkey template (both contain the Barge softkey). When a Barge is being set up, no media
interruption occurs and the only display change to the original call is a spinning circle that is
displayed at the right side of the prompt status message window at the target device.
■ Barge using shared conference; cBarge softkey—Conference Barge (cBarge) uses a shared
conference bridge. No standard softkey template includes the cBarge softkey. To allow users
to access the cBarge softkey, the administrator must add it to a nonstandard softkey template
and then assign the softkey template to a device.
When you press the cBarge softkey, a Barge call is set up by means of the shared conference
bridge, if it is available. The original call is split and then joined at the conference bridge, which
causes a brief media interruption. The call information for all parties changes to Barge. The barged
call becomes a conference call with the Barge target device as the conference controller. The
conference controller can add more parties to the conference or can drop any party. When only two
parties are left in the conference, they experience a brief interruption and then are reconnected as
a point-to-point call, which releases the shared conference resources.
When the initiator uses Barge to join the call, it becomes a three-way call. If the initiator hangs up,
the original call remains active. If the target hangs up, the caller who used Barge and the other
party connect in a point-to-point call. If the other party hangs up, the original call and the barged
call are released.
The Privacy feature was introduced in Cisco CallManager Release 4.0. With Privacy,
administrators can enable or disable the ability of users with telephones that share the same line
(DN) to view call status and to barge the call. Administrators enable or disable Privacy for each
telephone.
The Barge and Privacy features have some restrictions, including the following:
■ Built-in Barge supports a three-way Barge maximum, G.711 voice, and the Cisco IP Phone
7940, 7960, and 7970 models.
■ Barge and Privacy require Cisco CallManager Release 4.0.
Barge requires a shared line appearance. Cisco CallManager considers a DN on more than one
device in the same partition to be a shared line appearance. One example of a shared line
appearance is when a DN appears on line 1 of a manager telephone and also on line 2 of an
assistant telephone. Another example of a shared line is a single incoming 800 number that is set
up to appear as line 2 on every help desk telephone in an office.
These guidelines are helpful when using shared line appearances with Cisco CallManager:
■ You can create a shared line appearance by assigning the same DN and partition to different
lines on different devices.
■ If other devices share a line, the words “Shared Line” are displayed in red next to the DN in
the Directory Number Configuration window in Cisco CallManager Administration.
■ If you change the calling search space, call waiting, or call forward and pickup settings on any
device that uses the shared line, the changes are applied to all of the devices that use that
shared line.
■ To stop sharing a line appearance on a device, you can change the DN or partition number for
the line and update the device. (Deletion removes the DN on the current device only. The
deletion does not affect the other devices.)
■ Do not use shared line appearances on any Cisco IP Phone that will be used with the Attendant
Console.
■ Do not use shared line appearances on any Cisco IP Phone 7960 that requires the Auto Answer
capability.
Configuring Barge
To configure Barge with a built-in conference bridge, follow these steps:
Step 1 Assign the Standard User or Standard Feature softkey template (both
contain the Barge softkey) to each device that accesses Barge by using the
built-in conference bridge.
Step 2 To enable Barge clusterwide for all users, choose Service > Service
Parameters for the Cisco CallManager service and set the Built-In Bridge
Enable clusterwide service parameter to On. Alternatively, configure Barge
for each telephone by setting the Built-In Bridge field in the Phone
Configuration window on the device itself.
Step 3 Set the Party Entrance Tone to True if you desire tones when a Barge
occurs.
NOTE To configure Barge using a shared conference resource (rather than the built-in IP
phone conference resources), just add the Conference Barge (cBarge) softkey to the Remote In
Use call state of the device’s softkey template.
Configuring Privacy
Recall that when Privacy is enabled, users on a shared line can enable or disable the ability of other
users on the shared line to view call status and to barge the call.
To configure Privacy, follow these steps:
Step 1 Set the optional Privacy Setting clusterwide service parameter for Cisco
CallManager to True.
NOTE Do not set this parameter if only a few users need access to Privacy (see Step 3).
Step 2 For each phone button template for which you want to enable Privacy, add
Privacy to one of the feature buttons, as shown in Figure 16-16.
Figure 16-16 Call Privacy Phone Button Template Configuration
Step 3 For each telephone user who wants to enable Privacy, choose On in the
Privacy drop-down menu in the Phone Configuration window. If you have
configured Privacy clusterwide, you can leave the Privacy setting at Default
or set it to Off to selectively disable privacy.
Step 4 For each telephone user who wants to enable Privacy, choose the phone
button template that contains the Privacy feature button that you created in
Step 2.
Figure 16-17 shows the phone display when the Privacy feature is assigned to a feature button.
When Privacy is enabled, the Privacy button changes display—a black circle appears inside the
Privacy field. Now, when the other user on the shared line goes off hook on the shared line, the
Barge softkey does not appear.
Figure 16-17 Call Privacy Phone Display
Privacy Disabled
Privacy Enabled
IP Phone Services
Cisco IP Phone Services include Extensible Markup Language (XML) applications that enable the
display of interactive content with text and graphics on Cisco IP Phone 7970, 7960, 7940, 7912,
and 7905 models.
NOTE Cisco IP Phones 7912 and 7905 support only text-based XML applications.
Using the Cisco IP Phone, you can deploy customized client services that users can interact with
from the keypad, the softkeys, or a rocker key and can use to display helpful information on the
IP phone.
A user can access a service from the supported phone model in two ways. The user can press the
button labeled Services to use a preconfigured phone button. When a user presses the Services
button on a Cisco IP Phone, a session is initiated and a menu of services that are configured for
the telephone appears. When the user selects a service from the listing, the telephone display is
updated.
In addition to adding a service so that it is available to users on their telephones, you can assign
the service to a phone button that is configured as a service URL button. This option gives the user
one-button access to the service without using the Services button on the IP phone. With Cisco
CallManager Release 4.0 or later, you can use any line or speed dial button for one-touch access
to selected XML services such as MyFastDials or to access critical XML applications such as
those that check inventory levels.
The following list provides examples of services that can be supplied to Cisco IP Phones:
■ Conference room scheduler
■ E-mail and voice-mail messages list
■ Daily and weekly schedule and appointments
■ Personal Address Book entries
■ Weather reports
■ Company news
■ Flight status
You can create customized Cisco IP Phone applications for your site by using the Cisco IP Phone
Services Software Development Kit (Cisco XML SDK).
Configuring IP Phone Services

You can add services to Cisco CallManager by using the Cisco IP Phone Services Configuration
window. After services are configured in Cisco CallManager Administration, users or
administrators can subscribe to these services for the devices to which they have access.
To add an IP phone Service, complete the following steps:
Step 1 Choose Feature > Cisco IP Phone Services.

Step 2 In the upper-right corner of the window, click the Add a New IP Phone
Service link. The Cisco IP Phone Services Configuration window is
displayed, as shown in Figure 16-18.
Figure 16-18 Cisco IP Phone Services Configuration Window
Step 3 Enter the appropriate settings. This list describes the information that must
be configured for each service:
• Service Name—Enter the name of the service as it will be displayed on
the menu of available services in the Cisco IP Phone User Options
application. Enter up to 32 characters for the service name.
• Service Description (optional)—Enter a description of the content that
the service provides to help users decide whether they want to subscribe
to the service. (This parameter is not shown in the figure because the
service has already been added.)
• Service URL—Enter the URL of the server where the Cisco IP Phone
Services application is located. Make sure that this server remains
independent of the servers in your Cisco CallManager cluster. Do not
specify a Cisco CallManager server or any server that is associated with
Cisco CallManager (such as a TFTP server or directory database
publisher server).
• Character Set—If you are using a language other than English for the
service name or description, choose the character set for that language.
Step 4 To add the service, click Insert.
TIP You can use the service URL shown in Figure 16-18 to obtain free XML services,
including stock quotes, weather reports, and news from Berbee, a Cisco partner.
To add the service URL button to an IP phone, follow these steps after you add the service to Cisco
CallManager:
Step 1 Customize a phone button template by configuring a Service URL button,

Figure 16-19 IP Phone Service Line Button Configuration
Step 2 Add the customized phone button template to the telephone.

Step 3 Choose Device > Phone.
Step 4 To locate a specific telephone, enter the search criteria and click Find.
Step 5 Choose the telephone to which you want to add a service URL button.
Step 6 On the upper-right side of the window, click the Add/Update Service URL
Buttons link.
Step 7 From the Service drop-down menu, choose the service that you want to add
to the telephone.
Step 8 To add the service to the telephone button, click Update, or click Update
and Close to add the service to the phone button and return to the Phone
Configuration window.
Services Phone Display Examples

The Cisco IP Phones have extensive XML display capabilities (especially the 7970 and 7971 models).
Here is a list of typical Cisco IP Phone Services displays, as demonstrated in Figure 16-20:
■ Menu—The menu enables the user to scroll through a list of menu items and make choices.
■ Text—The text from a web page can be delivered for the user to view.
■ Input—Input enables the user to enter information using the keypad. The 7970 and 7971 also
support touch screen input.
■ Image—Cisco IP Phone Services can deliver images in black and white and in shades of gray.
Color images are available over Cisco IP Phones with a color display.
■ Directory—Cisco IP Phone Services can deliver a corporate directory to enable users to

conveniently locate and dial other employees.
■ Graphical—Cisco IP Phone Services can display graphics as well as text.
Figure 16-20 IP Phone Service Display Examples
Menu Text Input
Image Directory Graphical

Summary
This chapter covered the concepts and configuration of Cisco CallManager end-user features. The
Cisco CallManager core features include hold, redial, and transfer. By default, these features are
available on any Cisco IP Phone. In addition, you can configure additional features to be made
available through an IP phone’s softkey template. The softkey templates choose the availability
and order of the feature buttons on an end-user IP phone.
In addition to supporting core features, the Cisco CallManager supports enhanced features,
including Call Join, multiple calls per line appearance, and Immediate Divert. Call Park enables a
call to be picked up on a different phone than the one it came in on, whereas Call Pickup enables
users to cover incoming calls as a group. Configuring Cisco Call Back allows a user to receive call-
back notification when a called-party line becomes available. Barge adds a user to a call that is
already in progress, whereas Privacy disables the Barge capability.
Finally, Cisco IP Phone Services enable the display of interactive content with text and graphics
on XML-capable Cisco IP Phones.
Review Questions
1. What are two requirements for users to be able to view weather reports, airline arrival and
departure times, stock quotes, or other Cisco IP Phone Services that the system administrator
has made available to them?
a. Subscribe to the service in the User Options pages.
b. Enable CTI Application Use.
c. Go to http://<server_name>/Services.asp.
d. Press the Services button on the Cisco IP Phone.
2. Which three of these areas can you use to assign a softkey template to a device?
a. Location
b. Device defaults
c. Device pools
d. User profile
e. Region
f. Device
3. Which phone feature requires no administrator configuration?

a. Call Park
b. Transfer
c. Barge
d. Privacy
4. For which softkey is the initiating user NOT included in the call after the transaction is
completed?
a. DirTrfr
b. Barge
c. cBarge
d. Join
5. Barge requires which of the following?
a. Privacy
b. Shared line appearance
c. Attendant Console
d. Standard User softkey template
e. Cisco IP Phone 7960
6. Which of the following features allow a user to answer a ringing phone automatically through
their speakerphone?
a. Call Pickup
b. Group Call Pickup
c. Other Call Pickup
d. Auto Answer
7. You have configured the shared extension 1005 to support 30 maximum calls, and to initiate
a busy trigger when the calls have reached 15. Michael is using the shared DN 1005 and picks
up the extension to make an outgoing call. The current number of individuals using the line
is 15. What does Michael hear?
a. A fast-busy signal
b. An automated message
c. A dial tone
d. A single beep, followed by silence
8. You would like to provide Call Back capabilities to everyone in your organization. All IP
phones are configured to use the Standard User softkey template. What would be the easiest
way to provide Call Back capabilities to the users?
a. Add the Call Back softkey to the Ring Out and On Hook call states of the standard user
template.
b. Create a new softkey template, add the Call Back softkey to the Ring Out and On Hook
call states, and assign the softkey template to all users in your organization.
c. Configure the Call Back XML service and allow users to subscribe from the CCMUser
web pages.
d. Assign the users to the Standard IPMA Manager softkey template.
9. Which of the following accurately describes the Group Call Pickup feature?
a. Group Call Pickup allows users to pick up a ringing phone located within their Call
Pickup Group.
b. Group Call Pickup allows users to pick up a ringing phone located within their own Call
Pickup Group or another Call Pickup Group.
c. Group Call Pickup allows users to pick up a ringing phone in their pickup group or
another pickup group without dialing a group number.
d. Group Call Pickup allows users to pick up a call on hold by dialing the appropriate
extension.
10. A user notices that their shared line appearance is currently in use. Out of curiosity, the user
presses a softkey. Immediately, the user is in a conference call with the parties currently using
the shared line appearance. Which softkey did the user press?
a. Barge
b. Privacy
c. cBarge
d. iBarge
following topics:
■ Configuring Cisco CallManager Extension

Mobility to enable users to log in to any Cisco
IP Phone and obtain their default device
profiles
■ Configuring FAC and CMC to manage call

access and accounting
■ Configuring Call Display Restrictions to

selectively allow or restrict the display of
calling- or connected-line information
■ Configuring MCID to report a call of a

malicious nature
■ Describing the purpose of MLPP

CHAPTER 17
Configuring User
Features, Part 2
Administrators need to have a working knowledge of the various options that are available for
Cisco IP Phones to ensure that all of the desired features and functions are available to users and
that they are properly configured. This lesson, the second of two parts on features, discusses
many Cisco IP Phone features that are available to users in a Cisco IP telephony solution. The
chapter includes a discussion of Cisco CallManager Extension Mobility, Call Display
Restrictions, Forced Authorization Codes (FAC), Client Matter Codes (CMC), Malicious Call
Identification (MCID), and Multilevel Precedence and Preemption (MLPP) features.
Cisco CallManager Extension Mobility

Cisco CallManager Extension Mobility is an approach to organizing work environments so that
workspaces—including offices, cubicles, and desks—are not permanently assigned to
individuals. Instead, employees “check into” an office space by performing a login process at
the Cisco IP Phone where they want to receive their calls. Following the login, the IP Phone is
assigned the appropriate direct telephone number with all of its characteristics (ring type, speed
dialing, and so on). Environments where employees do not routinely conduct business in the
same office space every day, such as sales offices, commonly use Cisco CallManager Extension
Mobility. It is often referred to as “hoteling.”
Cisco CallManager provides a standard login service. The login service can be expanded by
using a set of Extensible Markup Language (XML) requests over HTTP. Third parties
(including customers and integrators) can replace the login user interface with one of their own
design. For example, capabilities can be added to the standard login service such as
authenticating via smart card readers or automating the login process according to a “desk
sharing” web application.
Because Cisco CallManager Extension Mobility is enabled by XML-based services, it is

available only on Cisco IP Phones.
Extension Mobility Example

Figure 17-1 illustrates an example of Extension Mobility.
Figure 17-1 Extension Mobility in Action

ADP000011112222 Terry
7000
1111
IP IP 7001
Autogenerated Logout Login

Device Profile
IP
SEP000011112222
Terry has configured a user device profile called Terry. This profile has a line appearance with a
directory number (DN) of 7000 and another with a DN of 7001. Terry is logging in to device
SEP000011112222, which has an autogenerated device profile, ADP000011112222, as its default
device profile. The default device profile is configured with a single-line appearance with the DN
of 1111.
When Terry logs in, the device restarts and displays lines 7000 and 7001. It no longer has line 1111
assigned to it. All of the speed dials and services configured for Terry replace those that are
normally on SEP000011112222.
When Terry logs out, the device restarts and loads the autogenerated device profile.
Configuring Extension Mobility

Configuring Extension Mobility is no small chore. In earlier versions of Cisco CallManager,
Extension Mobility was only available by purchasing separate server software. In CallManager
4.x versions, Cisco has integrated these server functions into a CallManager service managed by
the Tomcat Web Application Manager. Your first step in configuring Extension Mobility is to use
the Cisco CallManager Serviceability tool Service Activation to activate the Cisco CallManager
Extension Mobility service, as shown in Figure 17-2.
Figure 17-2 Extension Mobility Service Activation
TIP The Tomcat Web Application Manager is an open-source application manager originally
designed for the Apache web server. It allows you to start and stop applications (or Java servlets)
without starting and stopping the entire web server. It can be accessed at http://
<CCM_Server_IP>/manager/list.
Using the Tomcat Manager window, stop and start the Cisco CallManager Extension Mobility
service at http://<Cisco Extension Mobility server>/manager/list, where Cisco Extension Mobility
server specifies the IP address of the server that has the Cisco CallManager Extension Mobility
service running on it. Figure 17-3 illustrates the Tomcat Web Application Manager.
Figure 17-3 Tomcat Web Application Manager
When the service is activated, configure the following elements to enable Cisco CallManager
Extension Mobility:
■ Cisco CallManager Extension Mobility service parameters
■ A device profile for a Cisco IP Phone model
■ A device profile for a Cisco IP Phone user
■ A new user
■ Add the Extension Mobility service
■ Update Cisco IP Phones to support Extension Mobility
To avoid problems with deploying Cisco CallManager Extension Mobility, be sure to follow these
configuration guidelines:
■ Configure a device profile default for each Cisco IP Phone model in a cluster that you want
to support Cisco CallManager Extension Mobility.
■ If you want to enable all IP Phones within a Cisco CallManager cluster with Cisco
CallManager Extension Mobility, do not allow the users to control these telephones.
— When users go to their Cisco CallManager User Options web page to change their
services, they must choose Device Profiles from the Select a Device to Configure
drop-down list. They cannot control an individual IP Phone or modify the settings
for an individual IP Phone.
— As administrator, you can change the services for an IP Phone by using Cisco
CallManager Administration. After making the changes, if you update the
configuration using the main menu (not the popup menu), you must reset the IP
Phone for the changes to take effect. This action ensures that the new snapshot is
stored as the logout profile.
■ If a particular user controls a device, for example, the user’s office telephone, do not allow
anyone else to log in to that device.
Cisco CallManager Extension Mobility Service Parameters

Figure 17-4 shows the service parameters for the Cisco CallManager Extension Mobility service.
Figure 17-4 Extension Mobility Service Parameters

To access this window, choose Service > Service Parameters and from the Service drop-down
menu, choose Cisco Extension Mobility. Table 17-1 defines the settings in the Service Parameters
Configuration window.
Table 17-1 Cisco Extension Mobility Service Parameters
Enforce Maximum Login When set to True, this parameter enforces a maximum login time.
Time If set to False, no time limit is set on logins.
Maximum Login Time This parameter is the maximum amount of time a user is allowed to
(Hours:Minutes) remain logged in to a device. The maximum allowable value is 168:00,
entered as HHH:MM. Note that :MM is also an acceptable format for
durations under 1 hour. This parameter is ignored if the Maximum
Login Time parameter is set to False.
Maximum Concurrent This parameter specifies the maximum number of login or logout
Request operations that can occur simultaneously. This maximum prevents the
Cisco CallManager Extension Mobility service from consuming
excessive system resources.
Multiple Login Behavior This parameter specifies the behavior for multiple attempted logins by
the same user on different devices. The choices are to allow, not to
allow, and to cause a previous login to automatically log out.
Alphanumeric User ID Choose True to allow Cisco CallManager to accept alphanumeric
Cisco CallManager Extension Mobility logins or False to force
numeric logins only.
Remember the Last User When set to True, this parameter remembers the last user to log in.
Logged In
Configuring a Default Device Profile

Device profiles are very similar to user profiles in Microsoft administration. A user profile
consisted of a user’s entire workstation environment: the desktop background, the screen saver,
the files on the desktop, and so on. In the IPT environment, a device profile defines the common
settings for an IP Phone: the phone button template, the music on hold source, the softkey
template, and so on.
The device profile default is a clusterwide default used for each Cisco IP Phone that will be
supported by Cisco CallManager Extension Mobility. The IP Phone takes on the device profile
default whenever a user logs in to an IP Phone model for which the user has no device profile. To
access this window, from the Cisco CallManager Administration, choose Device > Device
Settings > Device Profile Default. From the left column, click the device profile, or if adding a
new profile, click Add a New Device Profile Default, and the Device Profile Default
Configuration window shown in Figure 17-5 appears.
Figure 17-5 Extension Mobility Default Device Profile
If you do not choose an audio source in the User Hold Audio Source field, Cisco CallManager
uses the audio source that is defined in the device pool. If the device pool does not specify an audio
source ID, the system default is used.
The User Locale field identifies a set of detailed information to support users, including language
and font. Cisco CallManager makes this field available only for IP Phone models that support
localization.
Configuring User Device Profiles

After you have created the default device profile(s) for the IP Phones, you must now create your
user profiles. These profiles will be capable of moving between Cisco IP Phones as the user logs
in and out of various devices. These profiles correspond on a 1:1 basis with Extension Mobility
users in your network. The user device profile contains attributes such as name, description, phone
button template, expansion modules, directory number, subscribed services, and speed dial
information.
To access this window from Cisco CallManager Administration, choose Device > Device Settings
> Device Profile. To add a new user device profile, choose Add a New User Device Profile. The
User Device Profile Configuration window shown in Figure 17-6 appears.
Figure 17-6 Creating User Device Profiles
From here, you can select the user device type and add DNs to the profile, just like you were
configuring a user IP Phone.
NOTE Device profiles are created for specific devices and only work on specific devices. For
example, a device profile created for a 7960 IP Phone will not be able to log in to a 7940 or 7970
IP Phone.
Associating Users with Device Profiles

New users must be configured before they can use Cisco CallManager Extension Mobility. To add
a new user for Cisco CallManager Extension Mobility, follow these steps:
Step 1 In Cisco CallManager Administration, choose User > Add a New User.
The new user creation window appears, as shown in Figure 17-7.
Figure 17-7 Adding a New User
Step 2 In the Add a New User window, enter the first name, last name, and
username.
Step 3 In the User Password and Confirm Password fields, enter a password.
Step 4 In the PIN field, enter a numeric personal identification number (PIN).
Confirm the PIN.
Step 5 Enter the user telephone number.
Step 6 Click Insert, and the window refreshes. From the left pane, choose Cisco
Extension Mobility.
Step 7 Associate the user device profile with the user account, as shown in Figure
17-8, and then click the Update Selected button.
Figure 17-8 Associating Device Profiles with User Accounts
Creating the Extension Mobility Service

Cisco implemented the Extension Mobility feature as an additional service in Cisco CallManager.
Because of this, for the IP Phones to support Extension Mobility, you must make it available as a
service in the Cisco CallManager and add the new service as an IP Phones subscription. To
accomplish this, perform the following steps:
Step 1 In Cisco CallManager Administration, choose Feature > Cisco IP Phone

Services.
Step 2 Click Add a New IP Phone Service in the upper-right corner of the
window.
Step 3 Choose a logical name such as “Extension Mobility” for the service name.
Step 4 As shown in Figure 17-9, in the Service URL, enter the following string:
http://<CCM_IP>/emapp/EMAppServlet?device=#DEVICENAME#
Figure 17-9 Creating the Extension Mobility Service
CAUTION The Extension Mobility service URL must be entered exactly as shown,
substituting the IP address of the CallManager running the Extension Mobility service for
CCM_IP. This URL is case sensitive.
TIP To provide redundancy for a Cisco IP Phone service, use a CallManager hostname rather
than an IP address. If that CallManager fails, the services will automatically failover to a
secondary Cisco CallManager. Be sure you have configured the IP Phone for DNS support if
you choose to do this.
Step 5 Click the Insert button. The Extension Mobility service is now saved.
Updating Cisco IP Phones to Add Extension Mobility Support

The final step in configuring Extension Mobility is to configure the Cisco IP Phones with the
correct service and profile information. If you do not subscribe the phones to the Extension
Mobility service, they will be stuck on the default profile without users having the ability to log in
with their custom profiles. To subscribe an IP Phone to the Extension Mobility service, perform
the following steps:
Step 1 Enter the device configuration mode for the Cisco IP Phone you want to
configure for Extension Mobility support.
Step 2 Click the Subscribe/Unsubscribe Services link in the upper-right of the
screen.
Step 3 Using the drop-down menu, select the Extension Mobility service you
created and click Continue.
Step 4 Click Subscribe to add the service to the IP Phone configuration.
Step 5 On the phone configuration window, scroll to the bottom and click the
Enable Extension Mobility Feature check box.
Step 6 At the Log Out Profile field, select Use Current Device Settings to use
the default profile you configured, as shown in Figure 17-10.
Figure 17-10 Configuring the IP Phone for Extension Mobility

NOTE Cisco CallManager gives you two options for the Log Out Profile field: Use Current
Device Settings, which uses the default device profile you configured for the IP Phone, or Select
a User Device Profile, which allows you to configure a unique user profile for the IP Phone.
Cisco strongly advises against using a user profile for this purpose.
Step 7 Click Update to save the changes.
NOTE You should also modify the Extension Mobility parameters on the default device
profile for the service subscription to be available at all times.
Client Matter Codes and Forced Authentication Codes

Forced Authentication Codes (FACs) and Client Matter Codes (CMCs) allow you to manage call
access and accounting. CMCs are often referred to as account codes and assist with call
accounting and billing for billable clients. FACs regulates the types of calls that certain users can
place.
The CMC feature benefits law offices, accounting firms, consulting firms, and other businesses or
organizations that need to track the length of the call for each client. To use the CMC feature, users
must enter a client matter code to reach certain dialed numbers.
You can use FAC for colleges, universities, or any business or organization when limiting access
to specific classes of calls proves beneficial. Likewise, when you assign unique authorization
codes, you can determine which users placed calls.
FAC Concepts
For each user, you specify an authorization code, and then you enable FAC for relevant route
patterns by checking the appropriate check box and specifying the minimum authorization level
for calls through that route pattern.
When you enable FAC through route patterns in Cisco CallManager Administration, users must
enter an authorization code to reach the intended recipient of the call. When a user dials a number
that is routed through a FAC-enabled route pattern, the system plays a tone that prompts for the
authorization code. Figure 17-11 illustrates the FAC process, which follows these steps:
Step 1 A user dials a number that goes to a FAC-enabled route pattern.

Step 2 Cisco CallManager tells the phone to play the tone.
Step 3 The user enters the authorization code, followed by the # key.
NOTE The pound sign (#) at the end of the client matter code cancels the interdigit timeout.
If users do not append the # to the authorization code or client matter code, the system waits for
the T302 timer to extend the call (System > Service Parameters for the Cisco CallManager
service). The default for the T302 timer is 15 seconds.
Step 4 The call is extended to the exiting gateway.

Step 5 CallManager generates a Call Detail Record (CDR) flagged with the FAC
number.
Figure 17-11 FAC Process

Route Pattern 9.5622XXX
Require
X
Authorization Code
Authorization Level 3
95622001
Play Tone Extend Call to Gateway V
888# (FAC Code)
User A
In Cisco CallManager Administration, you can configure various levels of authorization. Users are
given codes that assign them a certain authorization level. Route patterns are then assigned
authorization levels. If the user authorization code does not meet the level of authorization that is
specified to route the dialed number, the user receives a reorder tone. If the authorization is
accepted, the call is placed. The name of the authorization writes to Call Detail Records (CDRs),
so that you can organize the information by using CDR Analysis and Reporting (CAR), which
generates reports for accounting and billing.
To implement FAC, you must devise a list of authorization levels and corresponding descriptions
to define the levels. You must specify authorization levels in the range from 0 to 255. Cisco allows
authorization levels to be arbitrary, so that you define what the numbers mean for your
organization. For example, you could configure authorization levels as follows:
■ Configure an authorization level of 10 for interstate long-distance calls in North America.
■ Because intrastate calls often cost more than interstate calls, configure an authorization level
of 20 for intrastate long-distance calls in North America.
■ Configure an authorization level of 30 for international calls.

CMC Concepts
You apply CMC through route patterns, and you can configure multiple client matter codes. When
a user dials a number that is routed through a CMC-enabled route pattern, a tone prompts the user
for the client matter code. When the user enters a valid code, the call is placed; if the user enters
an invalid code, a reorder tone is played. CMC is written to the CDR, so you can collect the
information by using CAR, which generates reports for client accounting and billing.
TIP The FAC and CMC features are very similar and often confused. Cisco created the FAC
primarily to prevent toll fraud and to limit and track the users able to make calls by authorization
levels. CMC is a nearly identical feature allowing organizations to track calls based on the CMC
code dialed rather than authorization level.
Figure 17-12 shows a basic CMC call where the route pattern 8.@ is configured to require the user
to enter a client matter code:
Step 1 When the user dials 8-214-555-0134, the dialed string matches the 8.@
route pattern.
Step 2 Cisco CallManager plays a “zip-zip” tone to prompt the user to enter the
client matter code associated with the dialed string, in this example, 1234.
Step 3 If the user enters 1234 followed by the # key, the call is immediately
extended to the voice gateway. If the user does not enter a code or enters the
wrong code, the user hears a reorder tone, and the call is not extended.
Step 4 Because the user entered a valid code, Cisco CallManager extends the call
to the voice gateway for call completion.
Step 5 Cisco CallManager generates a CDR with the associated client matter code
for client-tracking and reporting purposes.
Figure 17-12 CMC Process
Route Pattern 8.@
Require Client
X
Matter Code
Code 1234
8-214-555-0134
Play Tone Extend Call to Gateway V
1234# (Code)
User A
CMC and FAC can be implemented together for a given route pattern. For example, you can allow
only certain users to have authorization to place certain long-distance calls, and then require the
user to enter a CMC for that call. The tones for CMC and FAC sound the same to the user, so the
feature tells the user to enter the authorization code after the first tone and enter the account code
after the second tone.
FAC and CMC Configuration

To implement CMC or FAC, you can perform the following steps:
Step 1 If using CMCs, create a document with a list of CMCs and associated client
names that you want to track. If using FACs, create a document listing the
authorization levels you want to create and the authorization levels required
for restricted route patterns.
Step 2 Insert the CMC or FAC codes by using Cisco CallManager Administration
(choose Feature > Client Matter Code or Feature > Forced
Authorization Code).
Step 3 To enable FAC or CMC, update route patterns by selecting Require Client
Matter Code or Require Forced Authorization Code and entering the
level of FAC required.
Step 4 Update dial plan documents as necessary.
Step 5 Provide information to users and explain how the feature works.
Creating CMC Codes

Complete the following steps to add CMCs in Cisco CallManager Administration:
Step 1 In Cisco CallManager Administration, choose Feature > Client Matter

Code.
Step 2 In the upper-right corner of the window, click the Add a New Client
Matter Code link.
Step 3 In the Client Matter Code field, enter a unique code of no more than 16
digits that users will enter when placing a call, as shown in Figure 17-13.
The client matter code displays in the CDRs for calls that use this code.
Figure 17-13 CMC Configuration
Step 4 In the Description field, enter a name of no more than 50 characters. This
optional field associates a client code with a client.
Creating FAC Codes

Complete the following steps to add FACs in Cisco CallManager Administration:
Step 1 In Cisco CallManager Administration, choose Feature > Forced

Authorization Code.
Step 2 In the upper-right corner of the window, click the Add a New Forced
Authorization Code link.
Step 3 In the Authorization Code Name field, enter a logical name representing
the authorization, as shown in Figure 17-14.
Figure 17-14 FAC Configuration
Step 4 In the Authorization Code field, enter the code users will dial to obtain this
level of authorization.
Step 5 In the Authorization Level field, enter the level of authorization the user
obtains if they enter the authorization code.
Enabling Route Patterns for FAC and CMC

Complete the following steps to enable route patterns for FAC and CMC:
Step 1 In Cisco CallManager Administration, choose Route Plan > Route/Hunt

> Route Pattern.
Step 2 Open the properties window for the route pattern you want to modify.
Step 3 If you are configuring the route pattern for CMC, check the Require Client
Matter Code check box, as shown in Figure 17-15.
Step 4 If you are configuring the route pattern for FAC, check the Require Forced
Authorization Code check box and enter the level of authorization
required to access this route pattern.
Figure 17-15 Route Pattern Configuration
Call Display Restrictions

The Call Display Restrictions feature allows you to selectively display or restrict the calling and
connected (called party) line information. A hotel environment frequently requires this
functionality and might have the following needs:
■ For calls between a guest room and the front desk, both the room and the front desk should
see both the calling-party information and the called-party information.
■ For calls between guest rooms, the rooms should not see the call information of other rooms
displayed.
■ For calls between guest rooms and other hotel extensions (such as a clubhouse), only the
rooms should see the call information displayed.
■ For external calls from the public switched telephone network (PSTN) to the front desk or
guest rooms, the call information of the caller should not be displayed if the display settings
are restricted.
■ For all calls to the front desk, the call information of internal calls should be displayed.
In Cisco CallManager Release 4.0 and later, the calling-party and connected-party presentation
can be controlled by the configuration at the translation pattern level. Starting with Cisco
CallManager Release 4.1, this functionality was extended by adding a new setting at the device
level that enables the device to ignore the caller ID restrictions of the other party for internal calls.
Understanding Calling Line ID Presentation

Cisco CallManager uses the Calling Line ID Presentation field information as a supplementary
service to allow or restrict display of the telephone number of the originating caller on a call-by-
call basis. Choose one of the following options to allow or restrict the display of the telephone
number of the calling party on the IP Phone display of the called party for this translation pattern,
shown in Figure 17-16:
■ Default—This option does not change the presentation of the calling-party identification
information.
■ Allowed—Cisco CallManager allows the display of the calling-party name or number.
■ Restricted—Cisco CallManager blocks the display of the calling-party name or number.
Figure 17-16 Calling Line ID Presentation Restrictions in Translation Patterns

If the incoming call goes through a translation pattern or route pattern and the Calling Line ID
Presentation setting is allowed or restricted, the system modifies the calling-line presentation with
the translation or route pattern setting.
Understanding Connected Line ID Presentation

Cisco CallManager uses the Connected Line ID Presentation field information as a
supplementary service to allow or restrict the display of the telephone number of the called party
on a per-call basis. Choose one of the following options to allow or restrict the display of the
telephone number of the connected party on the telephone of the calling party for this translation
pattern, as shown in Figure 17-16:
■ Default—This option does not change the presentation of the connected-line identification
information.
■ Allowed—Cisco CallManager allows the display of the connected-party name or number.
■ Restricted—Cisco CallManager blocks the display of the connected-party name or number.
If the incoming call goes through a translation or route pattern and the Connected Line ID
Presentation field is set to Allowed or Restricted, the system modifies the connected-line
presentation indicator with the translation or route pattern setting.
Understanding Ignore Presentation Indicators

The Ignore Presentation Indicators check box, introduced in CallManager 4.1, gives you the
flexibility to overrule internal caller ID restrictions on a device-by-device basis. To activate this
check box, access the IP Phone where you want to remove caller ID restrictions and place a check
in the Ignore Presentation Indicators (Internal Calls Only) check box, shown in Figure 17-17.
When this feature is active, the following caller ID modifications take place:
■ Cisco CallManager always displays the call information of the remote party if the other party
is internal, regardless of the other calling- and called-party presentation settings.
■ Cisco CallManager does not display the call information of the remote party if the other party
is external and the display presentation is restricted.
Figure 17-17 Setting Ignore Presentation Indicators Option for Select Devices
Ignore Presentation Indicators Option
Configuring Caller ID Restrictions

Because each IPT environment can be so different, it is impossible to create exact caller ID
restriction steps that apply to all organizations. Instead, complete the following general steps if
you want to use Call Display Restrictions:
Step 1 Configure partitions and calling search spaces before you add a translation
pattern.
Step 2 Configure translation patterns with different levels of display restrictions.
Step 3 Check the Ignore Presentation Restriction (Internal Calls Only) check
box for IP Phones such as lobby phones when you want to ensure that the
call information display for internal calls is always visible.
Step 4 Configure individual, associated translation patterns for each individual
Call Park directory number to work with the Call Park feature.
Malicious Call Identification 399
Malicious Call Identification

Malicious Call Identification (MCID), available starting with Cisco CallManager Release 4.0, is
an internetwork service that allows users to initiate a sequence of events when they receive calls
with a malicious intent from another network (typically, the PSTN). The user who receives a
disturbing call can invoke the MCID feature by using a softkey or feature code while connected
to the call. The MCID service immediately flags the call as a malicious call with an alarm
notification to the Cisco CallManager administrator. The MCID service flags the CDR with the
MCID notice and sends a notification to the off-net PSTN that a malicious call is in progress.
The MCID service is an ISDN PRI service, when using PRI connections to the PSTN. The MCID
service includes two components:
■ MCID-O—An originating component that invokes the feature at the request of the user
(victim) and that sends the invocation request to the connected network
■ MCID-T—A terminating component that receives the invocation request from the connected
network and responds with a success or failure message that indicates whether the service can
be performed
Typically, each function runs in separate network entities, and the two service components
communicate with each other to allow two networks to identify a call as malicious.
Cisco CallManager supports only the originating component at this time.
TIP MCID is a PRI-based service. If your organization does not use PRI PSTN connections,
MCID can still flag calls with malicious intent in the CallManager CDRs.
Configuring MCID
MCID, which is a system feature, comes standard with Cisco CallManager software. MCID does
not require special installation or activation.
To configure MCID, follow these general procedures:
Step 1 Ensure that the CDR flag is set to True.

Step 2 Configure the alarm.
Step 3 Configure a softkey template with the Malicious Call Trace softkey.
Step 4 Assign the MCID softkey template to an IP Phone.
Step 5 Notify users that the MCID feature is available.
Configuring CallManager to Support CDRs

To enable Cisco CallManager to flag a CDR with the MCID indicator, you must enable the CDR
flag. Use the following procedure in Cisco CallManager Administration to enable the CDR flag:
Step 1 From the drop-down list, choose Service > Service Parameters.
Step 2 Choose the Cisco CallManager server name.
Step 3 In the Service field, choose Cisco CallManager. The Service Parameters
Configuration window appears.
Step 4 In the System area, set the CDR Flag Enabled field to True if it is not
already enabled, as shown in Figure 17-18.
Figure 17-18 Configuring CallManager Service Parameters to Support CDRs
Step 5 If you have made a change, click Update.

Malicious Call Identification 401
Configuring MCID Alarms

To provide for the MCID alarm information to appear in the Event Viewer, you need to enable the
alarm event level. Use Cisco CallManager Serviceability and the following procedure to activate
alarms for MCID:
Step 1 Choose Application > Serviceability. The Cisco CallManager

Serviceability application opens.
Step 2 Choose Alarm > Configuration. The Alarm Configuration window is
displayed.
Step 3 From the list, choose the Cisco CallManager server.
Step 4 In the Configured Services drop-down list, choose Cisco CallManager.
The Alarm Configuration window updates with configuration fields.
Step 5 Under Event Viewer, in the Alarm Event Level drop-down list, choose
Informational, as shown in Figure 17-19.
Figure 17-19 Configuring MCID Alarms in Windows 2000 Event Viewer
Step 6 Under Event Viewer, check the Enable Alarm check box.
Step 7 If you want to enable the alarm for all nodes in the cluster, check the Apply
to All Nodes check box.
Step 8 Click Update to turn on the informational alarm.
Adding the MCID Softkey

For users to trigger MCID alerts, you must add the appropriate softkey to the softkey template
configured for their IP Phones. Use this procedure in Cisco CallManager Administration to add
the Malicious Call softkey to a template:
Step 1 Choose Device > Device Settings > Softkey Template. The Find and List
Softkey Templates window appears.
Step 2 Select the Softkey Template assigned to your users. If your users are using
one of the built-in softkey templates, you will need to create a new softkey
template, as described in Chapter 16, “Configuring User Features, Part 1.”
Step 3 In the upper-right corner of the window, click the Configure Softkey
Layout link. The Softkey Layout Configuration window appears.
Step 4 In the Call States area on the left, choose Connected. The list in the
Unselected Softkeys pane changes to display the available softkeys for this
call state.
Step 5 In the Unselected Softkeys pane, choose Toggle Malicious Call Trace, as
Step 6 To move the softkey to the Selected keys pane, click the right arrow.
Step 7 Click Update to ensure that the softkey template is configured.
Multilevel Precedence and Preemption 403
Figure 17-20 Adding the MCID Softkey
Multilevel Precedence and Preemption

Most telephone systems are designed to accommodate busy-hour traffic. However, should an
emergency occur, chances are that everyone would attempt to make a telephone call at the same
time, and this behavior could overwhelm the system. In a national emergency or when network
performance has degraded, an organization might want to give certain individuals calling
precedence over others to implement emergency plans.
The MLPP feature, introduced in Cisco CallManager Release 4.0, allows placement of priority
calls. Precedence is the priority level that is associated with a call. Preemption is the process that
terminates existing calls of lower precedence and extends a call of higher precedence to or through
(in the case of a gateway) the target device.
Cisco CallManager provides indication signals (tones and displays) to MLPP-enabled devices to
ensure that the calling and called parties are aware of an MLPP call. MLPP-indication-enabled
devices can play preemption tones and receive MLPP preemption announcements that the
announcement server (annunciator) generates when there is a high-precedence call. The
precedence ringback and ringer have a different cadence than the regular ringback and ringer.
MLPP indication settings are configured in the device windows in Cisco CallManager
Administration.
The six MLPP precedence levels presented in Table 17-2 are available in the Translation Pattern
Configuration window of Cisco CallManager Administration (Route Plan > Translation
Pattern).
MLPP Precedence Levels
MLPP Precedence Setting Precedence

Highest Executive Override
Starting in Cisco CallManager Release 4.1,

Executive Override is the highest precedence level
and has a precedence setting of 0. Six precedence
levels are available, 0 through 5.
Second highest Flash Override
(In Cisco CallManager Release 4.0, Flash

Override is the highest precedence level and has
a precedence setting of 0. Five precedence levels
are available in Release 4.0, 0 through 4.)
Third highest Flash
Fourth highest Immediate
Fifth highest Priority
Sixth highest Routine
Does not override the incoming precedence Default
level but rather lets it pass unchanged
Calls with a precedence level higher than Routine are considered precedence calls.
As an MLPP example, Phone B (extension 3116) is on a call with no precedence. Phone A

(extension 3101) attempts to place a call to Phone B. Because Phone B is on a call, the call is
forwarded to voice mail. Phone A hangs up and dials a Flash Override code of 555+3116. This
action causes a preemption annunciator message to play on Phone B. Phone B hangs up and hears
the precedence ring. Phone B picks up the call from Phone A, and a precedence call is established.
Summary
This chapter covered the concepts and configuration of Cisco CallManager enhanced end-user
features. A highly desired feature supported in Cisco CallManager is Extension Mobility. This
feature enables users to temporarily access their Cisco IP Phone configuration from other Cisco
IP Phones. This allows you to assign users roaming phone profiles and keep them from being tied
to a physical location.
Client Matter Codes (CMC) and Forced Authorization Codes (FAC) allow you to manage call
access and accounting. CMC codes are generally used to track calls on a per-client or per-
department basis, whereas FAC authorization levels are used to prevent toll fraud from internal
users.
Malicious Call Identification (MCID) allows users to invoke a softkey to initiate a series of events
when they receive a threatening call. This even can range from flagging a CDR record in the
CallManager database to notifying an emergency response group on the PSTN.
Call Display Restrictions give you the ability to control the caller ID information passed between
devices on the IPT network. Finally, Multilevel Precedence and Preemption (MLPP) give you the
ability to allow placement of higher-priority calls over lower-priority calls.
Review Questions
1. When a user logs in to a Cisco IP Phone using the Cisco CallManager Extension Mobility
feature, what is pushed to the IP Phone?
a. the device profile associated with the physical port
b. an automatically generated device setting
c. a temporary default phone button template
d. the device profile associated with the user
2. Which two of the following are requirements to implement the MCID feature in a Cisco
CallManager deployment? (Choose two.)
a. PRI service
b. critical alarm level
c. MCID-T
d. Cisco Emergency Responder Service
e. Defense Switched Network
3. What should you instruct users to do if they receive a fast-busy tone after accidentally entering
an invalid client matter code?
a. wait for the prompt and rekey the correct code
b. call technical support
c. enter the correct code within 3 seconds
d. enter the authorization code first
e. press the Speed Dial button
f. hang up and try the call again
4. Which four configuration items would enable a user (User A) in a hotel guest room to view
the name and number of a caller and prevent the caller from viewing the user A caller ID
information? (Choose four.)
a. Connected Line ID Presentation Restricted
b. Connected Line ID Presentation Allowed
c. Connected Name Presentation Restricted
d. Connected Name Presentation Allowed
e. Calling Line ID Presentation Restricted
f. Calling Line ID Presentation Allowed
g. Calling Name Presentation Restricted
h. Calling Name Presentation Allowed
5. A user logs in to a phone using the CallManager Extension Mobility functionality. They later
move to a different location and attempt to log in to another Cisco IP Phone while remaining
logged in to the original phone. What behavior occurs if the CallManager Extension Mobility
service parameters are left in the default configuration
a. The user will not be able to log in to the second phone.
b. The user will be able to log in to the second phone and remain logged in to the first IP
Phone.
c. The user will be able to log in to the second phone and be automatically logged out of
the first IP Phone.
d. The user will have to wait 8 hours for the first phone to log out before they are able to
automatically log in at the second.
6. You are configuring a CallManager-based IPT network for a customer. They want to require
users to dial traceable PIN numbers assigned to differing authorization levels to reach PSTN
numbers causing toll charges. What Cisco CallManager feature will accomplish this?
a. CMC
b. CAC
c. FAQ
d. FAC
7. How is a user able to trigger MCID functions on any given call?
a. by dialing the MCID sequence configured in CallManager
b. by signaling using a hook-flash sequence on the phone
c. by pressing the MCID softkey
d. any of the above
8. What is the highest MLPP level you can assign in CallManager 4.1?
a. Priority
b. Immediate
c. Flash
d. Executive Override
e. Flash Override
9. You want to allow a Cisco IP Phone to override the call display restrictions configured inside
your agency. What configuration setting will allow this?
a. the Calling Line ID Presentation configuration from the translation pattern configuration
b. the Calling Line ID Presentation configuration from the device configuration
c. the Ignore Presentation Indicators configuration from the translation pattern configura-
tion
d. the Ignore Presentation Indicators configuration from the device configuration
10. You have created a default device profile for a Cisco IP Phone 7960. How do you assign this
default device profile to your devices to enable Extension Mobility support?
a. Do nothing, the default device profile is automatically assigned.
b. You need to select the devices CallManager should apply the device profile to using the
device profile Find/List Phones dialog box.
c. You must select the Use Current Device Settings for the Logged Out profile of the
device.
d. You must create a specific user profile and assign it to the phone as the default.
following topics:
■ Explaining the features and functions of the

Cisco CallManager Attendant Console
■ Defining the key Cisco CallManager

Attendant Console components and
explaining their operation
■ Explaining the basic operation of call routing

and call queuing as it relates to Cisco
CallManager Attendant Console
■ Describing the Cisco CallManager Attendant

Console redundancy process
■ Configuring the server portion of Cisco

■ Installing and configuring the client portion

of Cisco CallManager Attendant Console
■ Using the user and administrative features of

Cisco CallManager Attendant Console
CHAPTER 18
Configuring Cisco Unified
Enterprises today can choose to route inbound telephone calls through numerous methods.
These methods are either completely automated, manually directed, or some hybrid of
automated and manual operation. An automated operation uses an application that can accept
inbound calls, query the caller for destination information, and rapidly dispatch the call without
operator intervention. A manual operation handles each inbound caller through a specially
trained and equipped operator who assesses the purpose of the call and intended destination and
uses tools to dispatch the call.
There are benefits associated with each method. Handling each inbound caller through an
operator can lead to a heightened sense of customer satisfaction and, in many cases, a more
reliably dispatched call. Automation of inbound call dispatch is efficient and affordable.
An operator or receptionist can use the Cisco CallManager Attendant Console to accept inbound
calls, query the caller for destination information, and rapidly dispatch the call without operator
intervention. For businesses that desire operator intervention, Cisco CallManager Attendant
Console is designed to more efficiently automate both the user operations and the administrative
operations of a manual attendant function. Cisco CallManager Attendant Console has an
additional benefit over traditional consoles and line extenders because each user line is
monitored, as opposed to monitoring only a select few in a time-division multiplexing (TDM)–
based system.
This chapter covers the features, operation, installation, and configuration of Cisco
CallManager Attendant Console on both Cisco CallManager and on the Attendant Console
station.
Introduction to Cisco CallManager Attendant Console

Cisco CallManager Attendant Console, a client/server application installed on a PC, allows you
to set up an Attendant Console to use with Cisco IP Phones. Employing a graphical user
interface, the Attendant Console uses speed dial buttons and quick directory access to look up
telephone numbers, monitor line status, and direct calls. A receptionist or administrative
assistant can use the Attendant Console to handle calls for a department or company, or other
employees can use it to manage their own telephone calls.
410 Chapter 18: Configuring Cisco Unified CallManager Attendant Console
The Attendant Console installs on a PC with IP connectivity to the Cisco CallManager system.
The Attendant Console works with a Cisco IP Phone that is registered to a Cisco CallManager
system. Multiple Attendant Consoles can connect to a single Cisco CallManager system. When a
server fails, the Attendant Console automatically fails over to another specified server in the
cluster.
The Cisco CallManager Attendant Console client application is shown in Figure 18-1. The client
is downloadable from the Cisco CallManager plug-in web page. (Choose Applications > Install
Plugins, and click Cisco CallManager Attendant Console.) The client installs on end-user
systems running Microsoft Windows 98, Me, 2000, and XP. The installation program places a
Cisco CallManager Attendant Console icon on the attendant desktop and can also be accessed
using Start > Programs.
Figure 18-1 Cisco CallManager Attendant Console

Terms and Definitions

Table 18-1 defines the terminology used for the Cisco CallManager Attendant Console
application.
Table 18-1 Cisco Attendant Console Terminology
Term Definition
Cisco CallManager Client application; web browser user interface; maximum of 96
Attendant Console client clients per Cisco CallManager cluster
Cisco CallManager Cisco CallManager database entry; represents the Cisco
Attendant Console user CallManager Attendant Console client; one per client
Cisco Telephony Call Server application; distributes calls, monitors line state, and
Dispatcher performs call control; one per Cisco CallManager
Hunt group Ordered list of directory numbers (DNs) to which Cisco TCD
distributes calls; maximum of 32 per Cisco CallManager cluster
Pilot number or pilot point DN that points to a hunt group; one per hunt group
Hunt group member Either a DN (extension) or user-line pair; maximum of 16 per hunt
group
The Telephony Call Dispatcher and Attendant Console Directory

The following sections provide additional detail about the Cisco Telephony Call Dispatcher
(TCD) and the Cisco CallManager Attendant Console Directory.
Cisco Telephony Call Dispatcher

As shown in Figure 18-2, the Attendant Console application registers with and receives call-
dispatching services from Cisco TCD. Cisco TCD, a Cisco CallManager service, provides
communication among Cisco CallManager servers, Attendant Consoles, and the Cisco IP Phones
that are used with the Attendant Consoles.
Cisco TCD handles Attendant Console requests for the following items:
■ Call dispatching from pilot point to the appropriate hunt group destination
■ Line status (unknown, available, on hook, or off hook)
■ User directory information (Cisco TCD stores and periodically updates directory information
for fast lookup by the Attendant Console.)
Cisco TCD monitors the status of internal devices and telephones only. An Attendant Console user
cannot see the line state for a telephone that is connected to a gateway.
Figure 18-2 Telephony Call Dispatcher Service
Attendant Attendant
Console Console
Client 1 Client 2
IP IP
TCD
Cisco
CallManager
Database/ Server
Directory
Cisco CallManager Attendant Console Directory

The CallManager Attendant Console directory is used to allow the Attendant Console application
to maintain an active list of the corporate directory of users. Using this list, a receptionist can
verify the state of any IP phone in the corporation and transfer calls simply by dragging and
dropping the active call to the applicable user in the directory.
The Attendant Console server reads and caches the user directory entries at startup. After an initial
handshake determines whether the directory entries have changed since the previous login, the
Attendant Console downloads the user list. The Attendant Console also downloads the user list
when the interval in the Directory Reload Interval field in the Attendant Settings dialog box
expires or when the user clicks the Reload button in the Directory window.
The Attendant Console searches the following files (in order) for the user list:
■ The user list file that is specified in the Path Name of Local Directory File field in the
Attendant Settings dialog box on the attendant PC.
■ The CorporateDirectory.txt file in the user list directory on the Cisco CallManager Attendant
Console server. You can create the CorporateDirectory.txt file if your user list is located on a
directory server that is separate from the Cisco CallManager server.
■ The AutoGenerated.txt file that is generated by the Cisco TCD service and stored in the user
list directory on the Cisco CallManager Attendant Console server. If the Directory Sync
Period service parameter does not equal zero, Cisco TCD generates the AutoGenerated.txt file
when the Cisco TCD service starts and when the directory synchronization period expires.
To modify the Directory Sync Period service parameter, choose Service > Service Parameters.
Choose the appropriate server from the Server drop-down list and choose the Cisco Telephony
Call Dispatcher Service from the Service drop-down list.
The user list file is in Comma-Separated Values (CSV) format and contains the following
information:
■ Last name
■ First name
■ Telephone number
■ Department
Pilot Points and Hunt Groups

A pilot point, a virtual DN that is never busy, alerts Cisco TCD to receive and direct calls to hunt
group members. A hunt group consists of a list of destinations that determine the call redirection
order.
For Cisco TCD to function properly, make sure that the pilot point number is unique throughout
the system (it cannot be a shared line appearance and will probably be the corporate primary DID
number). When configuring the pilot point, you must choose one of the following routing options:
■ First Available Hunt Group Member—Cisco TCD goes through the members in the hunt
group in order until it finds the first available destination for routing the call. (You can choose
this routing option from the Pilot Point Configuration window in Cisco CallManager
Administration.)
■ Longest Idle Hunt Group Member—This feature arranges the members of a hunt group in
order from longest to shortest idle time. Cisco TCD finds the member with the longest idle
time and, if available, routes the call. If not, Cisco TCD continues to search through the group.
This feature evenly distributes the incoming call load among the members of the hunt group.
(You can choose this routing option from the Pilot Point Configuration window in Cisco
CallManager Administration.)
■ Circular Hunting—Cisco TCD maintains a record of the last hunt group member to receive
a call. When a new call arrives, Cisco TCD routes the call to the next member in the hunt
group. Most people know this as round-robin hunting. (You can choose this option from the
Attendant Console Configuration Tool.)
■ Broadcast Hunting—When a call arrives at the pilot point, Cisco TCD answers the call,
places the call on hold, adds the call to the queue, and displays the call in the Broadcast Calls
pane on ALL Attendant Console applications. While on hold, the caller receives music on
hold (MoH), if it is configured. Any attendant can answer the call from the Broadcast Calls
pane. (You can choose this option from the Attendant Console Configuration Tool.)
Call Routing and Call Queuing

The following sections discuss the basic operation of call routing and call queuing as they relate
to Cisco CallManager Attendant Console.
Call Routing
Figure 18-3 shows the interaction of the Cisco CallManager Attendant Console components in a
basic call-routing example when a call is made to a number that is configured as a pilot point and
associated with a hunt group. In this example, 4000 is a support number associated with a support
hotline.
Figure 18-3 Attendant Console Call Routing Example

2
3
1 Cisco TCD 1024 (DN) Busy
1025 (DN) Not Online
1026 (DN) Available
Cisco 5060 (VM) Not Checked
CallManager
IP
Attendant Console Client
and IP Phone
The process numbered in Figure 18-3 is as follows:
1. Cisco TCD receives a call and directs it to the pilot point, DN 4000.
2. Because 4000 is a pilot point and First Available Hunt Group Member is specified as the call-
routing option, the Cisco TCD that is associated with the pilot point checks the members of
the hunt group in order, beginning with the first member, to determine where to route the call.
DN 1024 is busy, DN 1025 is busy, and DN 1026 is available.
Call Routing and Call Queuing 415
3. Cisco TCD routes the call to the first available DN, which is 1026. Because 1026 is available,
Cisco TCD never checks the 5060 voice-mail pilot point number. If all numbers would have
been busy, the call would forward to the voice-mail extension.
Call Queuing
You can configure a pilot point to support call queuing so that when a call comes to a pilot point
and all hunt groups members are busy, Cisco CallManager Attendant Console sends calls to a
queue. While in the queue, callers hear MoH if you have chosen an audio source from the Network
Hold Audio Source and the User Hold MoH Audio Source drop-down lists in the Device Pool
window. The attendants cannot view the queued calls, with the exception of Broadcast groups
(covered in the “Attendant Console GUI: Broadcast Calls” section later in this chapter). When a
hunt group member becomes available, Cisco TCD redirects the call to that hunt group member.
This sequence of events is reflected in Figure 18-4.
Figure 18-4 Attendant Console Call Queuing Example

1 2 3 4
Incoming All Members Send Call to Route Call to
Call Are Busy Queue Member
4000 1024 Busy 1024 Available

1025 Busy
1026 Busy
You enable queuing for a pilot point using the Cisco CallManager Attendant Console
Configuration Tool. You also use the tool to configure the queue size, which is the number of calls
that are allowed in the queue, and the hold time, which is the maximum time (in seconds) that
Cisco TCD keeps a call in the queue. Configuring the Cisco CallManager Attendant Console
Configuration Tool is covered in the “Using the Attendant Console Configuration Tool” section
later in this chapter.
If the queue is full, Cisco TCD routes calls to the “always-route” hunt group member that is
specified in the Hunt Group Configuration window. If you do not specify an always-route member,
Cisco TCD drops the call when the queue size limit is reached. If the call is in the queue for longer
than the hold time, the call is redirected to the always-route member. If an always-route member
is not configured, no action occurs. Configuring the Always Route option in the Hunt Group
Configuration window is covered in the “Hunt Group Configuration“ section later in the chapter.
Server and Administration Configuration

Follow this general procedure to configure the Cisco CallManager server to support the Cisco
CallManager Attendant Console:
Step 1 Add Cisco CallManager Attendant Console users. These individual

users will use the Cisco CallManager Attendant Console application. They
are not the same as directory users that you configure in the User area of the
Cisco CallManager Administration window.
Step 2 Create the ac user and associate all pilot point devices with the user.
You must configure one user named “ac” and associate the attendant IP
Phones and the pilot points with the user. If you do not configure this user,
the Attendant Console cannot interact with Cisco CTIManager, and the
attendant cannot receive calls.
Step 3 Configure the pilot point for the Cisco CallManager Attendant
Console. The pilot point is a DN that provides access to Cisco CallManager
Attendant Console users indirectly through hunt groups. The pilot point
usually maps to the general company number (switchboard number), but it
can use a DN outside of the Direct Inward Dial (DID) range.
Step 4 Configure hunt groups. Hunt groups consist of Cisco CallManager
Attendant Console users or DNs. When adding a Cisco CallManager
Attendant Console user with multiple line numbers to the hunt group, you
must specify which line to use.
Step 5 Activate Cisco TCD and CTIManager services. Verify that the Cisco
TCD service activates and runs on all servers that are running the Cisco
CallManager service. Verify that the Cisco CTIManager service activates
and runs on at least one server in the cluster.
Step 6 Access the Attendant Console Configuration Tool. At a minimum,
change the default username and password of the ac user and, optionally,
configure queuing and assess other hunting options not available directly
from the Pilot Point Configuration window in Cisco CallManager
Administration.
Adding Attendant Console Users

You must add users through the Cisco CallManager Attendant Console User Configuration
window and assign them a password before they can log in to a Cisco CallManager Attendant
Console client. Complete the following procedure to add a user:
Step 1 Open the Cisco CallManager Attendant Console User Configuration

window and choose Service > Cisco CM Attendant Console > Cisco CM
Attendant Console User.
Server and Administration Configuration 417
Step 2 In the upper-right corner of the window, click the Add a New Attendant
Console User link. The Attendant Console User Configuration window
Figure 18-5 Adding an Attendant Console User
Step 3 Create user accounts for each Cisco CallManager Attendant Console user
by entering the appropriate user ID (username) and password for each
account.
Step 4 Click Insert when you are finished.
Adding the “ac” User

You must create one generic user, the ac user, and associate the attendant Cisco IP Phones and the
pilot points with the ac user. This user is designed to give the Attendant Console and related Cisco
IP Phones CTI permissions with the CallManager cluster. When you configure the ac user, the
Cisco CallManager Attendant Console can then interact with the Cisco CTIManager service on
the Cisco CallManager server. Perform the following procedure to configure the ac user:
Step 1 Choose User > Add a New User in the Cisco CallManager Administration
window.
Step 2 Enter ac in the First Name and Last Name fields.
Step 3 Enter ac in the User ID field.
Step 4 Enter 12345 in the User Password field.

Step 5 Enter 12345 in the Confirm Password field.
Step 6 “ac” and “12345” are mandatory default settings when first configuring the
Attendant Console user.
Step 7 Enter a personal identification number (PIN) and telephone number.
Step 8 Check the Enable CTI Application Use check box. You must check this
box for the Cisco CallManager Attendant Console to interact with Cisco
CTIManager.
Step 9 Check the Call Park Retrieval Allowed check box. The user configuration
should now resemble the settings shown in Figure 18-6.
Figure 18-6 Adding the Attendant Console “ac” User

Step 11 Associate all attendant Cisco IP Phones and pilot points with the ac user.
TIP To enhance the security of your deployment, you should change the username and
password of the “ac” user to something beyond the default. To make this change, run the
acconfig.bat file located in the C:\Program Files\Cisco\CallManagerAttendant\bin directory on
the CallManager server. Specific instructions for accomplishing this are in the “Using the
Attendant Console Configuration Tool“ section later in this chapter.
Pilot Point Configuration

You must configure pilot points and hunt groups through the Cisco CallManager Administration
window before Cisco TCD can route calls. In Cisco CallManager Administration, choose Service
> Cisco CM Attendant Console > Pilot Point and in the upper-right corner, click the Add a New
Pilot Point link to open the Pilot Point Configuration window.
Table 18-2 defines the settings in the Pilot Point Configuration window shown in Figure 18-7.
Figure 18-7 Pilot Point Configuration

Table 18-2 Pilot Point Configuration Fields
Field Description
Pilot Name Enter up to 50 alphanumeric characters, including spaces, to specify a
descriptive name for the pilot point.
Device Pool Assign the pilot point to a device pool. The device pool has the Cisco
CallManager whose Cisco TCD service will service this pilot point.
Partition Choose the partition to which the pilot point belongs. Make sure that the pilot
point that you enter in the Pilot Number field is unique within the partition that
you choose. If you do not want to restrict access to the pilot number, choose
None for the partition.
Calling Search To designate the partitions that the pilot point searches when it attempts to route
Space a call, choose a calling search space from the drop-down list.
Pilot Number Enter a DN into this field to designate a DN for this pilot point.
Verify that this number is unique throughout the system; it cannot be a shared
line appearance.
Route Calls To Choose the mechanism to distribute calls to hunt group members:
• Choose the First Available Hunt Group Member option to route incoming
calls to the first available member of a hunt group. This is the default setting.
• Choose the Longest Idle Hunt Group Member option to order members
based on the length of time that each DN or line has remained idle.
If the voice-mail number is the longest-idle member of the group, Cisco TCD
will route the call to voice mail without first checking the other members of the
group.
If you want to use the Circular Hunting or Broadcast Hunting routing options,
use the Attendant Console Configuration Tool.
Location Designates a Call Admission Control (CAC) location for the pilot point to be
used with bandwidth control.
TIP If the pilot point is not the main or general telephone number, the main number can go to
a translation pattern, allowing CallManager to transform the call to the attendant pilot number.
Hunt Group Configuration

After you configure the pilot point, you must configure the hunt group. A hunt group consists of a
list of destinations (either DNs, or Cisco CallManager Attendant Console user or line numbers)
that determine the call-redirection order.
In Cisco CallManager Administration, choose Service > Cisco CM Attendant Console > Hunt
Group. The Hunt Group Configuration window appears. In the shaded column on the left, click
the pilot point for which you want to add hunt group members. You can then add member directory
numbers by clicking the Add Member button, entering the directory number you want to add, and
selecting Update. You can also add the Attendant Console users to the hunt group. Adding by user
has the added benefit of allowing whatever phone is associated with the Attendant Console user to
be included in the hunt group rather than just the specific extension. Figure 18-8 shows an example
of this configuration.
Figure 18-8 Hunt Group Configuration

Table 18-3 defines the settings in the Hunt Group Configuration window.
Table 18-3 Hunt Group Configuration Fields
Field Description
Partition If a hunt group member is a DN, you can click the drop-down arrow to view the
values for the Partition field and then choose the appropriate option. Enter the
DN in the Directory Number field in the Device Member Information section.
The partition designates the route partition to which the DN belongs.
If the DN for this hunt group member is in a partition, you must choose a
partition.
If the DN is not in a partition, choose None.

Directory Enter the DN of the hunt group member device in this field. When the DN is not
Number in the specified partition, an error message appears.
Always Route Always Route Member is an optional check box that applies to DNs only. If this
Member check box is checked, Cisco TCD always routes the call to this hunt group
member, regardless of whether it is busy. Cisco TCD does not check whether the
line is available before routing the call.
To manage overflow conditions, check this check box for voice-messaging or

Auto Attendant numbers that handle multiple simultaneous calls.
User Name If the hunt group member is a user and line number, fill in only the User Name
and Line Number fields in the User Member Information section.
From the drop-down list, choose Attendant Console users who will serve as hunt
group members.
Only Attendant Console users who are added in the Cisco CallManager
Attendant Console User Configuration window appear in this list.
Line Number From the drop-down list, choose the appropriate line numbers for the hunt group.
You can add the same user to the same line only once within a single hunt group.
For example, you cannot add Mary Brown, Line 1, more than once in the hunt
group.
If you have configured queuing, and the queue is full, Cisco TCD routes calls to the always-route
hunt group member that is specified in the Hunt Group Configuration window. If you do not
specify an always-route member, Cisco TCD drops the call when the queue size limit is reached.
If the call is in the queue for longer than the hold time, the call is redirected to the always-route
member.
Activating Cisco TCD and CTIManager Services

Before the Cisco CallManager is able to support the Attendant Console clients, you must activate
the Cisco Telephony Call Dispatcher and CTIManager services. To activate these services, follow
this procedure:
Step 1 In the Cisco CallManager Administration window, choose Application >

Cisco CallManager Serviceability. The Cisco CallManager
Serviceability window appears.
Step 2 Choose Tools > Service Activation. The Service Activation window
displays the list of servers.
Step 3 From the Servers list, choose the server where Cisco CallManager is
running. The Service Activation window shown in Figure 18-9 appears.
The window displays the service names for the server that you chose and
the activation status of the services. Verify the active status of the Cisco
Telephony Call Dispatcher and CTIManager services, or follow the
remaining steps to activate them.
Figure 18-9 TCD and CTIManager Service Activation

Step 4 Check the check boxes next to the Cisco Telephony Call Dispatcher and
CTIManager services.
Using the Attendant Console Configuration Tool

The Attendant Console Configuration Tool is a simple utility found on the Cisco CallManager
server that you can use to perform these tasks:
■ Set the JTAPI username and password beyond the default values of ac and 12345.
■ Set directory values.
■ Enable call queuing for a pilot point and set the maximum hold time.
■ Configure circular hunt groups and broadcast hunt groups.
The Attendant Console Configuration Tool is somewhat hidden on the Cisco CallManager hard
drive. To access the tool, open the acconfig.bat file, which is located in the C:\Program
Files\Cisco\CallManagerAttendant\bin directory on the Cisco CallManager Attendant Console
server. The Edit Server Properties dialog box shown in Figure 18-10 opens.
Figure 18-10 Attendant Console Configuration Tool - Basic Tab
From the Basic tab, you can change the default ac username and password to something else.
The Advanced tab (shown in Figure 18-11) enables you to access additional hunting options not
available in the CallManager Administration Hunt Group Configuration window: enable queuing,
change the queue size (the default is 32 calls), and change the hold time (the default is 0, which
keeps calls in the queue until an attendant becomes available).
Figure 18-11 Attendant Console Configuration Tool - Advanced Tab
TIP Making changes in the Attendant Console Configuration Tool might require a restart of
the CTIManager and Cisco TCD services for the changes to take effect. On rare occasions, a
complete restart of the CallManager server might be required.
Installing and Configuring the Attendant Console Client

After you activate Cisco TCD and CTIManager services and configure the Cisco CallManager
Attendant Console in Cisco CallManager Administration, you are ready to install and configure
the Cisco CallManager Attendant Console plug-in on each attendant PC.
The following steps provide the PC requirements for the Attendant Console:
Step 1 Verify that the PC is running Microsoft Windows 2000 or Windows XP and
has network connectivity to the Cisco CallManager.
Step 2 Install the Cisco CallManager Attendant Console plug-in on each attendant PC.
• Download the Cisco CallManager Attendant Console plug-in by
choosing Application > Install Plugins in the Cisco CallManager
• Save the CiscoAttendantConsoleClient.exe file to the local machine.
• Launch the CiscoAttendantConsoleClient.exe file.
Step 3 As soon as the Attendant Console application launches, the settings
window appears. Configure the Cisco CallManager Attendant Console
settings on the attendant PC, as shown in Figure 18-12.
• Specify the IP address of the Cisco CallManager TCD server and the DN
of the associated IP phone, and click Save.
• Provide the username and password.
Figure 18-12 Configuring the Attendant Console Client
After you install the Cisco CallManager Attendant Console, the user is ready to start the
application. After opening the Cisco CallManager Attendant Console application, the user will
have to log in and then go online. The user is then ready to answer calls.
To launch the application on the PC where the Attendant Console is installed, choose Start >
Programs > Cisco CallManager > Cisco CallManager Attendant Console or click the Cisco
CallManager Attendant Console icon on the desktop.
TIP Receiving the message “Initialization of Call Control Failed. Retrying…” upon opening
the Attendant Console indicates a problem with the ac user account.
Cisco Attendant Console Features

As shown in Figure 18-13, the features of the Cisco CallManager Attendant Console are menu-
driven. You can use the associated shortcut keys to access all of the menu functions.
Figure 18-13 Attendant Console Client User Interface
The following is a description of the menus:
■ File menu—From the File menu, you can go online or offline, log out, and exit the program.
■ Edit menu—From the Edit menu, you can create your own keyboard shortcuts. You can also
add, modify, and delete speed dial entries or groups and view or revise settings, which is an
optional task.
■ View menu—From the View menu, you can change the size of the text in the windows or the
color on the console.
■ Actions menu—You perform call-control tasks through the Actions menu or the Action Key
shortcuts shown in Figure 18-13. This menu includes many feature options, such as answering
calls, transferring calls, parking calls, and enabling other features on the system.
■ Help menu—Cisco CallManager Attendant Console provides online help and easy access to
the latest Cisco CallManager Attendant Console plug-in for an upgrade.
The Cisco CallManager Attendant Console efficiently automates the user and the administrative
operations of a manual attendant function. It has an intuitive and configurable GUI to handle calls
and monitor line state.
In a system with hundreds or thousands of users, a Cisco CallManager Attendant Console operator
can accept calls and perform a directory lookup by selecting the field title in the Directory section
and typing in the first few characters of the last name, first name, or department of the user. A
directory search returns information that matches the query.
An operator can view the status of a user line (busy, idle, or ringing) and advise the caller of the
line state. The operator can then transfer the call to the user by either initiating a traditional transfer
sequence through the Transfer icon or dragging and dropping the call from the selected loop to the
desired user record.
Attendant Console GUI: Call Control

The Call Control window, shown in Figure 18-14, has the following two components:
■ Call Details pane—This component includes the line status, the DN of the incoming call, the
name of the person (if available), the operator DN, and the elapsed time display.
■ Operator Line buttons—This component includes the line status and the DN of the
attendant Cisco IP Phone, displayed in the upper-right corner of the window.
Figure 18-14 Attendant Console GUI: Call Control Window

The Call Details pane displays the lines on the Cisco IP Phone that the Cisco CallManager
Attendant Console controls. The number of lines configured depends on the type of configuration.
For example, if you have a Cisco IP Phone 7960 with two attachments of the Cisco IP Phone 7914
Expansion Module, and you associate a DN with each line, then a total of 34 lines can be
displayed. Each of the lines will show a line status symbol describing the current state of the line.
The various line statuses are described in Figure 18-15.
Figure 18-15 Attendant Console GUI: Line Status Symbol
Line Status
Line Status Corresponding Icon
A call is ringing on the line.
The line is active.
The line is held.
The line is idle.
The status of the line is unknown.
Attendant Console GUI: Custom Speed Dials

The Speed Dials pane (shown in Figure 18-16) is located in the upper-right corner of the screen
and contains DNs and labels customized by the user of the Attendant Console. By clicking a speed
dial button, you place a call from the currently selected attendant line to the associated DN. Speed
dials can be added or deleted quickly by dragging and dropping users from the corporate directory.
Figure 18-16 Attendant Console GUI: Speed Dials
Attendant Console GUI: Directory Lookup

The Directory pane (shown in Figure 18-17) is located in the lower-right corner of the screen. By
clicking a displayed directory entry, you place a call from the currently selected attendant line to
the associated DN.
Figure 18-17 Attendant Console GUI: Directory Lookup
Attendant Console GUI: Parked Calls

From the Parked Calls pane (shown in Figure 18-18), you can view and pick up all calls that have
been parked by all attendants that are connected to the attendant server. If the call is not answered,
you can revert the parked calls in these ways:
■ Right-click the call that you want to park, and then click Revert Park from the context-
sensitive menu.
■ Click the call that you want to park, and then click the Revert Park button on the Call Control
toolbar.
■ Click the call that you want to park, and then choose Revert Park from the Actions menu.
■ On the PC keyboard, press Ctrl-P.

Figure 18-18 Attendant Console GUI: Parked Calls
Attendant Console GUI: Broadcast Calls

Broadcast hunting enables Cisco CallManager Attendant Console to answer calls and place them
in a queue. The Attendant Console displays the queued calls to all available attendants after
inserting the calls into the Broadcast Calls pane (shown in Figure 18-19).
Figure 18-19 Attendant Console GUI: Broadcast Calls
Any attendant in the hunt group that is online can answer the queued calls. Cisco TCD does not
automatically send the calls to an attendant. When an attendant answers a call, Cisco TCD
removes the call from the Broadcast Calls pane and displays it in the Call Control pane of the
attendant who answered the call.
NOTE For Broadcast Calls to appear in the Attendant Console window, the attendants must
be added to the Broadcast Queue hunt group using the attendant’s username and line number
rather than direct phone extension.
Summary
This chapter covered the concepts and configuration of Cisco CallManager Attendant Console.
The Attendant Console is a client/server application that allows you to manage a high volume of
calls through a GUI installed on a client PC. The key Attendant Console components include the
Cisco TCD and CTIManager, Attendant Console client software, Attendant Console user, pilot
points, and hunt groups.
The Cisco TCD provides call routing for incoming calls to the pilot point. You can configure the
pilot point for standard call routing or for queuing. If configured for standard call routing, the pilot
point receives the incoming call and works through the hunt group ordered list until it finds a free
call attendant. If configured for queuing, the TCD holds incoming calls, allowing the attendants to
answer the calls as they become available.
You can install the Attendant Console client by accessing the Cisco CallManager Administration
plug-in window. Configuring the client requires simply specifying the CallManager server IP
address and the attendant DN. The Attendant Console uses a GUI, which is fairly simple to use
and includes a vast number of features.
Review Questions
1. Match the Cisco CallManager Attendant Console terms with the answer choice that is most
closely associated with the term.
Term Answer
1. Pilot point
2. Hunt group
3. Queue
4. Distribution algorithm
5. Cisco TCD
a. First available hunt group member

b. Dispatches calls
c. List of ordered destinations
d. Virtual number
e. Holding place
2. When you configure Cisco CallManager Attendant Console server in Cisco CallManager
Administration, what should you do to prevent calls from being dropped when the queue is
full?
a. Configure the queue size limit to 1024.
b. Configure an always-route member.
c. Configure a hold time.
d. Configure a pilot point.
3. Which two services must be activated for Cisco CallManager Attendant Console to function
properly?
a. Cisco IP Voice Media Streaming Application
b. Cisco CTL Provider
c. Cisco TCD
d. Cisco MOH Audio Translator
e. Cisco CTIManager
4. Which item can be configured from the Pilot Point Configuration window in Cisco
CallManager Administration?
a. Queue size
b. Hold time
c. Longest-idle distribution algorithm
d. Circular distribution algorithm
5. How many hunt groups can you have in a Cisco CallManager cluster?
a. 10
b. 16
c. 32
d. 48
6. Which of the following hunting types allow the Attendant Console clients to receive a list of
currently queued calls and accept them as they become available?
a. Circular Hunting
b. Broadcast Hunting
c. First Available Hunting
d. Longest Idle Hunting
7. To allow the Attendant Console client to function in the cluster, you must create a user with
the username ________ and the password _________.
a. global, cisco
b. ac, 12345
c. ac, cisco
d. global, 12345
8. How do you access the Attendant Console Configuration Tool?
a. By using the Service > Cisco CM Attendant Console menu in the CallManager
Administration window
b. Through the Cisco CallManager Serviceability utility
c. By accessing the Properties window of the Attendant Console client application
d. By executing the acconfig.bat file on the CallManager server hard drive
9. A corporate operator has parked a call at extension 5529. After a few minutes, the operator
notices that the call has not been answered. How can the operator revert the Call Park process?
a. Right-click on the parked call in the Attendant Console software and choose Revert
Park.
b. Dial the parked number from the handset and answer the call to retrieve it.
c. Call the CallManager administrator to revert a parked call.
d. CallManager 4.x does not offer this capability.
10. You are adding an Attendant Console user to the CallManager configuration. You click User
> Add a New User and fill out the user information. What additional option should you select
for this user?
a. Enable CTI Application Use
b. Enable CTI Super User
c. Call Park Allowed
d. You should not be adding an Attendant Console User this way.
following topics:
■ Describing the features of Cisco IPMA and

its two modes
■ Explaining the function of the components

that make up the Cisco IPMA architecture
■ Configuring Cisco IPMA for shared-line

mode
■ Configuring the manager divert target for call

forwarding
■ Using the Assistant Console interface to

place and receive calls
CHAPTER 19
Configuring Cisco IP
Manager Assistant
Senior managers frequently employ assistants who have as one of their duties the management
of telephone activities for the manager or for a group of managers. The assistant normally sets
up conferences for the manager and also places, answers, and transfers calls. This assistant (and
to some extent, the managers) need a different set of features than other users. In addition to the
ability to handle the call activities of the manager, the assistant needs to be able to monitor that
call activity when the manager is on a call, in a meeting, or does not want to be disturbed; to
communicate with the manager using an intercom; and to change features and settings for the
manager.
Cisco IP Manager Assistant (IPMA) is an application that provides these capabilities to

managers and assistants. This chapter discusses the features and functions of Cisco IPMA, how
to configure the feature, and the end-user applications that constitute the application-manager
configuration and Assistant Console.
Cisco IP Manager Assistant Overview

Cisco IPMA enables managers and their assistants to work together more effectively. When
configuring Cisco IPMA, you can configure Cisco CallManager users as managers or assistants.
An assistant user handles calls on behalf of a manager or group of managers.
Cisco IPMA provides specific features for both managers and assistants. The features consist of
enhancements to IP phone capabilities for the manager and desktop interfaces that are primarily
used by the assistant. These enhancements give the assistant a type of “joint control” over the
manager’s lines in a similar style as shared-line extensions.
Cisco IPMA supports two modes of operation: proxy-line support (Cisco CallManager Release
3.3 or later) and shared-line support (Cisco CallManager Release 4.0 or later). The Cisco IPMA
service supports both proxy-line and shared-line support in a given cluster. The difference
between these operation types is covered in the following sections.
CallManager supports Cisco IPMA on Cisco IP Phone 7970, 7960, and 7940 models.
438 Chapter 19: Configuring Cisco IP Manager Assistant
Cisco IPMA with Proxy-Line Support

Cisco IPMA with proxy-line support intercepts calls that are made to managers and routes them
to the assistant or to preconfigured targets that are based on preconfigured call filters. Because the
Cisco IPMA service intercepts calls that are made to managers who are using proxy-line mode, it
requires configuration of partitions, calling search spaces, route points, and translation patterns.
The Cisco IPMA Configuration Wizard enables you to automatically create the partitions, calling
search spaces, route points, and translation patterns that are required for proxy-line mode. The
wizard also creates Bulk Administration Tool (BAT) templates for the Cisco IPMA manager
telephone, the Cisco IPMA assistant telephone, and all other user telephones. The Cisco IPMA
Configuration Wizard can be run only one time; however, you can make corrections and additions
manually in Cisco CallManager Administration.
Cisco CallManager Release 4.x supports the existing proxy-line configuration in earlier versions
of Cisco CallManager, but Release 4.x features such as Barge, Privacy, Call Join, Direct Transfer,
and multiple calls per line require the shared-line mode.
In the proxy-line mode, a manager can be assisted by only one assistant at a time. The proxy-line
IPMA configuration is now considered a “legacy” IPMA support method.
Cisco IPMA with Shared-Line Support

Cisco IPMA with shared-line support enables the assistant to share the primary line of the manager
(the assistant and the manager have the same DN configured on one line). In this support mode,
there is no call routing; calls for the manager ring on both the manager line and the assistant line.
The Cisco IPMA Configuration Wizard is not used in Cisco IPMA with shared-line mode because
there is no need to configure route points, partitions, calling search spaces, or translation patterns.
The Cisco IPMA in shared-line mode also supports Cisco CallManager features such as multiple
calls per line, Call Join, Direct Transfer, Privacy, and Barge.
The manager telephone uses the softkey template called Standard IPMA Shared Mode Manager.
This template has the following softkeys:
■ DND—Do not disturb; turns the ringer off. The manager telephone and the assistant’s IPMA
console display the DND state.
■ ImmDiv—Immediate Divert; diverts the selected call to a preconfigured target (typically the
assistant).
■ TransferToVM—Transfer to voice mail; redirects the selected call to the voice mail of the
manager.
Cisco IP Manager Assistant Architecture 439
One manager can be configured to have up to ten assistants. The manager IP phone will have speed
dials for all the configured assistants, allowing the manager to reach the assistants quickly.
Optionally, you can configure a dedicated incoming intercom line on the manager telephone to the
assistant(s). Cisco IP Phone Services are not supported in shared-line mode for managers.
Table 19-1 summarizes the key differences between the Cisco IPMA shared-line and proxy-line modes.
Table 19-1 Shared-Line and Proxy-Line Modes Compared
Feature Shared Proxy

Number of Managers per Assistant Up to 33 Up to 33
Number of Assistants per Manager Up to 10 One at a
time
IP Phone Services Support on Manager Phone No Yes
Barge, Privacy, Join, Direct Transfer, and Multiple Calls Per Line Yes No
Configuration Simple Complex
Cisco IP Manager Assistant Architecture

The Cisco IPMA feature architecture includes the Cisco IPMA service, the desktop interfaces, and
the Cisco IP Phone interfaces, as shown in Figure 19-1.
Figure 19-1 Cisco CallManager IPMA Architecture

IPMA Service Browser
HTTP
Tomcat IIS Assistant Console
TCP/IP Applications,
Manager Configuration
Applications
MA Servlet
HTTP Assistant
IP Phone
IP
Cisco Cisco Softkey
CallManager CallManager Display Manager
Database Directory IP Phone
IP
Cisco
Cisco
CTIManager
CTIManager
Cisco Cisco
CallManager CallManager
The following is a brief definition for each of the IPMA services:
■ Cisco IPMA service—The Cisco Tomcat web server loads the Cisco IPMA service, a servlet.
Cisco Tomcat, a Microsoft Windows NT service, is installed as part of the Cisco CallManager
installation. The Cisco IPMA service performs the following tasks:
— Hosts the web pages that the manager uses for configuration
— Communicates to a Cisco CallManager cluster through Cisco CTIManager for
third-party call control; Cisco IPMA requires only one computer telephony
integration (CTI) connection for all users in a cluster
— Accesses data from the database and directory
— Supports the Assistant Console application
■ Desktop interface—Cisco IPMA supports the following desktop interfaces for managers and
assistants (accessed from the PC):
— Assistant Console—Used for call control, login, assistant preferences, monitoring

the call activity of managers, and keyboard shortcuts
— Manager Configuration—Used for configuring the Immediate Divert target and to
configure the Send All Calls target and filters (proxy-line mode only)
■ Cisco IP Phone interface—Assistants and managers use softkeys to access Cisco IPMA
features.
NOTE A Cisco IP Phone 7960 or 7970 that is running Cisco IPMA can be equipped with a
Cisco IP Phone 7914 Expansion Module.
Configuring Cisco IPMA for Shared-Line Support

In Cisco CallManager 4.x versions, proxy-line support is still available; however, it is considered a
legacy configuration method due to its complexity and lack of feature support. Newer IPMA deploy-
ments use the shared-line support configuration. Follow these six steps to configure Cisco IPMA:
Step 1 Activate the Cisco IPMA service.

Step 2 Configure the appropriate service parameters.
Step 3 Restart the Cisco IPMA service from the Tomcat web page.
Step 4 Configure a Cisco IPMA manager and assign an assistant.
Step 5 Access Manager Configuration to set a divert target for calls.
Step 6 Install the IPMA Assistant Console on the IPMA Assistant’s PC.
Configuring Cisco IPMA for Shared-Line Support 441
IPMA Step 1: Activating the Cisco IPMA Service

The CallManager installation process installs the Cisco IPMA service on all Cisco CallManager
systems; however, you will still need to activate the service on the servers where you want to use
IPMA. Cisco supports one active and one backup IPMA server in a cluster.
To activate the Cisco IPMA service, follow these steps:
Step 1 In the Cisco CallManager Administration window, choose Application >

Cisco CallManager Serviceability. The Cisco CallManager
Serviceability window opens.
Step 2 Choose Tools > Service Activation. The Service Activation window
displays the list of servers.
Step 3 From the Servers pane, choose the server on which you want to activate
Cisco IPMA.
Step 4 Check the Cisco IP Manager Assistant check box, shown in Figure 19-2,
and click Update.
Figure 19-2 Activating the Cisco IPMA Service

IPMA Step 2: Configuring Service Parameters

Configuring the Cisco IPMA-related service parameters enables automatic configuration of some
IP phone features for the manager’s and assistant’s telephones. These features are the softkey
template and Auto Answer with speakerphone for the intercom line.
Service parameters for the Cisco IPMA service consist of two categories: general and clusterwide.
Specify clusterwide parameters once for all Cisco IPMA services. Specify general parameters for
each Cisco IPMA service that is installed.
Set the Cisco IPMA service parameters by first using Cisco CallManager Administration to access
them at Service > Service Parameters. Next, choose the server where the Cisco IPMA
application resides, and then choose the Cisco IPMA service.
Cisco IPMA includes the following service parameters that must be configured:
■ The general parameters, shown in Figure 19-3, include the following:
— Cisco CTIManager (Primary) IP Address—No default. Enter the IP address of

the primary Cisco CTIManager that will be used for call control.
— Cisco CTIManager (Backup) IP Address—No default. The administrator must
manually enter this IP address.
— Route Point Device Name for Proxy Mode—No default, necessary only if
configuring proxy-mode sharing. Choose the Cisco IPMA route point device name
(which you configure by using Device > CTI Route Point).
Figure 19-3 IPMA General Service Parameters
■ The clusterwide parameters, shown in Figure 19-4, include the following:
— Cisco IPMA Server (Primary) IP Address—No default. The administrator must

— Cisco IPMA Server (Backup) IP Address—No default. The administrator must
— Cisco IPMA RNA (Ring No Answer) Forwarding Flag—The default specifies

False. If the parameter is set to True, an assistant telephone that is not answered
forwards to another assistant telephone.
— Cisco IPMA RNA Timeout—The default specifies 10 seconds. RNA timeout
determines how long an assistant telephone can go unanswered before the call is
forwarded to another assistant telephone. If Call Forward No Answer (CFNA) and
RNA timeout are both configured, the first timeout to occur takes precedence.
— Desktop Heartbeat Interval—The default specifies 30 seconds. This interval timer
specifies how long it takes for failover to occur on the assistant desktop.
Figure 19-4 IPMA Clusterwide Service Parameters
Cisco IPMA includes the following softkey template parameters that must be configured as
clusterwide parameters if you want to use Cisco IPMA automatic configuration for managers and
assistants:
■ Assistant Softkey Template—The default specifies the Standard IPMA Assistant softkey
template. This parameter specifies the softkey template that is assigned to the assistant device
during Cisco IPMA assistant automatic configuration.
■ Manager Softkey Template for Shared Mode—The default specifies the Standard IPMA
Shared Mode Manager softkey template.
IPMA Step 3: Restarting the Cisco IPMA Service

After you have made changes to the IPMA service parameters, you must restart the Cisco IPMA
service. The Cisco IPMA service runs as an application on Cisco Tomcat. To start or stop the Cisco
IPMA service, log in to the Tomcat Web Application Manager window by using administrator
privileges. The URL of the Tomcat Web Application Manager web page is http://<IPMA server>/
manager/list, where <IPMA server> specifies the IP address of the server that has the IPMA
service running on it. After you access this URL, the web page shown in Figure 19-5 appears. To
restart the service, just click the Stop link followed by the Start link (or just click the Reload link)
for the Cisco IP Manager Assistant service.
Figure 19-5 Restarting the Cisco IPMA Service
The Tomcat Web Application Manager requires Cisco CallManager Release 4.0 or later. It enables
you to start or stop an existing application without having to shut down and restart Tomcat, which
restarts all applications that rely on Tomcat. If you have earlier versions of Cisco CallManager,
you will need to stop and then start Tomcat by choosing Start > Programs > Administrative
Tools > Services > Cisco Tomcat. Right-click on Cisco Tomcat and choose Restart.
IPMA Step 4: Configuring Managers and Assistants

Now that you have configured the IPMA services in Cisco CallManager, you can begin creating
the managers and their assistants. You should first configure Cisco IPMA manager information
before configuring Cisco IPMA information for an assistant.
Perform the following procedure to configure a Cisco IPMA manager and assign an assistant to the
manager. Before performing these steps, the managers and assistants must already exist in the Cisco
CallManager directory, be associated with their respective devices, and have a shared line appearance.
Step 1 Choose User > Global Directory.

Step 2 To find the user who will be the Cisco IPMA manager, click the Search
button or enter the username in the field and click the Search button.
Step 3 To display user information for the chosen manager, click the username.
The User Configuration window opens.
Step 4 To configure Cisco IPMA information for the manager, click Cisco IPMA
from the Application Profiles menu.
If this is the first time that this user is being configured for Cisco IPMA, the
User Configuration window displays a message to continue configuration
for a manager or to cancel if the user being configured is not a manager
(shown in Figure 19-6). Click the Continue button.
Figure 19-6 Configuring IPMA Manager

Step 5 The User Configuration window is displayed again and this time contains
manager configuration information, such as device name and profile, Cisco
IPMA-controlled lines, and intercom line (shown in Figure 19-7).
Figure 19-7 Configuring IPMA Manager
Step 6 Check the Uses Shared Lines check box.

Step 7 To associate a device name or device profile with a manager, choose the
device name or device profile from the Device Name/Profile drop-down
list.
NOTE If the manager telecommutes, check the Mobile Manager check box and optionally
choose Device Profile to support Extension Mobility functions.
Step 8 From the Intercom Line drop-down list, choose the intercom line
appearance for the manager, if applicable.
Step 9 From the Available Lines pane, select a line that you want to be controlled
by Cisco IPMA and click the right arrow. The line appears in the Selected
Lines pane. Configure up to five Cisco IPMA-controlled lines.
NOTE The Cisco IPMA-controlled lines (selected) must always be the shared-line DN.
Step 10 To automatically configure the softkey template and Auto Answer with
Speakerphone for the intercom line of the manager telephone, check the
Automatic Configuration check box. Automatic configuration configures the
softkey template and intercom line on the manager telephone or device profile.
Step 11 In the User Configuration window, click the Add/Delete Assistants link. The
Assign Assistants window opens.
Step 12 To find an assistant, click the Search button or enter the specific name (either
the full name or partial string) or user ID of the assistant in the search field.
A list of available assistants is displayed in the window, as shown in Figure 19-8.
Figure 19-8 Assigning IPMA Assistants
Step 13 Check the check box next to the name of the assistant(s) that you want to
assign to the manager. A manager can have a maximum of ten assigned
assistants.
Step 14 To save and continue, click the Insert button; otherwise, to return to the
Cisco IPMA Manager Configuration window, click the Insert and Close
button.
The User Configuration window displays the manager configuration, and
the assistant that you configured is displayed in the Assigned Assistants list,
Figure 19-9 Completed IPMA Manager Window

IPMA Step 5: Configuring Call Divert Target

Managers can modify manager preferences from the Manager Configuration window at
https://<ipma-server-address>/ma/desktop/maLogin.jsp (shown in Figure 19-10). Managers
can access this window from a website; assistants can access it from the Assistant Console
(Manager > Configuration).
Figure 19-10 IPMA Manager Configuration Features
Managers using Cisco IPMA in shared-line mode can set up a divert target and forward calls as
they come in by using the ImmDiv softkey. The divert screen is automatically displayed when you
access the Manager Configuration URL. By default, the divert target is the active assistant for the
manager. Managers and assistants can change this target by entering a valid telephone number in
the Directory Number field. Enter the number exactly as you would dial it from the manager’s
office telephone, as shown in Figure 19-11.
Figure 19-11 IPMA Manager Call Divert Configuration
IPMA Step 6: Installing the IPMA Assistant Console

The IPMA Assistant Console uses a very similar look and feel to the Attendant Console
application discussed in Chapter 18, "Configuring Cisco Unified CallManager Attendant
Console." However, these applications are not the same and must be installed from different
locations. To install the IPMA Assistant Console, access the following URL: http://<IPMA
server>/ma/Install/IPMAConsoleInstall.jsp, where <IPMA Server> is the CallManager server
running the IPMA service. An automated Java installation script runs and installs the IPMA
Assistant Console. By default, this script places shortcuts to the application in the Windows Start
menu and on the desktop.
When running the Assistant Console, the assistant logs in using standard user credentials, and a
window similar to that shown in Figure 19-12 opens. From here, the assistant can view and control
all manager lines for which you have given the assistant authority.
Figure 19-12 IPMA Assistant Console
The Assistant Console application includes the following features:
■ Menu bar—The menu bar is located along the top of the Assistant Console. You can use the
menu bar as follows:
— File—Go online or offline, log in or log out, and exit the console.
— Edit—Create and edit speed dials, personalize keyboard shortcuts, change the
Immediate Divert target, set preferences, and access administrator settings.
— View—Specify text size and color schemes and refresh the default layout.
— Call—Dial, answer, hang up, place on hold, transfer, divert, or add conference
participants to a call.
— Manager—Place an intercom call to a manager, access the Manager Configuration
window, and enable or disable features for a manager.
— Help—Access online help.
■ Call control buttons—Call control buttons are the row of icons that are located along the top
or side of the console. Roll your mouse over a call control button to see a description of its
function. You can use the call control buttons to perform numerous tasks, such as hold,
resume, transfer, join a call, hold a conference, Immediate Divert, and so on.
■ My Calls panel—The Assistant Console displays calls for the assistant and managers in the
My Calls panel. Each telephone line is displayed beneath one of the following headings:
— My Lines—Displays any currently active call that the assistant placed or received
using the assistant’s own telephone line
— Manager Lines—Displays active calls that the assistant is handling or can handle
on behalf of the manager
— Intercom—Displays the status of intercom lines, if applicable
■ My Managers panel—Assistants can use the My Managers panel in the Assistant Console
to monitor call activity and feature status for each manager. Assistants can also enable and
disable manager features from this panel.
■ Speed Dials panel—The speed dial feature allows assistants to set up a personal phone book
on the Assistant Console and to place calls and perform other call-handling tasks using speed
dial numbers.
■ Directory panel—Use the directory to search for a coworker, and then use the search results
to place and handle calls.
■ Status bar—The status bar is located along the bottom of the Assistant Console screen and
displays the following system information:
— Connected or Not Connected—Indicates the status of the assistant’s connection to

the Cisco IPMA server
— Online or Offline—Indicates the availability of the assistant to managers
— Call Control Up or Call Control Down—Indicates the availability of call-
handling features
— Filtering Down—Indicates the availability of call-filtering features (proxy mode)
Summary
This chapter covered the concepts and configuration of Cisco CallManager IP Manager Assistant
(IPMA) features. The IPMA supports proxy-line (legacy) and shared-line configurations. Because
of the complexity of configuring proxy-line environments, CallManager 4.x administrators
typically use shared-line configurations. Configuring the Cisco IPMA consists of configuring
service parameters, configuring a Cisco IPMA manager, and assigning the assistant. After you
have configured the base settings, you can then allow managers to access the IPMA Manager
Configuration window to choose the assistant who should receive their diverted calls. The
attendants can also install the IPMA Assistant Console on their PC to manage multiple manager
extensions using an Assistant Console–style graphic interface.
Review Questions
1. Managers can do which of the following when they access Cisco IPMA Manager
Configuration in shared-line mode?
a. Configure speed dials.
b. Change the divert target.
c. Assign an assistant.
d. Subscribe to services.
2. When you configure Cisco IPMA in shared-line mode, the Cisco IPMA-controlled line must
always be which line?
a. Intercom
b. Assistant’s primary
c. Shared DN
d. Proxy
e. Speed dial
3. Which statement is most closely associated with Cisco IPMA shared-line mode?
a. Supports newer features such as Call Join, Privacy, and Direct Transfer
b. Uses call filters and partitions to route calls to the assistant
c. Uses the Standard IPMA Manager softkey template
d. Uses Cisco Tomcat to load the Cisco IPMA service
4. Which of the following Cisco IP Phones can you use with a Cisco IPMA Configuration?
(Choose two.)
a. Cisco 7912
b. Cisco 7970
c. Cisco 7940
d. Cisco 7900
5. What is the maximum number of assistants per manager if you are using shared-line IPMA
mode?
a. 1
b. 10
c. 16
d. 32
e. 48
6. The IPMA Service is run as a servlet in what parent web server software?
a. Microsoft IIS
b. Apache Tomcat
c. Netscape Webservices
d. IBM Websphere
7. After modifying the IPMA service parameters, what must you do to ensure IPMA operates
properly?
a. Add managers and assistants through the IPMA wizard.
b. Create partitions.
c. Restart the MA Servlet through the Tomcat web server.
d. Restart the IPMA service through the Serviceability control center.
8. When configuring IPMA, which of the following must you do first?
a. Add IPMA assistants.
b. Add IPMA managers.
c. Install the IPMA Assistant Console.
d. Access the IPMA Manager Configuration web pages.
9. How is the IPMA Assistant Console installed?

a. The Assistant Console is downloaded from the CallManager Plug-Ins window and cus-
tomized for the assistant.
b. It can be downloaded from the Cisco website using a valid CCO account.
c. It is installed automatically when the assistant logs in to their IP phone.
d. It is installed from a custom IPMA page managed by the Tomcat web server.
10. How does the IPMA service communicate to all servers in a CallManager cluster?
a. Using SQL replication
b. Through the CTIManager
c. By using Skinny Messages
d. Through the TSP interface
Part V: IPT Security
Chapter 20 Securing the Windows Operating System
Chapter 21 Securing Cisco Unified CallManager Administration
Chapter 22 Preventing Toll Fraud
Chapter 23 Hardening the IP Phone
Chapter 24 Understanding Cryptographic Fundamentals
Chapter 25 Understanding the Public Key Infrastructure
Chapter 26 Understanding Cisco IP Telephony Authentication and

Encryption Fundamentals
Chapter 27 Configuring Cisco IP Telephony Authentication and Encryption

following topics:
■ Identifying security threats to the Windows

operating system
■ Explaining the Cisco security and hot fix

policy for keeping the operating system up to
date
■ Explaining how the optional operating

system security script and manual security
settings provide more stringent security
controls to the hardened Windows operating
system
■ Identifying the antivirus protection software

that is approved for use on a Cisco
CallManager server
■ Explaining the features and functions of

Cisco Security Agent as they relate to
securing Cisco CallManager
■ Defining and administering a password

policy for the administrator password
■ Explaining how to protect Windows against

the most common exploits
■ Explaining common security practices and

settings that are not recommended on Cisco
CallManager
CHAPTER 20
Securing the Windows
Operating System
The telephony system is a business-critical, system, 24 hours a day, seven days a week, to all
companies. Such critical environments must be as secure as possible to avoid breakdowns
related to failures in the system or attacks involving the network. The Microsoft Windows 2000
Server Operating System is the base of Cisco CallManager 3.X and 4.X versions. This chapter
discusses options for hardening and securing the Cisco CallManager operating system and gives
students an overview of possible vulnerabilities and of practices for tightening security in the
Cisco IP Telephony Operating System.
NOTE Cisco Unified CallManager 5.X versions use Red Hat Linux as a base operating
system. This operating system is inaccessible as the 5.X versions of CallManager run as an
appliance.
Threats Targeting the Operating System

When you are securing an operating system, several threats should be considered. Bugs in the
operating system, as well as in the services and applications that come with the operating
system, can pose severe security threats. Because applications are installed on top of the
operating system, even well-written and secure applications can be affected by vulnerabilities
in the underlying operating system. Built-in networking services and applications are especially
sensitive because they are exposed to remote attacks. That vulnerability also applies to the IP
stack in the Windows operating system. The IP stack has a strategic importance and,
unfortunately, also a long tradition of more and less severe security issues. These issues result
not only from the particular implementation of the IP protocol, but also from the lack of security
mechanisms in the protocol itself. As shown in Figure 20-1, password and account policies as
well as insecure Windows configuration settings pose security concerns in the foundation Cisco
CallManager operating system.
460 Chapter 20: Securing the Windows Operating System
Figure 20-1 Threats Targeting the Windows Operating System
Bugs and Exploits

in the Operating System
Password
Common
and
Windows
Account
Exploits
Issues
Virus, Trojan Horse, and

DoS Attacks
Microsoft Windows, as the most popular operating system, is well-known to the public. As a
result, many known issues are related to its password policies as well as vulnerabilities in the
operating system default settings. An attacker might try to log in to the operating system using the
Administrator account with commonly used passwords. In Microsoft networking, File and Print
Sharing services can be used (and might have been turned on by default in some versions of
Windows) to allow access to file shares without any security checking.
Another threat to the system is malicious code execution by viruses, worms, or Trojan horses.
Protection against these threats consists of blocking the threats from the system and detecting and
eliminating attacks that were not blocked.
Finally and extremely important is the fact that server-operating systems are vulnerable to denial
of service (DoS) attacks. If the server operating system cannot resist DoS attacks, an attacker can
tear down the whole IP telephony infrastructure with a single, focused attack against Cisco
CallManager nodes. Besides other methods (separating the server network from other parts of the
network and establishing access control), the server itself should be hardened to resist at least
simple and common DoS attacks.
Lowering the Threats in Windows Operating System

You can divide the possible countermeasures against attacks to the operating system itself into
measures that eliminate vulnerabilities to certain threats and methods to protect the system against
attacks exploiting the remaining vulnerabilities.
The following are practices to reduce possible vulnerabilities:
■ Harden the Windows operating system with Cisco operating system upgrades.
■ Deploy the Cisco security and hot fix policy.

Security and Hot Fix Policy 461
■ Implement a secure Windows password policy.
■ Protect against common exploits involving Windows.
■ Protect against attacks from the network by using the following:
— Antivirus software
— Cisco Security Agent
To protect against bugs and exploits involving Microsoft Windows, Cisco provides an already
hardened version of the Windows operating system called Cisco IP Telephony Operating System.
You must keep the Windows 2000 Server up to date to secure the operating system against new
security holes. For that reason, Cisco provides operating system upgrades and hot fixes. Cisco
CallManager and other Cisco IP telephony applications require these upgrades to function
properly.
Cisco uses the Cisco IP Telephony Operating System in several Cisco IP Telephony Application
Server components, such as Cisco CallManager, Cisco Emergency Responder (ER), Cisco IP
Contact Center (IPCC), and Cisco Interactive Voice Response (IVR). Cisco builds the IP
Telephony Operating System upgrades on top of each other and they are incrementally more
secure. The upgrades provide changes to, for example, the IP stack, file system, Registry, access
control lists (ACLs), and dynamic link library (DLL) engines.
NOTE Before you run an operating system upgrade provided by Cisco, read the release notes
for that upgrade carefully. The operating system upgrade might not apply to your installation and
could harm the running applications. Before upgrading, verify that you are using the proper
operating system upgrade for your Cisco CallManager version. It is also a good practice to
consider making a backup before upgrading the Cisco IP Telephony Operating System.
Cisco IP Telephony Operating System upgrades can be downloaded from Cisco.com at http://
www.cisco.com/cgi-bin/tablebuild.pl/cmva-3des (requires CCO account).
Security and Hot Fix Policy

Cisco closely monitors security bulletins from Microsoft and evaluates them based on the impact
to Cisco CallManager and other IP telephony applications.
When Microsoft posts a security patch, Cisco determines whether the patch affects applications
and operating system components in Cisco CallManager and applications that share the same
operating system installation process. Cisco then tests the relevant patches to verify correct
operation with Cisco applications. This is a list of applications and operating system components
that might be affected by a patch:
■ Microsoft Windows 2000 Server (including any Windows component or subcomponent

installed by Cisco)
■ Microsoft Internet Information Server (IIS)
■ Microsoft Internet Explorer
■ Microsoft Structured Query Language (SQL) Server 2000
CAUTION The operating system upgrades provided by Cisco are not the same as upgrades
provided by Microsoft. The operating system upgrades and patches provided by Cisco are
tailored for IP telephony applications. If a Microsoft service pack (SP) or hot fix is installed for
the Cisco IP Telephony Operating System, the applications running on the Cisco IP Telephony
Operating System might be adversely affected.
The security patch and hot fix policy for Cisco CallManager specifies that any applicable patch
deemed Severity 1 or Critical must be tested and posted to Cisco.com within 24 hours as a hot fix.
All other applicable patches are consolidated and posted once a month as incremental service
releases. Notification tools (e-mail service) for providing automatic notification of new fixes,
operating system updates, and patches for Cisco CallManager and associated products are also
available:
■ Cisco CallManager Notification Tool—This e-mail service provides automatic notification

of new fixes, operating system updates, and service releases that are available for Cisco
CallManager and related products, including Cisco CallManager Attendant Console, Cisco IP
Manager Assistant (IPMA), and Bulk Administration Tool (BAT). To subscribe, go to http://
www.cisco.com/cgi-bin/Software/Newsbuilder/Builder/VOICE.cgi and follow the
instructions on the web page.
■ Cisco Product Security Incident Response Team (PSIRT) Advisory Notification Tool—
This e-mail service provides automatic notification of all Cisco security advisories released
by Cisco PSIRT. Advisories that describe security issues that directly impact Cisco products
provide a set of actions required to repair these products. To subscribe, go to http://
www.cisco.com/en/US/products/products_security_vulnerability_policy.html and follow the
instructions on the web page.
NOTE The Cisco IP Telephony Operating System configuration and patch process does not
currently allow an automated patch-management process.
Operating System Hardening

One of the most important steps in securing an IP telephony server is installing all the applicable
product or component security updates and then making sure that they are kept up to date. Cisco
provides three types of updates for the Cisco IP Telephony Operating System:
■ Operating system upgrade—Comprehensive upgrade to all components of the operating

system that is released two to three times a year, including Microsoft Windows 2000 SPs,
Internet Explorer SPs, BIOS, firmware, drivers, Microsoft hot fixes, security configuration
changes, third-party software that is installed in the base operating system, and configuration
changes to match the currently shipping operating system version.
■ Operating system service releases—Primarily a comprehensive roll-up of security hot fixes

that are released the third Tuesday of each month when needed to deliver new security hot
fixes. The operating system service release occasionally contains nonsecurity hot fixes or
configuration changes needed to resolve a defect in the operating system.
■ Critical hot fixes—When Microsoft releases a hot fix that is critical for Cisco IP telephony
products, Cisco tests the hot fix and posts it to Cisco.com within one business day of the
release by Microsoft.
An IP telephony server must be used for IP telephony purposes only. The server should not be used
as a common file server that stores user data or has user applications, such as Microsoft Office
products, installed on it. File-share access has to be limited to the absolute minimum needed (for
instance, to access log files and generate reports). Strict file access control has to be deployed, and
auditing of network file access should be enabled. This practice also eliminates the need to add
user accounts to the server; only administrator and auditor accounts should exist. If network file
access is not needed at all, it should be disabled to enhance the security of the server.
The more services that are running on a server, the more likely it is that vulnerabilities can be
exploited by an attacker. To minimize this risk, only the services that are needed should be
activated. Many services that are not needed have already been disabled in the Cisco hardened
version of the Microsoft Windows 2000 Server operating system.
To make Cisco CallManager even more secure, Cisco provides additional security scripts and
information on how to protect the Cisco IP Telephony Operating System against common threats.
Do not install any other application on the servers unless it is approved software, such as Cisco
Security Agent or antivirus products. A hardened IP telephony server has to be stripped down to
run only the services and applications that are needed for its operation.
IP Telephony Operating System Security Scripts

Cisco IP Telephony Operating System 2.6 and later includes scripts and information guides on
additional security settings. These security guides and scripts provide additional security settings
beyond those that are installed by default in the Cisco IP Telephony Operating System. The
settings in the optional security script have not been included by default and are not intended for
all customers to use. When planning to use the optional security script, consider these points:
■ The optional security script settings are supported only for Windows 2000 servers that are
running Cisco CallManager Release 3.3(2) and later.
■ The optional security script settings might have an adverse impact on some of the other Cisco
IP telephony applications that use this operating system, on the interaction between Cisco
CallManager servers and some other Cisco IP telephony applications, and on some supported
third-party software.
■ Because these settings are not installed by default, they receive only a limited amount of
testing.
■ Only experienced Windows administrators should apply the optional security script or manual
settings.
CAUTION Applying the optional security script can destroy collocated applications, such as
Cisco IPCC Express, Cisco IP IVR, and Cisco IP Queue Manager (IP QM)!
As shown in Figure 20-2, the optional security script and some additional information are
available in the C:\utils\SecurityTemplates folder.
The script file is a batch job that can be started by clicking the CCM-OS-OptionalSecurity.cmd
file. Before doing so, read the CCM-OS-OptionalSecurity-Readme.htm file to identify possible
issues with applications running on the operating system:
■ The CCM-OS-OptionalSecurity-Readme.htm file contains information on what the

CCM-OS-OptionalSecurity script is changing in the operating system and provides additional
security settings that can be configured manually.
■ Some of the optional security settings cause upgrades to fail. Therefore, if the Cisco IP
Telephony Operating System optional security settings have been installed on the server and
you want to upgrade the server, read the “Before CallManager Upgrade” guide. It includes a
checklist of all settings to which you must revert for the upgrade to work.
Figure 20-2 CallManager Security Script Templates
In addition, Cisco has provided an additional security script in the C:\utils directory, shown in
Figure 20-3. This script is called the IP security filter and will block the fixed Windows 2000 and
SQL ports. Read the IPSec-W2KSQL-Readme.htm file for instructions on how to use the IP
security filter.
Figure 20-3 CallManager SQL Security Script Template
The security script provides the following benefits:
■ Blocking the fixed Windows 2000 and SQL ports adds an extra layer of protection from
viruses, worms, and hackers. A provided script eases the creation of the IP security filter. You
must customize the script with the IP addresses that the organization wants to allow through
the filter. For example, the Cisco CallManager uses SQL ports allowing every IP address to
connect to these port numbers. The IPSec-W2KSQL script would allow SQL connections
only from the IP addresses defined in the script. These consist mostly of the other Cisco
CallManager servers in the cluster and applications that need direct access to the database, for
example, to the call detail record (CDR) tables.
■ Using this IP security filter increases the management overhead of the servers. If the IP
infrastructure changes or additional servers are added to the Cisco IP telephony solution, the
permit lists on all the servers will need to be updated.
NOTE The name of the IPSec-W2kSQL file has nothing to do with the IPsec virtual private
network (VPN) umbrella standard.
Antivirus Protection 467
File-Sharing Considerations
To secure the access to file shares, limit access to the absolute minimum number of users who need
it, as shown in Figure 20-4. Check the share permissions on all shared folders, for example, the
CDR and TFTPPath folders. Avoid assigning share permissions for the Everyone group. You can
find detailed information on how to secure file shares in the CCM-OS-OptionalSecurity-Readme
guide, located in the folder C:\Utils\SecurityTemplates.
Figure 20-4 Managing CallManager Share Permissions
Antivirus Protection
In addition to hardening the Cisco IP Telephony Operating System by using built-in tools, you
should protect the system by using antivirus software. This practice will defend against attacks by
viruses, Trojan horses, and worms. Make sure that the software itself and the virus definition files
are kept up to date so that the newest viruses can be detected. Disable heuristic scanning in any
antivirus software because it can block Cisco CallManager web pages from operating. Cisco
currently supports these antivirus products on the Cisco IP Telephony Operating System:
■ McAfee VirusScan Enterprise 4.5, 7.0, and 7.1
■ Symantec AntiVirus Corporate Edition 7.61, 8.0, and 8.1

■ Trend Micro ServerProtect v5
The list of the latest supported antivirus software is available at http://www.cisco.com/en/US/

products/sw/voicesw/ps556/prod_bulletin0900aecd800f8572.html.
NOTE Heuristic scanning refers to the ability of the software to flag suspicious files or
attachments because they resemble known viruses. Heuristic scanning uses behavior-based
rules to identify and block new viruses without requiring you to first download a patch.
Cisco Security Agent

In addition to antivirus protection, you must protect the operating system against other threats
from the network, such as DoS attacks. For these issues, Cisco provides Cisco Security Agent
software to install on every Cisco CallManager system. The Cisco Security Agent implements the
host-based intrusion prevention system (HIPS), which provides an additional layer of protection
against known and unknown attacks. At the same time, CSA provides security services not offered
by the host operating system. Examples of these services are personal firewall protection, software
keylogger detection, and abnormal application behavior protection.
Cisco Security Agent is designed to protect the endpoint from network-borne attacks, and it
enforces its protection rules on several levels. One of them is the protection of the underlying
operating system from potentially hostile applications. Cisco Security Agent provides three basic
areas of operating system protection:
■ Protection of operating system integrity—Cisco Security Agent policy rules always

prohibit access to sensitive system files and Registry settings. For example, no application can
change files in the Windows system folder.
■ Prevention or restriction of application misbehavior resulting from injection of hostile

code or other network attacks—Several policy rules are dynamic and allow or disallow
local resource access based on the behavior of the application and, hence, its potential
“hostility.” For example, if a client application accesses the network, it is automatically
considered less trusted and its access to local resources is further restricted.
■ Endpoint, or “personal” firewalls—Cisco Security Agent can allow or deny network access
to any local application, and, hence, minimize access to and from the system, enforcing the
least-privilege rule.
Cisco Security Agent (CSA) operates independently of native operating system functions,
providing an independent layer of protection that prevents attacks even when the native operating
system access control methods are breached. You should never deploy the CSA in place of strong
host security, but as an additional protective layer to provide protection methods not available in
the host operating system.
The rationale behind the behavioral approach is that although the number of methods and exploits
to attack a system is extremely large, the number of possible consequences of these attacks is
relatively small. For example, a web server can be persuaded by the attacker to execute a local file
or an executable attachment in an e-mail attempting to access the Windows Registry. CSA can
recognize application behavior leading to or following an attack and prevent the malicious actions.
This ability is also why CSA does not require constant updates; its policies need to be updated
only if a completely different class of attacks is created, which is relatively rare.
CSA for IP telephony servers is available in two versions:
■ Headless agent—This version comes with a set of rules for a specific server platform, such
as Cisco CallManager; no further configuration is necessary.
■ Managed agent—This version has to be configured with rules for the appropriate IP
telephony servers. Predefined rules can be downloaded from Cisco.com.
CAUTION Do not use the headless agent when running Cisco CallManager with collocated
applications, such as Cisco IPCC Express, Cisco IP IVR, or Cisco IP QM, because the fixed
policy of the headless agent will not support these applications (and as a consequence they will
not work properly).
Cisco Security Headless Agent

The free headless agent has a fixed security policy and no centralized reporting capabilities. For
each type of IP telephony server, a different (predefined) agent kit is available for the headless
agent. The headless agent is configured with appropriate policies and exceptions for a typical
supported configuration of that server. The headless agent should be used in environments where
centralized reporting is not required or practical and the IP telephony servers are aligned with
Cisco specifications for installed software and system and application configuration and where
they feature no add-ons that might conflict with the security rules of the headless agent.
NOTE The headless agent is also commonly referred to as the standalone CSA agent on the
Cisco website.
Cisco Security Managed Agent

The managed version of CSA uses CiscoWorks VPN/Security Management Solution (VMS) and
CSA Management Center (MC) for centralized policy distribution and allows event correlation
and reporting. As with the headless agent, which comes in different configurations for different
types of IP telephony servers, CSA MC also allows the administrator to load predefined,
application-specific policies for each IP telephony server type.
NOTE Cisco offers a free, predefined policy for the CSA Managed Agent that deploys the
same CallManager security standards as the standalone CSA.
The managed agent should be used in environments where centralized reporting is required, where
servers do not use a typical configuration (for example, with nondefault TCP or UDP ports) or
have special application requirements (for example, custom systems management software), or
where the default policies need to be augmented with site-specific protection requirements.
Deployment of the managed agent also allows the use of CSA Profiler, an expert add-on tool that
can, to a large extent, automate generation of custom application policies. This add-on would
allow an expert CSA administrator to further enhance the built-in policies and confine every IP
telephony application to a sandbox, similar to the functions that the built-in Restrictive MS IIS
Module and Restrictive MS SQL Server Module provide for those two applications.
The CSA Profiler must be purchased separately, but it does not require any other software to be
installed on the profiled servers.
CSA Supported Applications

CSA is available for Cisco CallManager Release 3.2(3), 3.3, and later. To use CSA for another
Cisco IP telephony application, check the CSA administration manual to determine whether Cisco
supports CSA for that particular application.
This is a list of software add-ons that are supported with CSA on the same server:
■ BMC PATROL
■ Concord eHealth Monitor
■ Diskeeper Server Standard Edition 8.0.478.0
■ HP OpenView Operations Agent 7.1
■ HP OpenView Performance Manager 3.3
■ Integrated Research PROGNOSIS
■ McAfee VirusScan 7.0
■ Micromuse Netcool
■ NAI ePolicy Agent
■ NetIQ Vivinet Manager
■ RealVNC VNC
■ Symantec AntiVirus Corporate Edition 8.0
■ Trend Micro AntiVirus
■ Windows Terminal Services
NOTE The Cisco Security Agent headless agent and the Cisco Security Agent policies for the
Cisco Security Agent MC are both available at http://www.cisco.com/cgi-bin/tablebuild.pl/
cmva-3des (Cisco CCO account required).
CSA Protection
The CSA default operating system protection rules for IP telephony servers provide basic
operating system hardening and integrity protection and contain rule exceptions for supported
add-on applications. In regard to local resource access control, these policies can be summarized
as follows:
■ Allow specific actions required by basic operating system processes
■ Protect the integrity of the system binaries and other sensitive files from local applications
■ Protect the integrity of the system Registry from local applications
■ Allow all other actions (including network access and access to local files as dictated by the
native security of the local host, for example, file ACLs in Windows)
In addition to these basic rules, many other rule modules constitute the total CSA protection policy
of a system.
CSA Guidelines
At the minimum, for each server, deploy the headless CSA, as shown in Figure 20-5. The built-in
operating system protection policies are sound and generally do not require tuning for enhanced
protection, except where dictated by the site policy.
Figure 20-5 Headless (or Standalone) CSA Interface
So-called “false positives,” events that are erroneously classified as attacks, are very likely when
using unsupported server add-ons, such as system management and unsupported antivirus
software. To eliminate this erroneous behavior, deploy the managed agent and add the requested
permissions for these applications so that CSA will not consider them to be malicious.
CSA also provides personal firewall functions by restricting network connections to the server.
The headless agent has a fixed policy that allows all inbound connections to the server, and this
cannot be changed. If you want to use CSA to control network connectivity to the server, you have
to use the managed agent. Alternatively, you could use native Windows IP security filtering or rely
solely on packet filtering by network devices, such as routers or firewalls.
CSA by default allows the agent service to be stopped by the local administrator (using the net
stop csagent command). When using the managed version of CSA, you can apply an agent policy
that blocks the local administrator from stopping the agent.
TIP The CSA should be installed on the Cisco CallManager server after you have applied the
security template. Otherwise, the CSA will think many security template modifications are
attacks on the CallManager server.
Administrator Password Policy

One of the easiest and most frequently used attacks against Microsoft operating systems is to try
to log in to the Administrator account, using various well-known passwords. To block that security
Administrator Password Policy 473
hole, consider using strong password policies, renaming the Administrator account, and other
mechanisms to protect the Administrator account.
The Windows operating system gives administrators the ability to assign restrictions to password
and account policies, as shown in Figure 20-6.
Figure 20-6 Implementing Password Restrictions in Windows 2000
A general rule is not to create any user accounts on an IP telephony server. Only administrators
and operators should have access to the server. Make sure that these accounts have complex
passwords. If a password is too simple, not kept secret, or not changed for a long period, it can be
discovered and misused by unauthorized people. The account policy settings should not be
modified, because setting the lockout policies can adversely affect the system during the next
upgrade (requiring a new installation from scratch). Consider these issues:
■ Setting the account policy is more important for servers with user accounts because otherwise
the administrator has no control over the frequency of password changes by the users.
■ The Minimum Password Length parameter determines how short passwords can be. If it is set
to zero, blank passwords are allowed. It is recommended that you set this value to at least eight
characters.
■ The Passwords Must Meet Complexity Requirements parameter determines whether

password complexity is enforced. If this setting is enabled, passwords must meet these
requirements:
— The password is at least six characters long.

— The password contains characters from at least three of these categories:
— English uppercase characters (A through Z)
— English lowercase characters (a through z)
— Base-10 digits (0 through 9)
— Nonalphanumeric characters (For example: $, !, %, #, &)
— The password does not contain three or more characters from the username.
To configure the password policy for an account, complete these steps:
Step 1 From the Cisco CallManager server, click Start.

Step 2 Choose Settings.
Step 3 From the Settings menu, choose Control Panel.
Step 4 When the Control Panel window opens, click Administrative Tools.
Step 5 In the Administrative Tools window, click Local Security Policy.
Step 6 When the Local Security Settings window opens, click Account Policy.
Step 7 Click Password Policy.
You can configure the password policies to meet complexity requirements and set the minimum
length of the password.
TIP You should apply the password complexity settings before you install the Cisco
CallManager application. If the passwords applied in the installation process do not fit the
complexity requirements, the Cisco CallManager services will no longer be able to start.
Account and Password Considerations

When giving individual users the ability to log in to the Cisco IP Telephony Operating System as
administrators, you should create a separate account for each user and put each into the
Administrators group. Doing so enables tracking of changes made to the Cisco IP Telephony
Operating System. In addition, you could change the default administrator account to a decoy
Common Windows Exploits 475
Administrator account that has no rights but is strictly monitored (by enabling auditing of login
attempts or usage of that account).
NOTE Cisco CallManager installations and upgrades currently require the Administrator
account to be used. Before installing or upgrading Cisco CallManager, rename the decoy
Administrator account and change the name of the real Administrator account back to
“Administrator” on all Cisco CallManager servers in the cluster.
Some corporate security policies require separating the system auditors from the system
administrators. To enable more accurate auditing information regarding the identity of an
administrator, it is a good practice to create individual accounts for each administrator and make
them members of the Administrator group. In addition, separate administration from auditing by
creating separate auditor accounts. Auditor accounts should have full rights to logs but should not
have any other administrative permission, whereas administrator accounts should have only read
access to log files.
Follow general security guidelines for accounts and passwords, such as removing unnecessary
accounts and requiring complex passwords, but also harden the server by applying password
protection to complementary metal oxide semiconductor (CMOS) access, screen savers, and
Hewlett-Packard Integrated Lights-Out (iLO) access (used for out-of-band server management).
NOTE More information on iLO can be found at http://h71028.www7.hp.com/enterprise/

cache/98327-0-0-225-121.aspx.
Common Windows Exploits

A hardened Cisco IP Telephony Operating System can successfully defend against many common
Windows exploits. Some active services cannot be disabled because Cisco CallManager uses
them. To secure these areas, you must design the IP telephony-ready network properly and choose
the proper roles for the Cisco CallManager nodes in the cluster. You need to protect Windows
against some of the most common exploits.
One common exploit involves Extensible Markup Language (XML) applications running on
HTTP (TCP port 80). Most XML applications go to the Internet to get their data. Because of this,
Cisco recommends that you off-load XML services to a dedicated server that is isolated (as much
as possible) from the rest of the network.
The most important task for Microsoft IIS issues is to turn off IIS on all subscribers. IIS is the
parent process for HTTP, Simple Mail Transfer Protocol (SMTP), and FTP. Eighty percent of the
attacks against Windows are against the IIS parent process. Turn off IIS on the subscribers, where
all of the active call processing is taking place, and run it only on the publisher for administration
purposes. This practice will minimize the threats against Windows by 80 percent and actually
bring it closer to parity with what is considered to be the normal security settings of UNIX or
Linux operating systems.
In a Cisco CallManager cluster, different servers can have different roles and, hence, do not need
the same active services. One server could act as a pure management server by providing access
only to Cisco CallManager Administration web pages, while other servers are providing call-
routing functions and others are being used for applications such as phone services. Because IIS
is a common target, run it only where needed: at the Cisco CallManager Publisher. During
upgrades, IIS will also be needed on subscribers but will automatically be started when needed as
long as the service is set to manual rather than disabled. Therefore, set IIS to manual on all
subscribers and keep the setting automatic only at the publisher.
CAUTION IIS needs to be available during upgrades. If you have set the IIS Startup Type
option to Disabled, the upgrade will fail.
Table 20-1 shows what will happen during a Cisco CallManager upgrade when the IIS service is
set to different options.
Table 20-1 Behavior of Cisco CallManager During an Upgrade
IIS Service Resulting Upgrade Behavior

Parameter
Enabled The upgrade will work with no interference.
Disabled The upgrade will fail; no message is displayed.
Manual and The upgrade will stop, a message that the IIS is not running will pop up, the IIS
Stopped service will start, and the upgrade will continue.
On the next reboot, the IIS service will be in the Manual and Stopped state again.
Manual and The upgrade will work with no interference.
Running
Finally, to avoid attacks against the Dynamic Host Configuration Protocol (DHCP) server, which,
in most installations, is used to provide IP settings, push DHCP services as close to the endpoints
as possible. This might include using an intelligent Cisco switch or router for DHCP services.
Security Taboos
After you have protected Cisco CallManager against common Windows exploits such as attacks
on IIS, there are additional security settings that, from the point of view of a Windows
Security Taboos 477
administrator, are nice to have but that are not recommended in a Cisco CallManager environment.
Some of the settings that a Windows administrator would normally implement on the servers could
cause the Cisco CallManager installation to fail, increase the system downtime of the Cisco
CallManager server, and delay troubleshooting efforts.
Table 20-2 describes common Windows security settings never to be done in any circumstance on
a Cisco CallManager server.
Table 20-2 Windows Security Settings Not to Be Done in Any Circumstance
Security Setting Reason It Is Not Allowed

Do not delete, disable, or rename any Services like Cisco CallManager or SQL might not
service accounts. function.
Do not change “password never Services might not start.
expires” on service accounts created by
Cisco.
Do not change any file, folder, or A high probability exists that Cisco CallManager will
Registry key permissions unless not function.
documented.
Do not set the CMOS power-on The server will not boot after power failure until the
password. password is entered.
Do not delete, disable, or rename the Some IIS virtual directories require them.
IUSR_Guest or IWAM_Guest accounts.
Do not disable parent paths in IIS. Some Cisco CallManager web pages will no longer
work.
Do not remove the IIS application Some Cisco CallManager web pages will no longer
mappings for .asp, .cer, .cdx, and .asa. work.
Do not install unapproved third-party Issues with Cisco support arise.
software, utilities, or agents.
In addition to the security settings never to be implemented, Cisco also lists a few security settings
that are not recommended. Table 20-3 describes these settings.
Table 20-3 Not Recommended Windows Security Settings
Security Setting Reason It Is Not Recommended

Join an Active Directory domain Role-based administrator not supported
Complexity of Active Directory group policies (too many

possible permutations)
Shutdown if Unable to Write Not ideal for a strategic application
Security Log
continues
Table 20-3 Not Recommended Windows Security Settings (Continued)
Security Setting Reason It Is Not Recommended

Disable crash control Disabling Dr. Watson crash dumps adds complexity to
forensic troubleshooting
Convert disk D from FAT to NTFS Same server recovery will not work
Clear page file at reboot Reboots can take 30 minutes or longer
Account lockout after N failed Disables low-level service accounts
login attempts
NOTE Cisco CallManager Release 4.x uses the following service accounts: CCMCDR,
CCMService, CCMServiceRW, CCMUser, and SQLSvc.
CAUTION If the Cisco CallManager is integrated into the Microsoft Active Directory, the
installation will lose its Cisco TAC support.
Summary
This chapter covered the security concepts and protection of the foundation Cisco CallManager
operating system. There are many security threats targeting Microsoft Windows operating
systems. Because of this, Cisco has modified the default Windows 2000 security settings and
released their own Cisco IP Telephony Operating System, prehardened against many common
attacks and intrusions. To enhance this security, you should consider using the optional security
scripts located on the CallManager hard drive. Adding approved antivirus protection software can
prevent intrusion by viruses, worms, and Trojan horses.
Cisco also provides a free, standalone version of their Cisco Security Agent, preconfigured for a
CallManager installation. Installing this can protect the server from network intrusions, software
modifications, and DoS attacks.
In addition to all these security supplicants, you should also consider using complex passwords
and account policies for all administrative and auditing accounts. However, you should not rename
accounts or modify passwords of services related to the Cisco CallManager. Cisco also has a list
of security taboos that should be avoided on the Cisco CallManager to ensure you do not invalidate
your TAC support contracts.
Review Questions
1. When are critical hot fixes and patches to the Cisco IP Telephony Operating System posted
on Cisco.com for download?
a. 24 hours after the announcement from Microsoft
b. Monthly in a consolidated security release
c. With the next operating system upgrade
d. These should be downloaded from Microsoft as soon as they appear.
2. Which feature should not be enabled when using antivirus protection software?
a. Full-scan
b. Heuristic scan
c. E-mail scan
d. Pagefile scan
3. Which Cisco-provided software tool protects Cisco CallManager against malicious
applications?
a. CDR
b. CER
c. CSA
d. CRM
4. Which parameter has to be set on all service accounts?
a. Complex password requirement
b. Minimum password length of six characters
c. Password never expires
d. Enforce password history
5. What Microsoft service is most commonly attacked?
a. DNS service
b. DHCP service
c. IIS service
d. Active Directory service
6. Which setting on the Cisco IP Telephony Operating System is supported on the CallManager
platform but not recommended by Cisco?
a. Delete the IUSER_Guest account
b. Delete SQL service accounts
c. Install third-party utilities
d. Disable Dr. Watson
7. What automated method does Cisco support for security updates and hot fixes to the
CallManager server?
a. Receiving e-mail updates from the Cisco CallManager Notification Tool
b. Downloading updates directly from Microsoft using the Windows Update Services
c. Using the CSA standalone automatic update feature
d. Downloading updates directly from Cisco using the automated Cisco Update Services
8. Which of the following files represent an automated security script that will add additional
Cisco approved security to the CallManager server? (Choose two.)
a. CCM-OS-OptionalSecurity.cmd
b. CSA_SecurityScript.cmd
c. IPSec-W2KSQL.cmd
d. SecurityTemplace_CCM4xx.cmd
9. Which of the following antivirus platforms is NOT supported by Cisco for installation on a
Cisco CallManager platform?
a. McAfee VirusScan Enterprise
b. Symantec AntiVirus Corporate Edition
c. Trend Micro ServerProtect
d. WebX SecureServer
10. Which of the following does the headless CSA protect against? (Choose two.)
a. Operating system file integrity
b. Restriction of network-aware, locally installed applications to local resources
c. Protection against virus infection
d. Protects against preconfigured inbound network connections to the CallManager server
following topics:
■ Explaining the threats targeting remote Cisco

CallManager Administration and other
applications
■ Explaining how HTTPS provides secure
remote communication and logins to Cisco
CallManager Administration
■ Describing HTTPS certificate details, using
Microsoft IIS to save a certificate to a trusted
folder, and copying the certificate to a file
■ Describing how MLA provides multiple
levels of security to Cisco CallManager
Administration
■ Enabling MLA and assigning a password to
the superuser
■ Defining a functional group and identifying
the two types of functional groups
■ Defining a user group and identifying how its
privileges are mapped to functional groups
■ Creating a new functional group and a new
user group and assigning an access privilege
level to the members of the user group
CHAPTER 21
Securing Cisco Unified
CallManager Administration
By securing communication with the Cisco CallManager using Secure HTTP (HTTPS), a
hacker cannot listen to the data stream during HTTP sessions. In addition, in large voice
environments, it is important to distribute the various administration tasks. Cisco CallManager
Multilevel Administration Access allows you to assign different authorization levels to
administrators. This chapter discusses both the concepts and configuration of HTTPS and MLA
on your Cisco CallManager servers.
Threats Targeting Remote Administration

In releases earlier than Cisco CallManager Release 4.1, HTTP is the standard protocol for
accessing the Cisco CallManager Administration web pages. If an attacker intercepts the
connection and looks for the username and password of the administrator, the attacker can find
the relevant information easily because CallManager does not encrypt the connection.
Beginning with Cisco CallManager Release 4.1, HTTPS (RFC 2818) is the standard protocol
for accessing the Cisco CallManager Administration pages, without installing or configuring
any additional security parameters.
To add to the security woes, the default Cisco CallManager Administrator account is the same
as the Microsoft Windows Administrator account. If a hacker learned this login information, the
hacker could not only access the Cisco CallManager Administration pages, but could also log
in to the operating system of the Cisco CallManager server with full access to all information.
To address this issue, Cisco has added Multilevel Administration Access (MLA), giving
CallManager an alternate user database to manage administrative privileges.
Securing CallManager Communications Using HTTPS

HTTPS secures communication between the browser on the client PC and a web server. It
allows authentication of the web server (to ensure the client is not accessing an impersonating
website) and protects communication between the client and the web server. All packets are
signed to provide integrity, so that the receiver has a guarantee that the packets are authentic and
have not been modified during transit. In addition, all packets are encrypted to provide privacy,
484 Chapter 21: Securing Cisco Unified CallManager Administration
so that sensitive information can be sent over untrusted networks. These Cisco CallManager
applications support HTTPS:
■ Cisco CallManager Administration
■ Cisco CallManager Serviceability
■ Cisco IP Phone User Options web pages
■ Bulk Administration Tool (BAT)
■ Tool for Auto-Registered Phones Support (TAPS)
■ Cisco Call Detail Record (CDR) Analysis and Reporting (CAR)
■ Trace Collection Tool
■ Real-Time Monitoring Tool (RTMT)
When you are using HTTPS for browsing to Cisco CallManager Administration and user options
web pages, communication is secure. A hacker who sniffs the communication will find it very
difficult to re-create any information from the sniffed packets.
HTTPS secures not only the username and passwords in the communication, but also
configuration changes in Cisco CallManager Administration and other applications, such as Cisco
CallManager Serviceability. If a user configures parameters such as call forwarding or speed dials
on the user options web pages, the client and IIS communicate in a secure way.
HTTPS Certificates
HTTPS uses certificates for web server authentication. Certificates provide information about a
device and are signed by an issuer, the Certificate Authority (CA). By default, Cisco CallManager
uses a self-signed certificate, but it also allows you to use a certificate issued by a company CA or
even an external CA such as VeriSign. The file where the Cisco CallManager HTTPS certificate is
stored is C:\Program Files\Cisco\Certificates\httpscert.cer.
TIP A self-signed certificate provides the same functions as a certificate issued by a

recognized CA. The only problem that occurs with a self-signed certificate is that client web
browsers issue warning or caution messages the first time they access the secured website. For
internal and intranet server use, this should not cause any major problems.
Securing CallManager Communications Using HTTPS 485
The certificate will be used on the IIS default website that hosts the Cisco CallManager virtual
directories, which include the following:
■ CCMAdmin and CCMUser
■ CCMService
■ Administration Serviceability Tool (AST)
■ BAT and TAPS
■ RTMTReports
■ CCMTraceAnalysis
■ PktCap
■ Administrator Reporting Tool (ART)
■ CCMServiceTraceCollectionTool
To use a certificate issued by a CA after a Cisco CallManager installation or upgrade, delete the
self-signed certificate and install the CA signed certificate instead.
NOTE For more information on how to obtain a certificate from an external CA, contact a
vendor of Internet certificates such as VeriSign or consult with the administrator of your
company CA (if using your own CA).
Accessing CallManager When Using Self-Signed Certificates

The first time that a user accesses Cisco CallManager Administration or other Cisco CallManager
applications after the Cisco CallManager Release 4.1 installation or upgrade from a browser
client, a Security Alert dialog box (shown in Figure 21-1) asks whether the user trusts the server.
When the dialog box appears, clicking the buttons results in these actions:
■ Yes—Trust the certificate for the current web session only. The Security Alert dialog box will
display each time you access the application.
■ No—Cancel the action. No authentication occurs, and the user cannot access the Cisco
CallManager Administration pages.
■ View Certificate—Start certificate installation tasks, so that the certificate is always trusted.
After you install the certificate, the Security Alert dialog box no longer appears when you
access the Cisco CallManager Administration pages.
Figure 21-1 Self-Signed Certificate Security Alert
Click the View Certificate button. The Security Alert dialog box appears and the Certificate
window opens, shown in Figure 21-2. The General tab shows brief information about the
certificate, such as the issuer and the validation. For more detailed information, click the Details
tab. Another way to get information about the certificate is to check the certificate directly on the
Cisco CallManager. On the Cisco CallManager publisher, right-click the certificate name in
C:\Program Files\Cisco\Certificates\httpscert.cer and choose Open. It is not possible to change
any data in the certificate.
Figure 21-2 Viewing the SSL Certificate

To keep from seeing the security warning each time you navigate to the Cisco CallManager server,
click the Install Certificate button. By walking through the Microsoft Windows Certificate Import
Wizard, you will import the CallManager self-signed certificate into your local certificate store.
This keeps Microsoft Internet Explorer from prompting you with a security warning each time you
access the CallManager Administration interface.
Multilevel Administration
Prior to the availability of MLA, there was only one administrator login. Administrators had full
read and write access to Cisco CallManager configuration. An administrator could change any
parameter in the database or directory that is accessible through the Cisco CallManager
Administration and Cisco CallManager Serviceability pages. The entire system could be disabled
with a few mouse clicks that accidentally modified data to which the user did not need access.
MLA provides multiple levels of security to Cisco CallManager Administration. Cisco

CallManager menus are grouped in functional groups. Users are grouped in user groups. MLA
permits you to grant only the privileges required to a selected group of users and to limit access to
the configuration menu for a particular user group.
Different access levels can be assigned to each functional group, such as no access, read-only
access, and full access. The access rights can be set for every configured user group. MLA also
provides audit logs of user logins and access to and modifications to Cisco CallManager
configuration data.
Before installing MLA, Cisco CallManager administrators logged in using a local Windows 2000
Administrator account. After you have enabled MLA, usernames and passwords are stored in a
Lightweight Directory Access Protocol (LDAP) directory and provide the basis for login
authentication.
During enabling, MLA creates a predefined user called CCMAdministrator. The Windows
Registry stores the user ID and the encrypted password of the CCMAdministrator user. Thus, even
when the LDAP directory is unavailable, the CCMAdministrator user can log in to Cisco
CallManager Administration. Only the CCMAdministrator ID and password are stored in the
Windows Registry.
MLA was introduced with the Cisco CallManager Release 3.2(2c) and had to be separately
installed. With Cisco CallManager Release 4.0 and later, MLA is integrated in Cisco CallManager
but disabled by default.
NOTE If you are upgrading from an earlier version of Cisco CallManager to Cisco
CallManager release 4.0 or later and already have MLA installed and enabled, CallManager
migrates your existing MLA configuration to the new MLA version and keeps it enabled.
However, the upgrade process resets the existing CCMAdministrator password to a random
password (displayed at the end of the upgrade).
Enabling MLA
The Enable MultiLevelAdmin enterprise parameter designates whether MLA is enabled. This
enterprise parameter can be found in the Cisco CallManager menu User > Access Rights >
Configure MLA Parameters, shown in Figure 21-3. You can set the Enable MultiLevelAdmin
parameter to True (enabled) or False (disabled); False is the default value.
Figure 21-3 Enabling MLA
When you choose True, enter a new password at the New Password for CCMAdministrator
prompt and reenter the password at the Confirm Password for CCMAdministrator prompt. Only
the CCMAdministrator user can now log in to Cisco CallManager; the Windows NT Administrator
account no longer has access rights to Cisco CallManager Administration.
When the Enable MultiLevelAdmin enterprise parameter value is modified, the World Wide Web
Publishing Service has to be restarted. Then, reopen the browser and reauthenticate with Cisco
CallManager by using the new CCMAdministrator account.
MLA Functional Groups

Cisco MLA uses two group management functions: user groups and functional groups. A user
group simply contains user accounts. A functional group consists of a collection of Cisco
CallManager system administration submenus. All the web pages that compose each functional
group belong to a common administrative menu. Two types of functional groups exist:
■ Standard functional groups, which are the default functional groups
■ Custom-based functional groups
Standard functional groups are created as a part of MLA during Cisco CallManager installation
and cannot be modified or deleted. They contain typical permissions assigned to sublevel Cisco
CallManager administrators. You can define your own custom-based functional groups to allow a
group of administrators access to specific Cisco CallManager Administration menus.
When you enable Cisco CallManager MLA, a complete set of standard functional groups becomes
available (shown in Figure 21-4):
■ Standard Plugin
■ Standard User Privilege Management
■ Standard User Management
■ Standard Feature
■ Standard System
■ Standard Service Management
■ Standard Service
■ Standard Serviceability
■ Standard Gateway
■ Standard RoutePlan
■ Standard Phone
Figure 21-4 Built-In MLA Functional Groups
CAUTION In the Standard System functional group, all submenus of the Cisco CallManager
System menu, such as Server, Cisco CallManager, Cisco CallManager Group, and so on, are
enabled. A user with full access rights in the Standard System functional group could, for
example, change the IP address of the server. Be careful when you assign access rights to the
fundamental Cisco CallManager menus.
MLA User Groups

Various user groups are predefined and have no members assigned when you enable MLA. The
CCMAdministrator user can add users to these groups, set the access rights for the user groups,
and configure additional named user groups as needed. To add users, select the relevant user group
and click the Add a User to Group hyperlink in the upper-right corner of the administration
window.
These user groups are created at the time of installation (shown in Figure 21-5):
■ PhoneAdministration
■ ReadOnly
■ ServerMonitoring
■ SuperUserGroup
■ ServerMaintenance
■ GatewayAdministration
Figure 21-5 Built-In MLA User Groups
Assigning MLA Access Privileges

After users are added to a user group, access privileges are then set for each functional group. The
functional group defines the Cisco CallManager menus that can be used by the relevant user group.
Figure 21-6 shows the items in the CallManager Administration Device menu that are enabled for
the Standard Phone functional group.
Figure 21-6 Device Menu Access for the Standard Phone Functional Group
As shown in Figure 21-7, the PhoneAdministration user group has full access rights to the
Standard Phone and the Standard User Management functional groups. This means that a user
assigned to the PhoneAdministration user group can add, change, or delete the configuration of
the computer telephony integration (CTI) points and phones. A user in the PhoneAdministration
user group can also access all Device Settings submenus.
Creating New MLA Functional and User Groups

Cisco CallManager allows you to configure MLA groups outside the built-in defaults to provide
custom administration functions for your organization. For example, imagine you had a sublevel
administrator who should only have rights to change the music on hold preferences for your users.
To configure a custom administrative level for this administrator, you could follow these general
steps:
Step 1 Create a new functional group and assign privileges.

Step 2 Create a new user group and assign privileges.
Step 3 Verify proper operation.
Figure 21-7 Assigning Rights to User Groups
Initially, you would need to create a new functional group for the subadministrator defining the
areas of the CallManager Administration to which they have access. You could follow this
procedure to add a new functional group:
Step 1 Choose User > Access Rights > Functional Group, and click Add a New
Functional Group.
Step 2 In the Functional Group Name field, enter the name of a new functional
group (in this example, Music Only, shown in Figure 21-8).
Figure 21-8 Adding a New Functional Group
Step 3 Check the check box next to the menu that you want to include in the new
functional group. In this example, check only the Media Resource-
>Music on Hold Audio Source check box under Service, as shown in
Figure 21-9, and click Insert. Figure 21-9 results from scrolling down near
the bottom of the screen shown in Figure 21-8.
Figure 21-9 Assigning Functional Group Menus
After you have added the functional group to the configuration, you can create a user group for the
administrator and assign the necessary functional group privileges. To create the user group,
complete the following steps:
Step 1 Choose User > Access Rights > User Group, and click Add a New User
Group.
Step 2 In the User Group Name field, enter the name of the new user group (in
this example, MusicDJ), and click Insert.
Step 3 Click Add a User to Group, and enter the username (which must already
have been configured in Cisco CallManager Administration). In this
example, the user JeremyD was chosen from the directory, shown in
Figure 21-10.
Figure 21-10 Adding a New User Group
Now that you have created the user group, you can assign the necessary privileges to the music
administrator:
Step 1 Choose User > Access Rights > Assigning Privileges to User Group.
Step 2 Click the name of the user group to which you want to assign privileges. In
this case, the MusicDJ user group is chosen.
Step 3 Choose the privilege level for each functional group within the user group.
Three privilege levels are available from the drop-down list: No Access,
Read Only, and Full Access. In this case, the MusicDJ user group is
assigned Full Access for the functional group Music Only (shown in Figure
21-11). After you have selected the necessary permissions for the
functional groups, click Update.
Figure 21-11 Assigning User Group Permissions
Finally, the last step in this configuration is to verify and test the privileges. To verify whether a
specific user has the correct access rights, click the Key symbol in the Permission column in the
User Group Configuration window. In Figure 21-12, the user JeremyD was chosen, and, therefore,
the MusicDJ user group is displayed with the configured access rights. The privileges report shows
that JeremyD has no access to any functional group except the Music Only group, to which he has
full access.
Figure 21-12 Verifying User Permissions
To test the privileges, you can log in using a user account assigned to the specific user group you
want to test. After logging in, attempt to access allowed and disallowed areas of the CallManager
Administration. Disallowed pages will appear as shown in Figure 21-13. Allowed pages will
function normally, unless you have assigned them Read-Only access.
Summary 499
Figure 21-13 Testing User Permissions
Summary
This chapter covered the security concepts and protection of the Cisco CallManager
Administration web pages. In CallManager versions earlier than 4.1, HTTP was the rule for
administration access. This could open the CallManager servers to packet sniffing attacks. HTTPS
secures the connection between the browser and IIS. With CallManager releases 4.1 and later, the
server uses a self-signed certificate, which protects communication, but can cause warning
messages in client web browsers.
To increase the administrative security of the system, Cisco has released the Multilevel
Administration Access (MLA) utility for Cisco CallManager 3.2 and later. MLA allows you to
give users different access rights for Cisco CallManager Administration. Upon enabling MLA, the
old Windows 2000 Administrator account will be disabled from CallManager access. Standard
MLA functional groups are created with the MLA installation and can be supplemented by custom
functional groups. User groups unite access privileges to functional groups.
Review Questions
1. What is the danger when browsing unsecured and without MLA to the Cisco CallManager
Administration window?
a. There is no risk.
b. With a sniffed username and password, a hacker can log in to the Cisco CallManager
Administration window only.
c. With a sniffed username and password, a hacker can log in to the Cisco CallManager
Administration window as well as to the operating system.
d. The hacker can only listen to the conversation.
2. How do you browse securely to the CallManager Administration window?
a. http://CM-IP/SecureCCMAdmin
b. https://CM-IP/CCMAdmin
c. https://CM-IP/TTL/CCMAdmin
d. https://CM-IP/SSL/CCMAdmin
3. Which of the following is not a valid access level in MLA?
a. No access
b. Read-only
c. Read-write
d. Full access
4. After you enable MLA, what is the new administrator account?
a. MLAAdministrator
b. Administrator
c. Windows NT Administrator account
d. CCMAdministrator
5. Which functional group is not a default group?
a. Standard Device
b. Standard Service
c. Standard Gateway
d. Standard Plugin
e. Standard RoutePlan
6. Which user group is not a default group?

a. PhoneAdministration
b. FullAccess
c. SuperUserGroup
d. GatewayAdministration
e. ServerMaintenance
7. By default, client web browsers using HTTPS-secured administration will receive warning
messages. Why do these warning messages exist?
a. Because the Cisco CallManager uses 40-bit encryption
b. Because the Cisco CallManager uses a self-signed certificate
c. Because the Cisco CallManager uses weak encryption
d. Because there is a name conflict on the certificate
8. Where are the normal user accounts for MLA stored?
a. LDAP Directory
b. A local file
c. CallManager Administration
d. The Windows Registry
9. Where is the CCMAdministrator user account for MLA stored?
a. LDAP Directory
b. A local file
c. CallManager Administration
d. The Windows Registry
10. What is the default state of Cisco MLA? (Choose two.)
a. Disabled for a clean CallManager install
b. Enabled for a clean CallManager install
c. Enabled if upgrading from a previous MLA version that was enabled
d. Disabled if upgrading from a previous MLA version that was enabled
following topics:
■ Explaining how legitimate devices can be

exploited for fraudulent use of the IP PBX
system to make toll calls
■ Using partitions and calling search spaces,
restricting call forwarding based on user
classes, and restricting voice-mail transfers to
internal destinations
■ Explaining the usage of blocking commonly
exploited area codes
■ Configuring Cisco CallManager to route calls
to different locations based on the time of day
when a call is made
■ Designing and implementing FAC to require
user authorization for different classes of
calls
■ Providing external call transfer blocking by
setting service parameters and configuring
gateways, trunks, and route patterns as
OffNet devices
■ Configuring Cisco CallManager
Administration to drop a conference call
when the conference creator leaves the call or
when the last OnNet party leaves
CHAPTER 22
Preventing Toll Fraud
Business consumers have more control over their telecommunications services than ever before.
New technologies provide more information and more flexibility in how businesses use their
telecommunications services. Unfortunately, the shift in control has made businesses and
telephone companies more susceptible to toll fraud and led to an increase in fraud. This chapter
discusses ways to prevent toll fraud in a company.
Toll Fraud Exploits

A company telephony system can be subject to toll fraud by company employees or by external
people who try to find vulnerabilities in the system. The first group, employees, simply ignores
policies, hoping that their activities will not be detected because it is difficult to differentiate
between business calls and private calls based on the dialed number. The other group of people,
the external callers, is more technically oriented. They try to find vulnerabilities in network
devices, including IP telephony systems. Sometimes, they do not even specifically look for
voice systems; they just exploit whatever system over which they can get control.
The main difference between these two groups is the way in which you can mitigate the “attack.”
In the case of external attackers, the key is to prevent unauthorized access to the system and its
devices. For authorized users of the system, the administrator has to very carefully limit the
technical abilities and features of the system without compromising the flexibility and efficiency
of its users.
There are also some features in a telephony system that can be misused. These include call
forward and call transfer settings and voice-mail transfer options. If the features that are
commonly used for toll fraud are well protected, users might try to exploit the system using
other features. As an example, if a user is not allowed to transfer an external call to another
external destination, the user could try to set up a conference call for these two parties and then
leave the conference.
Usually, an administrator has to accept the fact that toll fraud cannot be eliminated completely.
The only way to achieve complete elimination would be to block all external calls and disable
all features that would allow employees to place calls outside the company. This technique
might be feasible for single-function telephones, such as public telephones located in a lobby,
504 Chapter 22: Preventing Toll Fraud
but is not desirable for telephones used by standard employees. Therefore, only those calls that
can be clearly identified as nonbusiness calls will be blocked. However, in many cases, you cannot
judge in advance whether the call being placed is business-related or private.
Figure 22-1 shows different types of toll fraud.
Figure 22-1 Forms of Toll Fraud

Toll Fraud 1: Call Forward All Toll Fraud 2: Transfer from Voice Mail
Call me at my work number
while I am on vacation.
Voice mail,
International, transfer me to International,
9011xxxxxxx.
Premium Premium
V V
Local Forward All Local
PSTN PSTN
IP IP
Toll Fraud 3: Social Engineering Toll Fraud 4: Inside Facilitators
Please transfer
my call to International, International, I will transfer
extension 9011. your call.
Premium Premium
V V
Local Local
PSTN PSTN
IP IP
The following list explains these types of toll fraud:
■ Call Forward All (CFA)—The first example describes a scenario in which an employee
forwards the office number to, for example, an international or mobile number. This employee
then tells friends to call the office number. The call is forwarded to the number that the
employee specified, making the company pay the costs of the calls.
Preventing Call Forward and Voice-Mail Toll Fraud Using Calling Search Spaces 505
■ Transfer from voice mail—The second toll fraud example shows an attacker making an
external call to the voice-mail system, which forwards the call to an international premium
destination. The attacker is billed only for a local call, whereas the company, from which the
call is forwarded, pays for the international call.
■ Social engineering—The third example shows a scenario in which an attacker calls from
outside the company and uses social engineering tricks (for instance, pretending to be an
employee working from home) to be transferred to an external number, such as 9011. The 011
prefix (plus 9 being the typical number dialed in corporations for outside dial tone) is used in
the United States to place international calls. This attacker is also charged only for a local call,
whereas the company again pays for the connection to an international telephone number.
■ Inside facilitators—The fourth example is very similar to the third one. But in this case, an
employee inside the company transfers the external call to another external number. In this
case, the toll fraud has an internal source.
Preventing Call Forward and Voice-Mail Toll Fraud Using Calling

Search Spaces
Call Forward All (CFA) is a feature in Cisco CallManager that allows an internal number (for
example, an employee office number) to be forwarded to an external number (for example, an
international number, mobile number, or premium number). For example, an employee can call
the office number, which is then forwarded to the number specified in the forwarding field. This
number can be an international or premium number. The user can configure the setting using the
web interface, so the forwarding configuration can be set up and removed very easily from home
or elsewhere. CFA exploits can be avoided by applying a calling search space to the CFA feature.
As shown in Figure 22-2, the administrator has applied a calling search space named IntLocCSS
(which allows only internal and local PSTN calls) to the CFA field.
Figure 22-2 Restricting Call Forward All
NOTE The only call forwarding ability available to the user (from either the CCMUser pages
or the IP phone) is Call Forward All. By applying the calling search space to only this field, you
have effectively restricted user forwarding.
Voice-mail systems, which can transfer a call to an extension, can be misused in a similar way if
they are configured to allow transfer of calls when the called party is not available. If such transfers
are not limited, a caller could connect to the voice-mail system by a local call and then transfer to
the public telephony network (for example, a long-distance number). Voice-mail forwarding
exploits can be avoided by applying a calling search space to the voice-mail port in the Cisco
CallManager configuration. As shown in Figure 22-3, the administrator has applied the same
IntLocCSS calling search space to the voice-mail port.
Blocking Commonly Exploited Area Codes 507
Figure 22-3 Restricting Voice-Mail Forwarding
Blocking Commonly Exploited Area Codes

When blocking commonly exploited area codes, create a unique route pattern for each area code
that you want to block. You can create different restriction levels; for example, general employees
are not allowed to call these numbers, but executives and managers are allowed to. Use different
route filters, partitions, and calling search spaces to generate restriction levels. As a general
recommendation to prevent toll fraud, you should block as many numbers as possible. For the
example shown in Figure 22-4, all calls to the Bahamas are blocked. For each number that you
want to block, create a route pattern that explicitly blocks the number. Often, the decision as to
whether a number should be blocked or allowed depends on company policies or simply on
whether it is necessary to call the number.
Figure 22-4 Restricting Specific Area Codes
Table 22-1 shows some of the most commonly exploited area codes that you might want to block.
It is not an exhaustive list, and some of these area codes might not apply to your organization.
Table 22-1 Commonly Exploited North American Area Codes
Country Area Code Blocked Cisco CallManager Pattern

Anguilla 264 9.1264xxxxxxx
Antigua/ 268 9.1268xxxxxxx
Barbuda
Bahamas 242 9.1242xxxxxxx
Barbados 246 9.1246xxxxxxx
Bermuda 441 9.1441xxxxxxx
British Virgin Islands 284 9.1284xxxxxxx
Cayman Islands 345 9.1345xxxxxxx
Dominica 767 9.1767xxxxxxx
Dominican Republic 809 9.1809xxxxxxx
Using Time-of-Day Routing 509
Table 22-1 Commonly Exploited North American Area Codes (Continued)
Country Area Code Blocked Cisco CallManager Pattern

Grenada 473 9.1473xxxxxxx
Jamaica 876 9.1876xxxxxxx
Montserrat 664 9.1664xxxxxxx
Puerto Rico 787 9.1787xxxxxxx
St. Kitts & Nevis 869 9.1869xxxxxxx
St. Lucia 758 9.1758xxxxxxx
St. Vincent & the 784 9.1784xxxxxxx
Grenadines
Toll Charge 900 9.1900xxxxxxx
976 9.1976xxxxxxx
Trinidad & Tobago 868 9.1868xxxxxxx
Turks & Caicos Islands 649 9.1649xxxxxxx
U.S. Virgin Islands 340 9.1242xxxxxxx
In the worldwide country code numbering scheme, several countries do not use their own country
codes.
These numbers have the same format as the NANP—[access code] [area code] [number] (for
example, 9.142xxxxxxx)—but a call to one of these destinations results in an international toll
charge.
Administrators should make sure that all of the devices in the IP telephony network can reach only
the destinations that they should be able to reach. For example, a lobby phone should not be able
to call international numbers. In situations in which individuals in your organization have
legitimate business in one of these countries, the recommendation is to explicitly configure route
patterns that will match those businesses, while still blocking the area code as a whole.
Using Time-of-Day Routing

With the 4.1 release of Cisco CallManager, time-of-day routing routes calls to different locations
based on the time of day when a call is made. For example, User A wants to configure the CFA
number to an international number. User A can set this number only during the 8:00 a.m.-to-5:00
p.m. time period because outside these hours the system does not find the international number in
the partition that is used to validate the CFA number.
Time-of-day routing can be used for other purposes as well. With time-of-day routing, a least-cost
routing system can be designed to choose the cheapest provider for a specific time of day. For
example, assume that there are two telephony providers, ABC and XYZ. Between 8 a.m. and 1
p.m., provider ABC is the cheaper provider, so Cisco CallManager routes calls over this provider
during that period. After 1 p.m., provider XYZ becomes the cheaper provider, so the Cisco
CallManager uses this provider then.
NOTE A complete description of time-of-day routing and its configuration can be found in
Chapter 13, "Implementing Telephony Call Restrictions and Control."
Using FAC and CMC

Forced Authorization Codes (FAC) became available in Cisco CallManager Release 3.3(4). It is
not included in Cisco CallManager Release 4.0 but has been included since Cisco CallManager
Release 4.1. In Cisco CallManager Administration, various levels of authorization can be
configured. With FAC, sensitive destinations can be “secured” by requiring use of authorization
codes for such destinations. When a call is routed through a FAC-enabled route pattern, Cisco
CallManager plays a tone and requests an authorization code. If the authorization code entered by
the user does not meet or exceed the level of authorization that is specified to route the dialed
number, the user receives a reorder tone. If the authorization is accepted, the call is routed. The
authorization is logged to call detail records (CDRs) so that the information can be used by the
CDR Analysis and Reporting (CAR) tool to generate reports for accounting and billing.
FAC is useful for colleges and universities or any business or organization for which limiting
access to specific classes of calls proves beneficial. An additional benefit is that when you assign
unique authorization codes, the users who can place calls can be determined. For example, for
each user, a unique authorization code can be specified.
Enable FAC for relevant route patterns by checking the appropriate check box and specifying the
minimum authorization level for calls through that route pattern. After updating the route patterns
in Cisco CallManager Administration, the dial plan documents have to be updated to define the
FAC-enabled route patterns and configured authorization level.
To implement FAC, devise a list of authorization levels and corresponding descriptions to define
the levels. Authorization levels must be specified in the range of 0 to 255. Cisco allows
authorization levels to be arbitrary, so define what the numbers mean for your organization. Before
defining the levels, review the following examples of levels that can be configured for a system:
■ Configure an authorization level of 10 for intrastate long-distance calls in North America.
■ Because interstate calls often cost more than intrastate calls, configure an authorization level
of 20 for interstate long-distance calls in North America.
■ Configure an authorization level of 30 for international calls.
TIP Incrementing authorization levels by 10 establishes a structure that provides scalability

when more authorization codes need to be added. The range for authorization codes is from 0
to 255.
Client Matter Codes (CMC) can also be used by companies to keep track of private calls placed
by their employees. For example, a company could allow employees to place private calls using
the company telephony infrastructure but require the employee to pay the cost. External route
patterns (for example, those for long-distance and international calls and those for the 900 area
code) can be configured to request a Client Matter Code to be entered and, therefore, to be logged
accordingly.
This feature does not prevent users from making private calls using business telephones, but it
allows a company to have a policy that does not deny private calls in general but requests that users
identify them as such and pay for them. In both situations (denying private calls in general or
permitting them if properly flagged), additional tools (logging, reporting) are needed to detect
improper usage.
When CMC is configured, users hear a tone prompting them to enter any valid Client Matter Code.
The CDR will include the code that is entered for later processing.
CMC was first available in Cisco CallManager Release 3.3(4). It is not included in Cisco
CallManager Release 4.0 but has been included since Cisco CallManager Release 4.1.
NOTE A complete description of Forced Authorization Codes (FAC) and Client Matter Codes
(CMC) and their configuration can be found in Chapter 17, "Configuring User Features, Part 2."
Restricting External Transfers

Call transfer is typically a feature that employees use to transfer from internal DNs to internal DNs.
When calls are transferred to an external destination, very often the reason is that the caller whose
call is transferred does not have permission to dial that external destination. The operator or an
employee, for instance, could be asked to transfer a call of a colleague who is home on vacation
to an international destination. Or an employee who is not allowed to call international numbers
could ask a colleague who is allowed to call international numbers to transfer the call to that
international number. An employee could also save money by having family members call the
employee in the office when they need to place costly calls. All that the employee has to do is to
transfer the call for them. To eliminate the cost of their call to the office of the employee, the
employee could even hang up and call them back before transferring the call.
You can configure Cisco CallManager to block external-to-external call transfers. This
configuration involves setting a simple service parameter and configuring gateways, trunks, and
route patterns as OffNet (external) devices. After you have configured this, external-to-external
call transfers will not be allowed. This feature provides an OnNet or OffNet alerting tone to the
terminating end of the call (determined by the configuration of the device as either OnNet or
OffNet). For incoming calls, trunks or gateways determine OffNet versus OnNet classification.
For outgoing calls, the route pattern determines OffNet versus OnNet status.
NOTE The external call transfer restriction requires the Cisco CallManager Release 4.1 or
later software component.
Defining OnNet and OffNet

Cisco CallManager classifies internal and external calls as OnNet and OffNet. A call coming from
an external PSTN is classified as an OffNet call. A call that is placed internally (from one
telephone to another, or between two Cisco CallManager clusters, where the call is routed over the
WAN) is classified as an OnNet call, as illustrated in Figure 22-5. When you are using Automated
Alternate Routing (AAR) in OnNet or OffNet implementations, it is important to know from
where the call is coming. With AAR, the source can be either the WAN connection or the PSTN
connection. This can cause restrictions when a call is rerouted. For example, if a user attempts to
call across the IP WAN and AAR reroutes the call through the PSTN, the called party will not be
able to transfer the call to another external phone number due to the Block OffNet to OffNet
Transfers service parameter.
Figure 22-5 OnNet Versus OffNet Calls
PSTN
V V
IP WAN
OnNet
IP IP IP IP IP IP
OffNet
You can configure gateways and trunks as OnNet (internal) or OffNet (external) by using gateway
configuration, using trunk configuration, or setting a clusterwide service parameter to classify
devices automatically. When the feature is used in conjunction with the clusterwide service
parameter Block OffNet to OffNet Transfer, the configuration determines whether calls can
transfer over a gateway or trunk.
These devices can be configured as OnNet and OffNet to Cisco CallManager, as shown in
Figure 22-6:
■ H.323 gateway
■ Media Gateway Control Protocol (MGCP) Foreign Exchange Office (FXO) trunk
■ MGCP T1/E1 trunk
■ Intercluster trunk
■ Session Initiation Protocol (SIP) trunk
Figure 22-6 Configuring a Gateway as OnNet or OffNet

By default, all gateways use OffNet classification.
To classify an outgoing call as OnNet or OffNet, you can set the Call Classification field in the
Route Pattern Configuration window to OnNet or OffNet, as shown in Figure 22-7. You can
override the route pattern setting and use the trunk or gateway setting by checking the Allow
Device Override check box in the Route Pattern Configuration window. All route patterns default
to OffNet if the Provide Outside Dial Tone check box is checked. If the check box is unchecked,
the route pattern defaults to OnNet.
Figure 22-7 Configuring a Route Pattern as OnNet or OffNet
Configuring Call Transfer Restrictions

Now that you have seen the method used to categorize gateways and route patterns, you can
complete these steps to block external calls from being transferred to external devices:
Step 1 You can specify the OnNet or OffNet classification on the following:
• Route patterns
• Intercluster trunks
• Gateways
Step 2 For incoming calls, configure individual gateways or trunks as OffNet.

Step 3 For outgoing calls, configure the route pattern Call Classification field as
OffNet. Note that you can choose the Allow Device Override check box
under the route pattern to allow the associated gateway to dictate whether
the outgoing call is OnNet or OffNet.
Step 4 Set the Block OffNet to OffNet Transfer clusterwide CallManager service
parameter shown in Figure 22-8 to True.
Figure 22-8 Setting the Block OffNet to OffNet Service Parameter
Call Transfer Restriction Example

In the Cisco CallManager configuration shown in Figure 22-9, the route pattern that is used to
reach the external destinations is classified as an OffNet pattern. The gateway is also classified as
OffNet. The service parameter Block OffNet to OffNet Transfer in the Cisco CallManager
service parameters is set to True. Thus, the attempt by party A to call party B and transfer the call
to party C will not work because OffNet-to-OffNet call transfers are restricted.
Figure 22-9 Blocking OffNet-to-OffNet Calls

Party C
1 Party A calls party B.
Phoenix
2 Party A presses the transfer
softkey and calls party C.
3 Party A cannot transfer
party B to party C because
party A calls go through the
OffNet route pattern.
2
PSTN
V
1 OffNet
3
IP
Party A Party B
Berlin Sydney
Dropping Conference Calls

As seen earlier, another method that a user can use for toll fraud is the ad hoc conferencing feature
of their IP phone. This is accomplished by dialing multiple OffNet individuals into a conference
call and then exiting from the call, leaving the OffNet individuals connected. Beginning in Cisco
CallManager Release 3.3(4), you can configure the conference call to drop when its creator leaves
the conference. Beginning in Cisco CallManager Release 4.1, you can also configure the value
When No OnNet Parties Remain in the Conference. Determine the OnNet status for each party
by checking the device or route pattern that the party is using, as with call transfer restrictions.
IP phones are always classified as OnNet devices. Gateways, trunks, and route patterns can be
classified either as OnNet or as OffNet.
Valid values for the Drop Ad Hoc Conference CallManager service parameter, shown in
Figure 22-10, are as follows:
■ Never—The conference call stays active even when the conference creator hangs up. This
behavior retains the original behavior of the conference feature. This is the default value.
■ When Conference Creator Drops Out—When the conference creator hangs up, the
conference call is dropped. When the conference creator transfers, redirects, or parks the call
and the retrieving party hangs up, the conference is also dropped.
Summary 517
■ When No OnNet Parties Remain in the Conference—When the last OnNet party in the
conference hangs up, the conference is dropped.
Figure 22-10 Configuring Drop Ad Hoc Conference Service Parameter
Summary
This chapter discussed combating voice network toll fraud using the Cisco CallManager. Sources
of toll fraud can be external or internal. The attacks typically focus on using the call forwarding
and conference features to avoid toll charges. You can stop call forwarding fraud through partitions
and calling search spaces. An additional method you can integrate to restrict call transfers is the
built-in CallManager ability to restrict OffNet-to-OffNet transfers.
You should also block commonly exploited area codes using route patterns. In addition, you can
use FAC and CMC to restrict or track the users making toll calls. Time-of-day routing is used to
change user permissions to place calls at certain hours or days. Finally, you can configure
CallManager to drop ad hoc conferences when no OnNet parties remain on the call or when the
conference creator drops out.
Review Questions
1. Which two of the following are typical sources of toll fraud?

a. Voice mail-to-voice mail transfer
b. Call Forward All
c. Transfer from voice mail
d. Transfer to an internal destination
e. Transfer to a conference system
2. When you are restricting external call transfers, which statement is correct?
a. Route patterns and gateways can be classified as OnNet or OffNet.
b. Calls must be classified to use call transfer restrictions.
c. Only restricted OnNet-to-OffNet transfers and OffNet-to-OffNet transfers can be used.
d. Only OnNet devices must be classified.
3. Which two service parameters can be configured when using ad hoc conference restrictions?
a. Never
b. When First OnNet Party Leaves the Conference
c. When No OffNet Parties Remain in the Conference
d. When Conference Creator Drops Out
4. What is the default value for the Drop Ad Hoc Conference service parameter?
a. Never
b. Drop Ad Hoc Conference When Creator Leaves
c. When No OnNet Parties Remain in the Conference
d. When No OffNet Parties Remain in the Conference
5. What is the purpose of FACs?
a. Time-based call routing
b. Restricting call routing to destinations based on the time of day
c. Restricting call routing to destinations based on user codes
d. Restricting call routing to destinations based on OnNet or OffNet classification
6. Why is it important to block commonly exploited area codes when you want to restrict
international calls?
a. Commonly exploited countries have special premium numbers as country codes.
b. Commonly exploited countries have special country codes that look like area codes of
the United States.
c. Countries such as the Bahamas can be reached either over the country code or over an
area code.
d. When these numbers are not blocked, they are treated like normal local telephone calls.
7. What automated method does Cisco support for security updates and hot fixes to the
CallManager server?
a. Receiving e-mail updates from the Cisco CallManager Notification Tool
b. Downloading updates directly from Microsoft using the Windows Update Services
c. Using the CSA standalone automatic update feature
d. Downloading updates directly from Cisco using the automated Cisco Update Services
8. Which of the following describes the configuration of time-of-day routing?
a. Create a time period, and then assign the time period to the calling search space.
b. Create time periods, add the time periods to a time schedule, assign the time schedule to
a partition, and assign the partition to a calling search space.
c. Create time schedules, add the time schedules to a time period, create partitions, assign
the partitions to a calling search space, and then assign the time period to the calling
search space.
d. Create a time period, add the time period to a time schedule, create partitions, assign the
partitions to a calling search space, and then assign the time schedule to the calling
search space.
9. Which of the following requires a code to be dialed, which is typically tracked to a specific
department or user, before making outgoing calls? (Hint: This code can be used to make all
outgoing calls.)
a. CBC
b. FAQ
c. FAC
d. CMC
10. Which of the following devices can be classified as OnNet or OffNet? (Choose three.)
a. Route pattern
b. Route group
c. Route list
d. Trunks
e. Gateways
f. IP phones
following topics:
■ Identifying potential threats against IP

phones and the attack tool or method
■ Explaining how signed firmware images
prevent rogue or incorrect images from being
placed on the IP phone
■ Configuring parameters in the Phone
Configuration window of Cisco CallManager
Administration to harden the IP phone
■ Explaining how disabling the PC port, the
Settings button, and web access help secure
the IP phone
■ Explaining how, by ignoring gratuitous ARP,
the IP phone can help prevent a man-in-the-
middle attack
■ Explaining how blocking the PC from
accessing the voice VLAN through the IP
phone prevents eavesdropping on the voice
conversation
■ Explaining how authentication and
encryption on Cisco CallManager and the IP
phones prevent identity theft of the phone or
Cisco CallManager server, data tampering,
and call-signaling and media-stream
tampering
CHAPTER 23
Hardening the IP Phone
The IP phone is a target for attacks just like all other components of the network. Very often
endpoints, such as IP phones, are not protected; only servers and network infrastructure devices
are hardened. This is not a good practice because IP phones have default settings that make them
vulnerable to certain attacks. However, several options are available to harden IP phones and, thus,
protect them against various attack and infiltration methods. This chapter discusses these methods.
Threats Targeting Endpoints

As shown in Figure 23-1, there are many attack paths against an IP phone, including a
connection through the network or through the integrated switch port to which a PC is attached.
Corrupt images and altered configuration files can sabotage the IP telephony environment.
Further attacks can be started from an infiltrated IP phone that is generally trusted and has access
to the network. The physical access to the IP phone can be misused for violations of the IP phone
integrity and the privacy of the user. Information can be gathered by browsing to the IP phone
as well. In addition, IP phone conversations are vulnerable to various attacks when the network
has been infiltrated, so the privacy of calls must be protected.
Figure 23-1 Attacks Against IP Phone Endpoints

Corrupt the image and Physical and web access to
the configuration file. network configuration settings.
Attack from network Connected PC sniffing the

listen to the communication. voice VLAN.
IP
Endpoints are a common target of attacks because they are usually less protected than strategic
devices, such as servers or network infrastructure devices. If an attacker gets control of an
endpoint, such as an IP phone, the attacker could use that device as a jumping-off point for
further attacks. Because the endpoints are trusted devices and have certain permissions in the
network, an attacker can use them to target devices that they would not be able to reach directly.
To get control of an IP phone, an attacker could try to modify the image and configuration file
(for example, by spoofing the TFTP server or by replacing the file on the TFTP server itself or
while in transit).
524 Chapter 23: Hardening the IP Phone
Another major threat is eavesdropping on conversations. If an attacker has physical access to the
IP phone, the attacker can “tap the wire,” either by connecting between the IP phone and the switch
or by connecting to the PC port of the IP phone. If the attacker does not have physical access to
the IP phone or its network connection, the attacker could launch a man-in-the-middle attack from
any network between two communicating endpoints. In a man-in-the-middle attack, the attacker
pretends to be a neighboring system (such as the default gateway when the communication is
between two IP networks or a peer on the same IP network) and, hence, receives all packets. A
common type of man-in-the-middle attack is to use gratuitous Address Resolution Protocol (ARP)
for redirection of packets at the MAC address layer.
A lot about the IP phone and the telephony infrastructure can be learned just by looking into the
network settings or browsing to the built-in HTTP server of the IP phone. This information
contains Dynamic Host Configuration Protocol (DHCP), Domain Name System (DNS), default
router, TFTP, and Cisco CallManager addresses. With this information, a hacker can direct an
attack at the TFTP or Cisco CallManager server, because Windows hosts are generally more
vulnerable than network components.
Overall, attacks on the endpoints can be broken down into four major categories:
■ Eavesdropping on VoIP conversations
■ Modifying the IP phone image
■ Attacking system and CallManager services
■ Hacking network devices and services
The simplest way to eavesdrop on the conversations of a user is to tap the wire between the IP
phone and the PC attached to it. A variety of tools exist to accomplish this feat:
■ Ettercap—A suite for man-in-the-middle attacks that allows sniffing and on-the-fly
manipulation of data
■ Voice Over Misconfigured Internet Telephones (VOMIT)—A tool that can create .wav
files from captured G.711 conversations
■ Ethereal—A sniffer and network protocol analyzer that allows both capturing conversations
and converting them to playable files
An attacker could also try to get control of an IP phone by modifying the IP phone image or
configuration file. This attack is carried out either at the TFTP server by manipulating the files
themselves or by replacing the content while it is in transit. For the first method, the attacker needs
access to the directory of the TFTP server; for the second, the attacker has to launch a successful
man-in-the-middle attack.
The hacker might want to direct the attack at the most critical telephony components: the servers.
An easy way to gather information about the IP addresses of critical components (such as the
Cisco CallManager addresses, default gateway address, TFTP server address, DNS server address,
and voice VLAN ID) is to retrieve them from the IP phone. This retrieval can be done locally at
an IP phone by using the Settings button or by connecting to the IP address of the IP phone with
a web browser. From the retrieved information, the hacker can build a topology map, associate it
with services, and use the topology map to attack relevant devices.
If the attacker manages to get access to network devices, such as routers and switches, the attacker
could redirect traffic to any destination using various kinds of tunnels. These include Generic
Route Encapsulation (GRE), IPsec, Layer 2 Protocol Tunneling (L2TP), or Switched Port
Analyzer (SPAN).
Blocking Endpoint Attacks

Cisco IP Phones are network devices that receive an IP address assignment. This allows them to
communicate on the network, but also opens them to attack from network intruders. The following
sections discuss mitigation techniques you can employ to protect the IP phone devices.
Stopping Rogue Images from Entering IP Phones

Cisco IP Phone image authentication was introduced with Cisco CallManager Release 3.3(3). In
this and later releases, phone images are signed by Cisco manufacturing. Such a signature proves
the authenticity of the origin, and the Cisco IP Phone will not accept images that are not published
by Cisco manufacturing. The image also contains information about the IP phone model, so an IP
phone will not accept an image for a different model. Since the introduction of image
authentication, the current image verifies the signature of a new image. Only if the new image is
properly signed by Cisco manufacturing is the new image accepted.
Cisco IP Phone configuration file authentication was introduced with Cisco CallManager Release
4.0 for Cisco IP Phone 7940, 7960, and 7970 models. The configuration files are signed by Cisco
CallManager. The phone now verifies the signature and accepts the new configuration file only if
it is properly signed by Cisco CallManager. Cisco IP Phone configuration file authentication
requires additional configuration and hardware (Universal Serial Bus [USB] security tokens) and
is not on by default.
IP phones will reject images with invalid signatures. They verify that the signature really validates
the corresponding file whether it was obtained from an attacker or from the legitimate TFTP
server. Since the introduction of this feature, the actual (current) image used in the phone includes
the code to accept new images only if they have a valid signature. The actual image also includes
the public key that is needed to verify the signature of new images. This way, only images that
have been signed with the correct private key (owned by Cisco engineering) can be loaded. This
also means that after you load the first image that features signature verification, you cannot load
an older image that does not include a valid signature. This limitation guarantees that no image
that has been tampered with can be loaded to your phone.
With image signature information about the IP phone model having been added to the image as
well, the IP phone can verify that the image that is loaded is not for a different phone model.
Before the introduction of this feature, a phone became inoperable if an image for another model
was loaded, whereas today, such a wrong image is simply not accepted.
Disabling Phone Settings in Cisco CallManager Administration

The product-specific configuration parameters of Cisco IP Phones are set by default to achieve the
greatest functionality but are considered insecure. To secure Cisco IP Phones, these settings can
be modified:
■ Disable Speakerphone and Disable Speakerphone and Headset—Disable these features to

prevent eavesdropping on conversations in the office.
■ PC Port—Disable the PC port to prevent a PC from connecting to the network via the IP
phone switch.
■ Settings Access—Disable or restrict access to the IP phone settings to avoid the risk that
details about the network infrastructure could be exposed.
■ Gratuitous ARP—Disable this feature to prevent gratuitous ARP-based man-in-the-middle

attacks.
■ PC Voice VLAN Access—Disable this feature to stop the IP phone from forwarding voice
VLAN traffic to the PC.
■ Web Access—Disable access to the IP phone from a web browser to avoid the risk that details
about the network infrastructure could be exposed.
Disabling PC Port and Settings Access

The PC port is typically disabled in special areas such as a lobby or areas where no additional PC
access is allowed. It is not a common setting, though, because it entails a major functionality
constraint. Disabling the settings access prevents users from gathering information about, for
example, DHCP server, TFTP server, default router, and Cisco CallManager IP addresses. Cisco
CallManager Release 4.1 and later releases offer the Restricted option for settings access. With
restricted access, the user can modify the contrast and ringer settings but cannot see any other
information.
You can access both of these settings through the IP Phone Configuration window in Cisco
CallManager Administration, as shown in Figure 23-2.
Figure 23-2 Modifying IP Phone Security Settings
Disabling IP Phone Web Access

You can use a web browser to connect to the HTTP server of the IP phone by simply browsing to
the IP address of the phone, as shown in Figure 23-3. The HTTP server displays similar
information that can be viewed directly on the IP phone using the Settings button, enhanced by
some additional statistics. A hacker can use the intelligence gained by discovering the network
configuration to direct attacks at the most critical telephony components, such as Cisco
CallManager and the TFTP server. Therefore, from a security perspective, it is recommended that
you disable web access to the phone. When web access is disabled, the IP phone will not accept
incoming web connections and, hence, does not provide access to sensitive information.
NOTE Disabling web access at the IP phone stops Extensible Markup Language (XML) push
applications from working. If you want to use XML push applications on some IP phones (for
instance, for an emergency notification application), you cannot disable web access to the IP
phone.
Figure 23-3 Cisco IP Phone Built-in Web Server
Ignoring Gratuitous ARP

Usually, ARP operates in a request-and-response fashion. When a station needs to know the MAC
address of a given IP address, it sends an ARP request. The device with the corresponding IP
address replies and, thus, provides its MAC address. All receiving devices update their ARP cache
by adding the IP and MAC address pair. Gratuitous ARP packets are packets that announce the
MAC address of the sender even though this information has not been requested. This technique
allows receiving devices to update their ARP caches with the information. Usually, such gratuitous
ARP messages are sent after the MAC address of a device has changed to avoid packets being sent
to the old MAC address until the related entry has timed out in the ARP caches of the other devices.
Gratuitous ARP, however, can also be used by an attacker to redirect packets in a man-in-the-
middle attack.
Figure 23-4 illustrates a gratuitous ARP attack against an IP phone. By default, Cisco IP Phones
accept gratuitous ARP messages and update their ARP cache whenever they receive a gratuitous
ARP packet.
Figure 23-4 Manipulating Gratuitous ARP to Accomplish a Man-in-the-Middle Attack

10.10.10.1 Default Gateway
Communication
Gratuitous ARP IP
claims to be
10.10.10.1
PC of the
Hacker
The attacker located in the VLAN of the IP phone repeatedly sends out gratuitous ARP packets
announcing its MAC address to be the MAC address of the default gateway of the IP phone. The
IP phone accepts the information, updates its ARP cache, and forwards all packets meant for the
default gateway to the attacker. With tools such as Ettercap, the hacker can copy or modify the
information and then relay it to the real default gateway. The user does not notice that someone is
listening to the data stream as long as the hacker does not significantly increase the delay and does
not drop packets.
In this example, only traffic from the IP phone toward the default gateway is sent to the attacker,
but if the attacker also impersonates the IP phone toward the router, the attacker could control
bidirectional traffic. In this case, the router would also have to listen to gratuitous ARP packets.
NOTE There are several methods to prevent gratuitous ARP attacks in the network. You can
disable it on end devices or you can use features such as Dynamic ARP Inspection (DAI) and IP
Source Guard on Catalyst switches. You can find more information about DAI and IP Source
Guard in your Cisco IOS or Cisco Catalyst operating system switch configuration guide.
To prevent gratuitous ARP-based attacks against an IP phone, the gratuitous ARP feature of the IP
phone should be disabled. This setting is found at the bottom of the IP Phone Configuration
window, as shown in Figure 23-2.
Disabling Voice VLAN Access

By default, an IP phone sends all traffic that it receives from the switch out its PC port. This
enables the PC to see not only the traffic of the native VLAN, the data VLAN, but also to see the
traffic of the voice VLAN. When the PC receives voice VLAN traffic, the traffic can be captured
and, hence, the conversation can be sniffed.
Further, the PC can also send packets to the voice VLAN if they are tagged accordingly. This
breaks the separation of voice VLANs and data VLANs, because the PC that is supposed to have
access to the data VLAN only is now able to send packets to the voice VLAN, bypassing all
access-control rules (access control lists [ACLs] in routers or firewalls) that might be enforced
between the two VLANs.
Usually, the PC does not need access to the voice VLAN, and, therefore, you should block PC
access to the voice VLAN.
NOTE Some applications, such as call recording or supervisory monitoring in call centers,
require access to the voice VLAN. In such situations, you should not disable the PC Voice VLAN
Access setting.
To block the PC from accessing the voice VLAN, set the PC Voice VLAN Access configuration
parameter shown in Figure 23-2 to Disabled. When a phone is configured this way, it will not
forward voice VLAN-tagged traffic to the PC when it receives such frames from the switch. In
addition, the phone will not forward voice VLAN-tagged traffic to the switch if it receives such
frames from the PC. Although this setting is recommended from a security perspective, it makes
troubleshooting more difficult because you cannot analyze voice VLAN traffic from a PC
connected to the PC port of the IP phone. Whenever you need to capture voice VLAN traffic to
analyze network problems, you will have to sniff the traffic on the network devices. On Cisco
Catalyst switches, you can configure SPAN ports to duplicate traffic from certain ports or VLANs
to another port where you attach the PC that is running your protocol analyzer. Remote SPAN
(RSPAN) even allows you to send the selected traffic to another switch for remote analysis.
NOTE Cisco 7912 IP Phones do not currently support disabling PC voice VLAN access.
Enabling IP Phone Encryption and Authentication

Cisco CallManager Release 4.0 introduced certificate-based authentication and encryption of
signaling and media. Skinny Call Control Protocol (SCCP), used between the IP phone and Cisco
CallManager for call signaling, can be secured by Transport Layer Security (TLS), formerly
known as Secure Sockets Layer (SSL). This protects the signaling channels from most types of
attacks. The media stream between two IP phones is protected with Secure Real-Time Transfer
Protocol (SRTP), which encrypts and authenticates the voice data. A hacker cannot modify the
packets and is not able to listen to the audio stream because it is now encrypted.
NOTE Because of its complexity, IP phone encryption and authentication is discussed fully in
Chapters 24 through 27.
Summary
This chapter discussed the potential threats against Cisco IP Phones and the attack tools or
methods a hacker can use. Hackers typically begin with the weakest points in the network, such
as Cisco IP Phones. To combat this, the IP phones can validate images and configuration updates
using signature security. Cisco administrators can disable access to the Settings button and built-
in web server to prevent hackers from viewing key network information. In addition, disabling
gratuitous ARP prevents man-in-the-middle attacks.
Finally, it is a good practice to disable the PC port of an IP phone if no PC is attached to it, and
generally block access to the voice VLAN to avoid unauthorized network access.
Review Questions
1. What could be of interest to a hacker planning to attack an IP phone?

a. The attacker can learn about the IP telephony environment.
b. The attacker can start attacks from the IP phone.
c. With a modified image and configuration file, the attacker can bring down the Cisco
CallManager.
d. The attacker can sabotage a special user.
2. Which IP phone does not support configuration file authentication?
d. Cisco IP Phone 7970
3. In which window are IP phone security settings configured?
a. Directory Number Configuration
b. Phone Configuration
c. Phone Security Configuration
d. Product Specific Configuration
4. How do you browse to the built-in web server of an IP phone?
a. http://IP-Phone’s-IP-address
b. https://IP-Phone’s-IP-address
c. https://IP-Phone’s-IP-address/CCMAdmin
d. https://IP-Phone’s-IP-address/Admin
5. Which statement is not true about gratuitous ARP attacks?

a. Gratuitous ARP is a man-in-the-middle attack.
b. Gratuitous ARP attackers usually operate from the Internet.
c. Gratuitous ARP is normally used for HSRP.
d. Ettercap is a tool used for gratuitous ARP attacks.
6. Which of the following statements about authentication and encryption is not true?
a. It was introduced with Cisco CallManager Release 4.0.
b. Media streams use SRTP.
c. Signaling uses Secure SCCP.
d. TLS was formerly known as SSL.
7. Which of the following network information cannot be found out from a Cisco IP Phone?
a. DHCP server address
b. DNS server address
c. TFTP server address
d. Intranet server address
e. Cisco CallManager address
8. You want to prevent users from accessing the PC port of a 7912 IP Phone. What option is
available to you?
a. Use the Cisco CallManager Phone Configuration window to disable the PC port.
b. Use the CallManager service parameters to disable all PC ports.
c. Use the Bulk Administration Tool to disable the PC port for all 7912 IP Phones.
d. Fill the PC port of the phone with glue.
9. Which of the following was the predecessor of Transport Layer Security?
a. IPsec
b. SSL
c. DES
d. AES
10. What must you do to implement signed firmware validation on the Cisco IP Phones?
a. Nothing; the feature is already enabled since CCM 3.3(3).
b. Change the signed firmware setting from the Phone Configuration window.
c. Change the signed firmware setting from the CallManager service parameters.
d. Change the signed firmware setting from the IP phone itself.
following topics:
■ Defining cryptography and explaining the

four cryptographic services: confidentiality,
integrity, authentication, and nonrepudiation
■ Explaining the basic operation and uses of
symmetric encryption algorithms and
identifying the common algorithms in use
today
■ Explaining the basic operation and uses of
asymmetric encryption algorithms and
identifying the common algorithms in use
today
■ Explaining how hash functions provide data
integrity and listing common hash functions
in use today
■ Defining digital signatures and explaining
how they provide identity and integrity
CHAPTER 24
Understanding Cryptographic
Fundamentals
Configuring Cisco CallManager for security is relatively straightforward; however, the

underlying security services, algorithms, and operations are often not well-known to the Cisco
CallManager administrators who must secure the Cisco CallManager installation. This chapter
provides information about cryptographic fundamentals. It helps you understand the elements
that are the basis of cryptography in the data world.
What Is Cryptography?
Cryptography is the science of transforming readable messages into an unintelligible form and
the later reversal of that process. The application is to send the transformed, unreadable message
over an untrusted channel. In the data world, this untrusted channel very often is a public
network, such as the Internet.
Cryptography provides four services:
■ Data authenticity—This service should guarantee that the message comes from the source
that it claims to come from. When an application such as e-mail or protocols such as IP do
not have any built-in mechanisms that prevent spoofing of the source, cryptographic
methods can be used for proof of sources.
■ Data confidentiality—This service provides privacy by ensuring that messages can be read
only by the receiver.
■ Data integrity—This service ensures that the messages are not altered in transit. With data
integrity, the receiver can verify that the received message is identical to the sent message
and that no manipulation was done.
■ Data nonrepudiation—This service allows the sender of a message to be uniquely

identified. With nonrepudiation services in place, a sender cannot deny having been the
source of that message.
All these services are based on encryption and authentication methods. However, for different
applications, different kinds of encryption and authentication techniques are used. Figure 24-1
illustrates examples of the four services.
536 Chapter 24: Understanding Cryptographic Fundamentals
Figure 24-1 Services of Cryptography
Authenticity Was it really

Integrity Was it modified
sent by A? by others?
Love Love Hate

Letter Letter Letter
A B A B
C C
Confidentiality Was it read

Nonrepudiation Can it be proven
by others? that A sent it even
if A denies that?
Love Hate
Letter Letter
A B A B
C C
These scenarios are possible:
■ Authenticity—If B receives a love letter that says it is coming from A, how can B be sure that
it was really sent by A and not someone else? Without any reliable service that ensures
authenticity of the source, user B will never know.
■ Confidentiality—On the other hand, if there are means of guaranteeing the authenticity of
the source, B might be afraid that somebody else read the love letter while it was in transit,
resulting in a loss of privacy. This problem could be solved by a service providing
confidentiality.
■ Integrity—If B were to receive a hate letter, formed in a way that it proved the authenticity
of the source, how can B know that the content has not been modified in transit? A service
that ensures integrity of the message is needed to eliminate this kind of threat.
■ Nonrepudiation—However, if B receives a hate letter from A that seems to be authentic, can

B prove to others that it must have been sent by A? A nonrepudiation service is needed in this
case.
What Is Cryptography? 537
It might appear that the authenticity service and the nonrepudiation service are fulfilling the same
function. Although both address the question of the proven identity of the sender, there is a small
difference in the two, which is sometimes quite important: When the receiver needs to be sure
about the authenticity of the source, the method and the means that are used to achieve the proof
of authenticity can be available to both the sender and the receiver. Because the receiver knows
that he or she was not the source, it does not matter that the sender and receiver both know how to
treat a message to provide authenticity of the source.
If, however, the receiver has to prove the source of the sender to others, it is not acceptable that the
receiver know how the sender treated this message to prove authenticity because the receiver could
then have pretended to be the sender.
An example for authenticity versus nonrepudiation is data exchange between two computers of
the same company versus data exchange between a customer and a web shop. When the two
computers do not have to prove to others which of them sent a message, but just need to make sure
that whatever was received by one was sent by the other, the two computers can share the same
way of transforming their messages. This practice is not acceptable in business applications such
as a web shop. If the web shop knows how a customer transforms messages to prove authenticity
of the source, the web shop could easily fake “authentic” orders. Therefore, in such a scenario, the
sender must be the only party having the knowledge how to transform messages. Then, the web
shop can prove to others that the order must have been sent by the customer. The customer could
not argue that the order was faked by the web shop when the web shop does not know how to
transform the messages from the customer to make them authentic.
Authentication and Encryption

Authentication functions are used to provide authenticity, integrity, and nonrepudiation. To
achieve this, the sender adds (appends) verification data to the actual data. The authenticated data
can be information about the sender (such as its identity) or the information that should be passed
from the sender to the receiver itself. The receiver checks the verification data added by the sender
and, if successful, can confirm authenticity.
There are various ways to create the verification data, the most common being Hash-based
Message Authentication Code (HMAC) or digital signatures.
Confidentiality functions are provided by encryption. More precisely, the transformation of

cleartext to ciphertext is called encryption, whereas the transformation of the ciphertext back to
the original cleartext is called decryption.
Encryption utilizes an encryption algorithm and keys. If the key that is used to encrypt the data
and the key that is used to decrypt the data is the same, the encryption algorithm is considered
symmetric (with symmetric keys). If the encryption and decryption keys are different, the
encryption algorithm is asymmetric (with asymmetric keys).
Although the encryption algorithms are usually well-known, the keys that are used for the
encryption have to be secret. Symmetric keys have to be known by both endpoints that want to use
a symmetric encryption algorithm for their data exchange. With asymmetric encryption, the
sender needs to know only the encryption key, whereas the receiver needs to know only the
decryption key.
Desirable features of an encryption algorithm are as follows:
■ Resistance to cryptographic attacks—The algorithm itself must be trusted by the

cryptographic community and there must be no shortcut to decipher data other than knowing
or guessing the decryption key.
■ Variable key lengths and scalability—The longer the encryption key, the longer it will take
attackers to break it if they try all the possible keys (for example, a 16-bit key = 216 = 65,536
possible keys, whereas a 56-bit key = 256 = 71,892,000,000,000,000 possible keys).
Scalability provides flexible key length, and the strength or speed of encryption can be
selected as needed.
■ Avalanche effect—When only a small part of the plaintext message is changed (a few bits),
and that small change causes its ciphertext to change completely, the algorithm has an
avalanche effect. The avalanche effect is a desired feature because it allows very similar
messages to be sent over an untrusted medium, with their encrypted (ciphertext) messages
being completely different.
Symmetric Encryption
Symmetric encryption has two main characteristics: It is very fast (compared to asymmetric
encryption) and uses the same key for encryption and decryption, as shown in Figure 24-2. As a
consequence, the same key has to be known by the sender and the receiver. To ensure
confidentiality, nobody else is allowed to know the key. Such keys are also called shared secrets.
Figure 24-2 Symmetric (Shared Secret) Encryption

Encryption/Decryption Encryption/Decryption
Key Key
Shared Secret Key
Encrypt Decrypt
$1000 $!@#IQ $1000
Symmetric Encryption 539
Symmetric encryption has been used for decades, and several algorithms are commonly used.
Among the best-known and most widely trusted symmetric encryption algorithms are Triple Data
Encryption Standard (3DES), Advanced Encryption Standard (AES), International Data
Encryption Algorithm (IDEA), the RC series (RC2, RC4, RC5, RC6), Software Encryption
Algorithm (SEAL), and Blowfish. They are all based on the same concept: They have two types
of input (the cleartext and the key) and produce unreadable output (the ciphertext). For decryption,
the ciphertext and the key are the input data and the original cleartext is the output.
Symmetric algorithms are usually very simple in their structure, therefore quite fast, and as a
consequence, they are often used for wire-speed real-time encryption in data networks. They are,
in their essence, based on simple mathematical operations and can be easily hardware-accelerated
using specialized encryption application-specific integrated circuits (ASICs). Typical applications
are e-mail, IPsec, Secure Real-Time Transfer Protocol (SRTP), or Secure HTTP (HTTPS).
Keys should be changed frequently because they could be discovered otherwise, and loss of
privacy would be the consequence. The “safe” lifetime of keys depends on the algorithm, the
volume of data for which they are used, the key length, and the time period for which the keys are
used. The key length is usually 128 to 256 bits. Because of the limited lifetime (usually hours to
days) and the fact that each pair of devices should use a different key, key management is rather
difficult.
Symmetric Encryption Example: AES

For a number of years, the industry recognized that Data Encryption Standard (DES) would
eventually reach the end of its useful life. In 1997, the AES initiative was announced, and the
public was invited to propose encryption schemes, one of which could be chosen as the encryption
standard to replace DES.
On October 2, 2000, the U.S. National Institute of Standards and Technology (NIST) announced
the selection of the Rijndael cipher as the AES algorithm. The Rijndael cipher, developed by Joan
Daemen and Vincent Rijmen, has a variable block length and key length. The algorithm currently
specifies how to use keys with lengths of 128, 192, or 256 bits to encrypt blocks with lengths of
128, 192, or 256 bits (all nine combinations of key length and block length are possible). Both
block length and key length can be extended very easily to multiples of 32 bits, allowing the
algorithm to scale with security requirements of the future. The U.S. Department of Commerce
approved the adoption of AES as an official U.S. government standard, effective May 26, 2002.
AES was chosen to replace DES and 3DES because they are either too weak (DES, in terms of
key length) or too slow (3DES) to run on modern, efficient hardware. AES is, therefore, more
efficient on the same hardware (much faster, usually by a factor of around five compared to 3DES),
and is more suitable for high-throughput, low-latency environments, especially if pure software
encryption is used. However, AES is a relatively young algorithm, and, as the golden rule of
cryptography states, a more mature algorithm is always more trusted. 3DES is, therefore, a more
conservative and more trusted choice in terms of strength, because it has been analyzed for around
30 years. AES has also been thoroughly analyzed during the selection process, and is considered
mature enough for most applications.
AES is the algorithm for encrypting both IP phone-to-Cisco CallManager communication

(signaling with Transport Layer Security [TLS] protection) and phone-to-phone and phone-to-
gateway (media with SRTP protection) channels in Cisco IP telephony.
Asymmetric Encryption
Asymmetric algorithms (also sometimes called public-key algorithms) are designed in such a way
that the key used for encryption is different from the key used for decryption, as shown in Figure
24-3. The decryption key cannot (at least in any reasonable amount of time) be calculated from
the encryption key and vice versa.
Figure 24-3 Asymmetric (Public Key) Encryption

Encryption Decryption
Key Key
Encrypt Decrypt
$1000 %3f7&4 $1000
The main feature of asymmetric encryption algorithms is that the encryption key (often called the
public key) does not have to be secret; it can be published freely and anyone can use this key to
encrypt data. The corresponding decryption key (often called the private key) is known only to a
single entity that can decrypt data encrypted with the encryption key. Therefore, when you need
to send an encrypted message to someone else, you first obtain the public (encryption) key of the
other person and transform the message with it. Only the recipient knows the private (decryption)
key and can, therefore, decrypt the message.
Asymmetric algorithms are relatively slow (up to 1000 times slower than symmetric algorithms).
Their design is based on computational problems, such as factoring extremely large numbers or
computing discrete logarithms of extremely large numbers.
The best-known asymmetric cryptographic algorithms are the Rivest, Shamir, and Adleman
(RSA); ElGamal; and elliptic curve algorithms. RSA is recommended because it is widely trusted
for its resistance against attacks and well-known internals. Because of their lack of speed,
Hash Functions 541
asymmetric encryption algorithms are usually used to protect small quantities of data (such as
digital signatures or key exchange). Key exchange allows you to use the slower, more secure
asymmetric algorithm to protect the exchange of a faster symmetric key algorithm over a public
network, such as the Internet.
Key management tends to be simpler compared to symmetric (secret key) algorithms. As stated
earlier, with asymmetric encryption, each device has a pair of keys (public and private). The public
key of each device has to be publicly available (known by all other devices) to allow a full mesh
of encrypted communication, whereas with symmetric encryption different symmetric keys have
to be safely distributed for each combination of two peers. Asymmetric keys are usually used for
a longer time (months to years).
Symmetric Encryption Example: RSA

Ronald L. Rivest, Adi Shamir, and Leonard M Adleman invented the RSA algorithm in 1977. It
was a patented public-key algorithm, and its patent expired in September 2000, putting the
algorithm in the public domain. Of all the public-key algorithms proposed over the years, RSA is
still the most strongly preferred.
RSA has withstood years of extensive cryptoanalysis, and although analysis has neither proven
nor disproven the security of the RSA algorithm, it does suggest a justifiable confidence. The
security of RSA is based on the difficulty of factoring very large numbers, that is, breaking them
into multiplicative factors. If an easy method of factoring these large numbers were discovered,
the effectiveness of RSA would be destroyed (and, as a side effect, mathematics might take a huge
leap). RSA keys are usually 1024 to 2048 bits long.
RSA, like all asymmetric encryption algorithms, can be used in two different ways:
■ Confidentiality—The sender encrypts the data with the public key of the receiver. This
guarantees that only the receiver can decrypt the data.
■ Authenticity of digital signatures—The sender uses its private key to sign (encrypt) the data.
Such a signature can be verified by everybody because only the public key is needed to verify
(decrypt) the signature.
RSA is used for device authentication (IP phone to Cisco CallManager and vice versa) in Cisco
IP telephony.
Hash Functions
Hash functions are used for several cryptographic applications. They can be used for secure
password verification or storage and are also a base component for data authentication.
Hashing is a one-way function of input data, which produces fixed-length output data, the digest.
The digest uniquely identifies the input data and is cryptographically very strong, that is, it is
impossible to recover input data from its digest, and if the input data changes just a little, the digest
(fingerprint) changes substantially (avalanche effect). Therefore, high-volume data can be
identified by its (shorter) digest. For this reason, the digest is called a fingerprint of the data. Given
only a digest, it is not computationally feasible to regenerate the data that was used to compute the
digest.
Figure 24-4 illustrates how hashing is performed. Data of arbitrary length is input to the hash
function, and the result of the hash function is the fixed-length hash (digest, fingerprint). Hashing
is similar to the calculation of cyclic redundancy check (CRC) checksums, except that it is much
stronger from a cryptographic point of view. With CRC, given a CRC value, it is easy to generate
data with the same CRC. However, with hash functions, this is not computationally feasible for an
attacker.
Figure 24-4 Hashing Process

Message
Data of Arbitrary
Length
Hash
Function
Fixed-Length e883aa0b24c09…
Hash
The two best-known hashing functions are these:
■ Message Digest 5 (MD5), with 128-bit digests
■ Secure Hash Algorithm 1 (SHA-1), with 160-bit digests
There is considerable evidence that MD5 might not be as strong as originally envisioned and that
collisions (different inputs resulting in the same fingerprint) are more likely to occur than designed
for. Therefore, MD5 should be avoided as an algorithm of choice and SHA-1 should be used
instead.
NIST developed SHA, the algorithm specified in the Secure Hash Standard. SHA-1 is a revision
to SHA that was published in 1994; the revision corrected an unpublished flaw in SHA. Its design
is very similar to the MD4 family of hash functions developed by Rivest. The algorithm takes a
message of no less than 264 bits in length and produces a 160-bit message digest. The algorithm
Hash Functions 543
is slightly slower than MD5, but the larger message digest makes it more secure against brute-
force collision and inversion attacks.
Figure 24-5 illustrates hashing in action. The sender wants to ensure that the message will not be
altered on its way to the receiver. The sender uses the message as the input to a hashing algorithm
and computes its fixed-length digest or fingerprint. This fingerprint is then attached to the message
(the message and the hash are cleartext) and sent to the receiver. The receiver removes the
fingerprint from the message and uses the message as input to the same hashing algorithm. If the
hash computed by the receiver is equal to the one attached to the message, the message has not
been altered during transit.
Figure 24-5 Hashing Example

1 Sending PC
generates data
Confirm
Order
3 Hash added
to packet
Confirm Hashing 4 Destination

PC compares
Order Algorithm hash
2 Sender Hashing e8F0s31a …
hashes
data Algorithm
e8F0s31a … e8F0s31a …
Hash Digest Same Hash
Digest?
Be aware that there is no security added to the message in this example. When the message
traverses the network, a potential attacker could intercept the message, change it, recalculate the
hash, and append the newly recalculated fingerprint to the message (a man-in-the-middle
interception attack). Hashing only prevents the message from being changed accidentally (that is,
by a communication error). There is nothing unique to the sender in the hashing procedure;
therefore, anyone can compute a hash for any data, as long as they know the correct hash
algorithm.
Thus, hash functions are helpful to ensure that data was not changed accidentally but cannot
ensure that data was not deliberately changed. For the latter, you need to employ hash functions
in the context of Hash-based Message Authentication Code (HMAC). They will extend hashes by
adding a secure component.
HMAC uses existing hash functions, but with the significant difference of adding an additional
secret key as the input to the hash function when calculating the digest (fingerprint). Only the
sender and the receiver share the secret key, and the output of the hash function now depends on
the input data and the secret key. Therefore, only parties who have access to that secret key can
compute or verify the digest of an HMAC function. This defeats man-in-the-middle attacks and
also provides authentication of data origin. If only two parties share a secret HMAC key and use
HMAC functions for authentication, the receiver of a properly constructed HMAC digest with a
message can be sure that the other party was the originator of the message because that other party
is the only other entity possessing the secret key. However, because both parties know the key,
HMAC does not provide nonrepudiation. For the latter, every entity would need its own secret key
instead of having a secret key shared between two parties.
HMAC functions are generally fast and are often applied in these situations:
■ To provide a fast proof of message authenticity and integrity among parties sharing the secret
key, such as with IPsec packets or routing protocol authentication
■ To generate one-time (and one-way) responses to challenges in authentication protocols (such

as PPP Challenge Handshake Authentication Protocol [CHAP], Microsoft NT Domain, and
Extensible Authentication Protocol-MD5 [EAP-MD5])
■ To provide proof of integrity of bulk data, such as with file-integrity checkers (for example,
Tripwire), or with document signing (digitally signed contracts, Public Key Infrastructure
[PKI] certificates)
Some well-known HMAC functions are as follows:
■ Keyed MD5, based on the MD5 hashing algorithm, which should be avoided
■ Keyed SHA-1, based on the SHA-1 hashing algorithm, which is recommended
Cisco IP telephony uses SHA-1 HMAC for protecting signaling traffic and media exchange.
Digital Signatures
Digital signatures are verification data appended to the data that is to be signed. They provide three
basic security services in secure communications:
■ Authenticity of digitally signed data—Authentication of source, proving that a certain party

has signed the data in question.
■ Integrity of digitally signed data—Guarantee that the data has not changed since being
signed by the signer.
■ Nonrepudiation of the transaction—The recipient can take the data to a third party, which
will accept the digital signature as a proof that this data exchange really did take place. The
signing party cannot repudiate (that is, deny) that it has signed the data.
Digital Signatures 545
Digital signatures are usually based on asymmetric encryption algorithms to generate and verify
digital signatures. Compared to using asymmetric encryption for confidentiality, the usage of the
keys is reversed when creating digital signatures: The private key is used to create the signature,
and the public key is used to verify the signature.
Because digital signatures are based on asymmetric (slow) algorithms, they are not used today to
provide real-time authenticity and integrity guarantees to network traffic. In network protocols,
they are usually used as a proof of endpoint (client, server, and phone) identity when two entities
initially connect (for example, an IP phone authenticating to Cisco CallManager, or a Cisco VPN
Client authenticating to a Cisco VPN Concentrator). For real-time protection of authenticity and
integrity, which do not require nonrepudiation (for example, signaling messages between IP
phones and a Cisco CallManager, or IPsec packet protection), HMAC methods are used instead.
Digital Signatures and RSA

Digital signatures require a key pair for each device that wants to create signatures. One key is
used to create signatures and the other is used to verify signatures. RSA can be used for that
purpose. The usage of the RSA keys for digital signatures is opposite to their usage for encryption:
■ Digital signatures—The signer uses its private key to sign (encrypt) data. The signature is
checked by a recipient who is using the public key of the signer to verify (decrypt) the
signature.
■ Encryption—The sender encrypts the data with the public key of the receiver. This
guarantees that only the receiver can decrypt the data because the encrypted data can only be
decrypted by the holder of the private key.
RSA is extremely slow and not designed for real-time encryption of a large volume of data.
Therefore, when it is used to create signatures, the data that is to be signed is first hashed, and only
the hash digest is signed by RSA (encrypted with the public key). This practice significantly
improves performance because RSA transforms only the fingerprint of the data (not all of the
data). The signature and verification process, illustrated in Figure 24-6, is as follows:
Step 1 The signer makes a hash (fingerprint) of the document, which uniquely
identifies the document and all its contents.
NOTE A hash of the data is created for two reasons: First, RSA is extremely slow, and it is
more efficient to sign only the (shorter) fingerprint than to sign the whole of the data. Next, if
the transferred information should be out-of-band verified, it is simpler to compare the shorter
fingerprint than to compare all the transferred information.
Step 2 The signer encrypts the hash only with its private key.
Step 3 The encrypted hash (the signature) is appended to the document.
Step 4 The verifier obtains the public key of the signer.
Step 5 The verifier decrypts the signature with the public key of the signer. This
process unveils the assumed hash value of the signer.
Step 6 The verifier makes a hash of the received document (without its signature)
and compares this hash to the decrypted signature hash. If the hashes
match, the document is authentic (that is, it has been signed by the assumed
signer) and has not been changed since the signer signed it.
Figure 24-6 RSA Signature Process
Purchase Order
$100,000
1 SHA-1 Hash
Untrusted Network
49eD0e3A7c44 …
Purchase Order Purchase Order
$100,000 $100,000
Signature
2 RSA Encrypt e10d6200aCe …
3
5
4 RSA Decrypt SHA-1 Hash
Private Key Public Key
of User A of User A 49eD0e3A7c44 …
Same Hash Digest?
6
Summary
This chapter discussed the concepts behind cryptography. Cryptography is the science of
transforming cleartext into ciphertext and transforming the ciphertext back into cleartext.
Symmetric encryption is one of the fastest encryption methods because it uses the same key for
both encryption and decryption. With symmetric encryption, a different key is needed for each pair
of devices. Asymmetric encryption uses a different key for encryption and decryption. When using
asymmetric encryption, each device needs a pair of keys: one public and one private.
Hashes are one-way functions that can be used to authenticate data by adding a secret value to the
data being sent. Digital signatures sign data by using the private, asymmetric encryption, key to
encrypt fingerprints (hashes) of the data.
Review Questions
1. Which two of the following are not cryptographic services?

a. Authenticity
b. Confidentiality
c. Integrity
d. Nonrepudiation
e. Resistance against DoS
f. Defense in depth
2. Which two statements about symmetric encryption are true?
a. Symmetric encryption is a good choice for real-time encryption of bulk data.
b. Symmetric encryption is commonly used to sign asymmetric keys.
c. Symmetric encryption uses asymmetric keys.
d. RSA is an example of a symmetric encryption algorithm.
e. ASE is an example of a symmetric encryption algorithm.
f. With symmetric encryption, the encryption key equals the decryption key.
3. Which two statements about asymmetric encryption are true?
a. Asymmetric encryption is considerably faster than symmetric encryption.
b. Asymmetric encryption keys should have about half the lifetime of symmetric encryp-
tion keys.
c. With asymmetric encryption, the private key can only encrypt data, whereas the public
key can only decrypt data.
d. With asymmetric encryption, either of the two keys can be used for encryption and the
opposite key can be used for decryption.
e. NSA is an example of an asymmetric encryption algorithm.
f. Asymmetric encryption is often used to create signatures.
4. Which two statements about hash functions are true?

a. A hash digest can never be reverted to the hashed data.
b. It is computationally difficult to revert a hash digest to the hashed data.
c. Data can be encrypted if hashed with a secret key.
d. Data can be signed by appending a hash of the data.
e. AES can use SHA-1 for data encryption.
f. AES can use MD5 for hashing data.
5. Which of the following statements does not apply to digital signatures?
a. Digital signatures provide data authenticity.
b. Digital signatures provide data integrity.
c. Digital signatures provide nonrepudiation.
d. Digital signatures do not provide data confidentiality.
e. Digital signatures are based on asymmetric cryptographic algorithms.
f. Digital signatures are created by hashing the result of an asymmetric encryption.
6. Which of the following represent a more reliable hashing algorithm?
a. DES
b. 3DES
c. AES
d. SHA-1
e. RSA
f. MD5
7. You are looking for a method that can prove to others that a certain source sent some data.
What cryptography method accomplishes this?
a. Nonrepudiation
b. Authentication
c. Confidentiality
d. Integrity
e. Encryption
8. Which of the following is not a symmetric encryption algorithm?

a. AES
b. RSA
c. DES
d. Blowfish
9. When using asymmetric encryption, which key is transmitted from the sending to the
receiving host?
a. Public key
b. Private key
c. Shared secret key
d. No keys are transmitted
10. When using asymmetric encryption, the public key is a reverse encryption algorithm of the
private key. What is to keep a host from capturing the public key and generating the opposite
algorithm to find the private key?
a. The public key is never communicated to another host.
b. The public key is first encrypted with a shared secret key.
c. Asymmetric keys are always paired with symmetric keys to ensure the algorithm cannot
be broken.
d. It is feasibly impossible to find the opposite formula with just the public key.
following topics:
■ Describing the problem of secure, scalable

distribution of public keys and presenting
PKI as a solution
■ Explaining the concept of a trusted introducer
■ Explaining certificates, CAs, certification
paths, certificate trust, and revocation lists
■ Explaining the certificate enrollment
procedure
■ Explaining PKI certificate revocation
■ Explaining the use of PKI in existing
applications
CHAPTER 25
Understanding the
Public Key Infrastructure
Cisco CallManager Release 4.0 and later supports many security features that are based on a
Public Key Infrastructure (PKI) solution. In earlier releases, Cisco CallManager did not use
PKI-like features, and many Cisco CallManager administrators are not familiar with PKI. This
chapter discusses the concept of a PKI, its components, and its applications.
The Need for a PKI

Scalable and secure key exchange is the main issue when deploying cryptography. Depending
on the cryptographic algorithm that is used, there are different needs. In symmetric encryption,
the keys should be changed frequently. They are shared between two peers. But how can you
exchange these keys safely between the peers? Symmetric keys should be known only by the
two peers using them, and, therefore, confidentiality must be ensured for the key exchange.
In asymmetric encryption, the public key of a device has to be known by all other devices (made
public). But how can you distribute the public keys safely to your devices? You have to ensure
that the public keys that are exchanged over the network are authentic, and hence authenticity
must be ensured for that key exchange.
Key Exchange in Symmetric Cryptography

When exchanging keys in symmetric cryptography, you have two possible options: You can use
out-of-band manual key exchange or you can use in-band automated key exchange.
Manual key exchange is the simplest method of exchanging secret keying material. However, it
does not scale and often relies on the human operator to perform the procedure securely. Every
peer with which the entity wants to exchange encrypted traffic must go through a one-time
manual key exchange. After the keys are generated, the two parties exchange the keys manually,
through a secure channel (for example, by telephone or in person). This process should include
an out-of-band method of authentication to ensure that the keys were exchanged unaltered with
the correct party. This concept is often applied when using authenticated routing protocol
updates, where the symmetric key that is used for the authentication has to be entered on all
participating routers. If multiple router administrators are involved, they can exchange the
symmetric key in person or use some other protected channel (such as encrypted e-mail and
Secure Shell [SSH Protocol]).
552 Chapter 25: Understanding the Public Key Infrastructure
Most key exchanges are automated and do not require any human intervention. A couple of good
methods for automatic key exchange are heavily used in modern cryptosystems. One of them is
the Diffie-Hellman algorithm, which allows two peers to compute the same value (key) without
exchanging all information that is needed for that computation. Another method is to send the
actual symmetric keys but encrypt them using an asymmetric encryption algorithm first. In Cisco
IP telephony, asymmetric encryption algorithms are used to exchange symmetric keys securely.
Key Exchange Protected by Asymmetric Encryption

When using asymmetric encryption to secure the automated exchange of symmetric keys, the keys
are encrypted with the public key of the receiver and then sent over the untrusted network. Only
the receiver can decrypt the message (the keys) because only the receiver knows the corresponding
private key. This solution relies on the knowledge of public keys of all possible peers at all the
participating devices.
In the example shown in Figure 25-1, user A wants to use symmetric encryption with user B.
Figure 25-1 Using Asymmetric Cryptography to Securely Exchange Symmetric Keys

AES Key AES Key
RSA RSA
A B
Public Key Private Key

of User B of User B
For secure key exchange, asymmetric encryption will be used in this way:
Step 1 User A generates the symmetric key.

Step 2 User A encrypts the symmetric key with the public RSA key of user B and
sends the encrypted key over the untrusted network to user B.
Step 3 User B decrypts the key using his private RSA key.
Step 4 Now both of them know the symmetric key and can start using it for
encrypting their communication channel.
As mentioned previously, this solution assumes that user A knows the public key of user B.
The Pitfall of Asymmetric Key Exchange

Asymmetric algorithms offer the advantage of one of the keys being public, which simplifies key
exchange and distribution. The pitfall of this approach is not obvious at first glance. Obtaining the
public key from another person can be very tricky. Although it is true that public keys are public
information and can be published in a well-known directory, an extremely important issue
remains: When you receive a public key from someone, how do you really know it belongs to that
person?
When a public key is requested or is sent over an untrusted network, an attacker could intercept
that key and substitute another public key for it. This man-in-the-middle attack would cause the
message sender to encrypt all messages with the public key of the attacker. A mechanism is,
therefore, needed that allows verification of the relationship between a name of an entity and its
public key. The PKI is the solution to this problem and allows such systems to scale, although, on
a smaller scale, alternative, manual solutions can be devised.
PKI as a Trusted Third-Party Protocol

PKI does not eliminate the need for authenticity when exchanging public keys in an asymmetric
encryption environment, but PKI solves the scalability issues associated with that process. It uses
the concept of a single, trusted introducer. Instead of securely exchanging all public keys among
all devices, only the public key of the trusted introducer has to be securely distributed to all
devices, as shown in Figure 25-2. This is usually done by downloading the public key and then
verifying it out of band. The trusted introducer performs the role of authentication for the devices:
If the devices are authenticated by the trusted introducer, they are considered authenticated to each
other. If they are not authenticated by the trusted introducer, they are not authenticated to each
other. Essentially, the devices have an explicit (configured) trust to believe anything the trusted
introducer tells them.
Figure 25-2 Using the Public Key of the Trusted Introducer

Trusted
Introducer
Public Key
Public Key of Trusted Public Key
of Trusted Introducer of Trusted
Introducer Introducer
Trust
Public Key Public Key
of User A of User C
A C
When all devices know the authentic key of the introducer, the introducer can guarantee the
authenticity of the public keys of all devices by using a certificate for each device in the topology.
The certificate includes information about the identity of a device and its public key. The (publicly
trusted) introducer then signs the certificates of the individual devices, and the devices can directly
distribute their public keys by sending their certificates. A device receiving such a certificate can
verify it by checking the signature of the issuer (the introducer).
Every user in the system trusts information provided by the introducer. In practice, this is
accomplished by digital signatures. Anything that the introducer signs is considered to be trusted.
To verify the signatures of the trusted introducer, each user of this system must first obtain the
public key of the trusted introducer. To become a part of the trust system, all end users enroll with
the introducer; that is, they submit their identity and their public key to the introducer, as shown
in Figure 25-3.
Figure 25-3 Exchanging Public Keys

of User A of User C
Trusted
Introducer

of Trusted of Trusted

of User A of User C
A C
Private Key Public Key Public Key Private Key
of User A of Trusted of Trusted of User C
The trusted introducer then verifies the identity and public key of each enrolling user and, if they
are correct, the trusted introducer digitally signs the submitted public key with the private key of
the introducer. The result is a kind of “document” (certificate) for each user that includes the
identity (name) of the user and the public key of the user. The trusted introducer provides each
user with a signed document, containing the name and public key of the user, bound together by
the signature of the trusted introducer. As shown in Figure 25-4, each user now possesses a public
and private key pair, the public key of the trusted introducer, and a document with the identity and
public key of the user. This document is signed by the trusted introducer.
Figure 25-4 Generation of a PKI Certificate

Trusted
Introducer
of Trusted of Trusted
Trusted Trusted
Public Key Public Key Public Key Public Key

of User A of User C of User A of User C
A C
Private Key Public Key Public Key Private Key
of User A of Trusted of Trusted of User C
Because all users now have their own documents containing the correct name and public key,
signed by the trusted introducer, and the public key of the trusted introducer, they can verify all
data signed by the trusted introducer. The entities can now (independently of the trusted
introducer) establish point-to-point trusted relationships by exchanging information about
themselves in the form of that document.
In practice, this means that at this stage the end users can mutually exchange signed public keys
over an insecure medium and use the digital signature of the trusted introducer as the protection
mechanism for the exchange. Again, the signature of the trusted introducer is trusted because it
can be verified (the entities have the public key of the trusted introducer), and the trusted
introducer and its operations are considered to be secure.
PKI Entities
A PKI is the service framework needed to support large-scale public key–based technologies. The
PKI is a set of all the technical, organizational, and legal components needed to establish a system
that enables large-scale use of public key cryptography to provide authenticity, confidentiality,
integrity, and nonrepudiation services.
Two very important terms need to be defined when talking about a PKI:
■ A Certificate Authority (CA) is the trusted third party (trusted introducer) that signs the public
keys of all end entities (PKI users).
PKI Entities 557
■ A certificate is a document that, in essence, binds the name of the entity and its public key
that has been signed by the CA, so that every other entity will be able to trust it.
NOTE Certificates are not secret information and do not need to be encrypted in any way. The
idea is not to hide anything but to ensure the authenticity and integrity of the information
contained in the certificate.
Another term that is often used with a PKI is certificate revocation list (CRL). The CRL is a list
of certificates that should not be trusted anymore. Examples of when a certificate is added to the
CRL (“revoked”) include exposure or loss of the private key. A PKI user who receives a certificate
should verify the CRL to ensure that the received certificate is not on the list of revoked
certificates.
Many vendors offer CA servers as a managed service or as an end-user product:
■ Microsoft Windows 2000/2003 Certificate Services (http://www.microsoft.com) is a

Windows Server add-on that allows an organization to set up its own CA server.
■ VeriSign (http://www.verisign.com) offers outsourced PKI services.
■ Entrust Technologies (http://www.entrust.com) offers both PKI products and outsourcing

services.
■ The CA Proxy Function (CAPF) in Cisco CallManager can act as a standalone CA.
X.509v3 Certificates
X.509 is the ubiquitous and well-known standard that defines basic PKI data formats, such as
certificate and CRL format, to enable basic interoperability. This format is already extensively
used in the infrastructure of the Internet. X.509 is used for these applications:
■ With secure web servers for website authentication in the Secure Sockets Layer (SSL)
protocol
■ With web browsers for services that implement client certificates in the SSL protocol
■ With user mail agents that support mail protection using the Secure/Multipurpose Internet
Mail Extensions (S/MIME) protocol
■ In IPsec virtual private networks (VPNs) where certificates can be used as a public key
distribution mechanism for Internet Key Exchange (IKE) Rivest, Shamir, and Adleman
(RSA)–based authentication
Figure 25-5 shows an example certificate format, following X.509 Version 3 (X.509v3). The most
important pieces of information contained in the certificate are these:
■ Subject X.500 Name (Name of the holder)
■ Subject Public Key Information (Public key)
■ CA Signature
Other fields include these:
■ Certificate serial number
■ Validity period of the certificate
■ Signature Algorithms
Figure 25-5 Sample X.509 Certificate Format
Certificate Format Version Version 3
Certificate Serial Number 12457801
Signature Algorithm
RSA with SHA-1
Identifier for CA
Issuer X.500 Name C= US O = Cisco CN = CA
Start = 04/01/04
Validity Period
Expire = 04/01/09
C= US O =Cisco
Subject X.500 Name
CN = CCMCluster001
Subject Public Key
756ECE0C9ADC7140…
Information
Extension(s) (v3)
2C086C7FE0B6E90DA396AB…
CA Signature
Self-Signed Certificates
In a PKI system, all public keys are distributed in a form of a certificate, including the certificate
of the trusted introducer, the CA. The obvious question is: Who signs the certificate of the CA, if
it is itself the signer of all other certificates? In reality, the CA also issues a certificate to itself, just
to have a consistent format for distributing its public key. This process is how the end entities
obtain the public key of the CA—by obtaining its self-signed certificate. The signature of a self-
PKI Enrollment 559
signed certificate of the CA cannot be verified using the standard method (verification by using the
public key of the signer) because that public key should actually be protected by the signature.
Therefore, other methods (such as manual verification) are needed to ensure the authenticity of a
CA certificate.
Sometimes end entities also sign their own certificates. This happens if that particular end entity
is not a part of a PKI but uses a PKI-enabled application. For example, a web server could generate
a private and public RSA key and sign its public key with its private key to create a self-signed
certificate. This certificate could then be used in Secure HTTP (HTTPS), where the web server
would present a self-signed certificate to the connecting web browser. However, how does the web
browser verify the presented certificate, if it was not issued (signed) by a known CA for which the
web browser has a locally available certificate? This web server certificate, therefore, cannot be
accepted automatically, but needs to be verified using some other method (such as the manual, out-
of-band verification that is also used in pre-PKI protocols).
■ Organizations will typically use self-signed certificates internally to save on the continual cost
of obtaining certificates from a publicly known Certificate Authority (CA). This causes web
browser clients to initially display warning messages when connecting to a server using a self-
signed certificate.
PKI Enrollment
PKI enrollment is the process of adding a PKI user (such as a person, a device, or an application)
to the PKI. The enrollment is done in the following way:
Step 1 An enrolling user obtains the CA certificate (self-signed) in which the

public key of the CA is embedded. This public key will be used to verify
the digital signature on certificates of the other entities.
Step 2 The enrolling user sends its identity information and public key to the CA.
Step 3 The CA verifies (authenticates) the user, signs the submitted information,
and returns the signed data in the form of a certificate.
Step 4 The user verifies the returned certificate using the public key of the CA
from the previously obtained CA certificate.
The enrollment procedure is the initial step of establishing trust between a user and the CA. If the
process is executed over an untrusted network, it would be vulnerable to man-in-the-middle
attacks. Therefore, it has to be secured in such cases.
Man-in-the-Middle PKI Enrollment Attack

Without any additional protection for the enrollment process, a man-in-the-middle attack can be
used to spoof identities. The attacker could replace the submitted public key of the user with the
public key of the attacker, causing the CA to possibly issue a certificate to the attacker instead of
to the legitimate user. The attacker could replace the real CA certificate with the false CA
certificate of the attacker when the end user requests the certificate of the CA. The end user would
then trust the CA of the attacker instead of the real CA.
NOTE The attacker would replace only the public key of the user, not the identity (name) of
the user. When the CA issues the certificate, the attacker can pretend to be the user by presenting
the certificate with the name of the user but the public key of the attacker.
Secure PKI Enrollment

To mitigate the risk of interception and key substitution during enrollment, the enrollment
procedure needs to incorporate two out-of-band authentication procedures:
■ Verification by the enrolling PKI user that the correct CA certificate has been received
■ Verification by the CA that it has received the correct enrollment information from the
enrolling PKI user
This can be done by out-of-band exchange of fingerprints of the messages (certificates). If the out-
of-band received fingerprint matches the fingerprint of the received message, the message is
authentic. However, if the enrollment is completed over a secure network, where interception is
not possible, those security procedures might be relaxed or omitted completely.
To verify that the correct CA certificate has been received, a local hash (fingerprint) of the received
information is calculated, as shown in Figure 25-6. This fingerprint is compared to the true CA
certificate fingerprint, obtained over the telephone or another secure channel. If they match, the
true CA certificate has been received.
When the user submits identity and public key information, a local hash (fingerprint) of the
submitted information is calculated again. The CA also performs a hashing procedure of the
received information. The CA then compares its hash of the received information to the hash of
the user of the submitted information over the telephone or any other secure channel. If the two
hashes match, the CA has received an unmodified enrollment request.
PKI Revocation and Key Storage 561
Figure 25-6 Manually Securing PKI Enrollment

of CA of User A
C99fe50B24p0… SHA-1 SHA-1
CA
Untrusted 19e7cD25k4L8…
C99fe50B24p0… Network
SHA-1 SHA-1 19e7cD25k4L8…

of CA of User A
PKI Revocation and Key Storage

A certificate and the public key included in the certificate and its associated private key have a
lifetime. When a certificate is issued, the CA sets the lifetime of the certificate. The lifetime of
certificates is usually relatively long (months to years). But how do you handle a situation in which
a key becomes compromised before its expiration? This would happen if, for instance, an intruder
steals a server’s private key. In such a case, all other entities have to know not to trust that private
key (and its corresponding public key).
The most common reasons why a certificate should not be trusted anymore include the following:
■ Private key compromise
■ Contract termination for that PKI user
■ Loss of private keys (for instance, because of device replacement)
A PKI can offer such a solution by revoking a certificate. Certificate revocation is the
announcement that a private key is not trustworthy anymore. You can revoke a certificate using
different methods.
PKI Revocation Methods

Keys that are not trusted anymore could be manually revoked by deleting the certificates and the
corresponding keys on all affected systems. This process does not scale, so a form of automatic
revocation is needed.
Automatic revocation can be achieved by different methods:
■ Certificate revocation lists (CRLs)—These lists contain all certificates that are no longer
valid. The CRL is signed by the CA and has a lifetime. It is stored in a Lightweight Directory
Access Protocol (LDAP)-accessible directory or on a web server and made publicly available.
It is the duty of the end user to download a fresh CRL after the lifetime of the current CRL
has expired. Whenever an end user wants to use a certificate, it should be checked against the
downloaded CRL.
■ Online Certificate Status Protocol (OCSP)—OCSP is a protocol designed for real-time

verification of certificates against a database of revoked certificates. Upon receipt of a
certificate of another user, the end user or device queries the OCSP server in real time to verify
whether the received certificate has been revoked. OCSP is newer and not yet widely used in
network infrastructures.
The main advantage of OCSP over CRLs is that it ensures up-to-date information because of the
real-time verification of the certificate. CRLs might contain stale information because they are
issued periodically, usually every couple of hours. If a key is compromised, a window of
vulnerability exists until the end user downloads a new CRL listing the certificate of the
compromised system. To at least limit this window of vulnerability, the CRL lifetime is used.
Key Storage
Secret (for symmetric algorithms) and private (for asymmetric algorithms) keys must be stored
securely because forgery and loss of privacy could result if their secrecy is compromised. The
measures taken to protect a secret or private key must be at least equal to the required security of
the messages encrypted with that key. Ideally, keys are never stored in cleartext form or in user-
accessible storage.
Keys, especially long-term keys (such as RSA), should be protected especially well. They are very
often stored on nonvolatile storage media:
■ Hard drives—For example, storing private RSA keys on a PC
■ Flash memory—Sometimes, in the form of a Personal Computer Memory Card International

Association (PCMCIA) card
■ Read-only memory (ROM)—For example, encryption keys that are hard-coded in hardware
PKI Example 563
Ideally, RSA keys are stored on Smart Cards or tokens where all key-related operations are done
so that the key itself does not even have to leave that device.
Smart Cards and Smart Tokens

A Smart Card or smart token is essentially a small computer, capable of performing basic
cryptographic operations and containing the protected secret keys within its internal memory. The
host computer, to which the Smart Card reader is attached, simply passes challenges to the card,
which, for example, computes an authentication response. This technique ensures that the private
key never leaves the card and provides one of the strongest key-protection methods available
today.
Any PKI-based application that uses certificates to distribute public keys can store the relevant
private key on a Smart Card instead of in some less well-protected memory (such as the hard disk
of the end user). The application software then off-loads all public key operations to the Smart
Card.
In Cisco IP telephony, the private RSA key used to sign the Certificate Trust List (CTL) is stored
on a smart token and never leaves it. The smart token is a small computer that can sign data fed to
it over the Universal Serial Bus (USB) interface.
NOTE More information on Smart Card and smart token technology can be found at http://
www.opencard.org, http://www.chipcard.ibm.com, and http://www.gemplus.com.
PKI Example
Among the first applications of a PKI were web browsers and web servers using SSL. With these
applications, the web server authenticates to the browser using a PKI system. HTTPS is widely
used on the Internet today and whenever secure web communication is needed (for example,
online banking and e-commerce).
Transport Layer Security (TLS) is the successor of SSL and is application-independent, hence not
limited to HTTP traffic. TLS is very similar to SSL and also uses a PKI system to provide secure
communication.
S/MIME, used for secure messaging, is another example of an application that relies on a PKI.
PKI and SSL/TLS

If a web server runs sensitive applications, SSL or TLS is used to secure the communication
channel between the client and the server. A company that needs to run a secure web server (a
server supporting authenticated and encrypted HTTP sessions) first generates a public and private
key on the web server. The public key is then sent to one of the Internet CAs, which, after verifying
the identity of the submitter, issues a certificate to the server by signing the public key of the web
server with the private key of the CA.
Internet CAs are mainly run by either specialized private companies (such as VeriSign),
telecommunications companies, or governments. The certificates of those CAs are embedded at
installation into client operating systems (such as Microsoft Windows) or inside browsers (such as
Mozilla). The collection of embedded CA certificates serves as the trust anchor for the user. The
user can then verify the validity of signatures of any other certificate signed using the public keys
contained in those CA certificates.
Web Server Certificate Exchange

Figure 25-7 shows part of the list of the CA certificates embedded in the web browser. To see the
CA certificates installed in your computer, open Microsoft Internet Explorer, choose Tools >
Internet Options, choose the Content tab, click Publishers, and choose the Trusted Root
Certification Authorities tab.
Figure 25-7 Default Trusted Root Certificate Authorities in Windows XP
When a browser contacts a secure web server using HTTPS, the first step of the protocol is to
authenticate the web server—to verify that the browser indeed has connected to the correct web
server, as desired by the user. Figure 25-8 depicts an example in which a user connects to https://
www.amazon.com.
PKI Example 565
Figure 25-8 Web Server Certificate Verification
VeriSign Amazon.com
Web Browser Web Server
Public Key
of VeriSign
Verify
2 Signature
VeriSign VeriSign Public Key of

Web Server
Internet
1
Public Key of Public Key of Public Key of Private Key of
Web Server Web Server Web Server Web Server
Authentication of the web server uses a challenge-response method, with which the server will
prove that it possesses the private key of the desired server (www.amazon.com). However, to prove
possession of the private key of the server, the browser needs the public key of the server first. Here
is the sequence of events:
Step 1 When the client connects to the www.amazon.com web server, the web
server first sends its certificate, signed by a well-known Internet CA to the
client.
Step 2 The client uses one of the local root CA certificates (the issuer’s certificate
of the server certificate) to verify its validity, and optionally downloads the
CRL to verify that the server certificate has not been revoked.
Step 3 If verification is successful, the client now knows that it has an authentic
public key of the www.amazon.com server in its possession.
Next, the client challenges the web server to verify that the web server has the private key that
belongs to the public key that the client received in the certificate of the web server. This private
key should be known only to the www.amazon.com web server:
Step 1 The client generates random data, and sends it to the web server to be
encrypted with the private key of the web server (challenge), as shown in
Figure 25-9.
Figure 25-9 Authenticating the Web Server
Web Browser Web Server
Random String Sign

Challenge 1
Rj@as94iDg… RSA
Private Key of
Internet Web Server
RSA Response
p2CksD1f3r…
2
Public Key of
3
Web Server
Rj@as94iDg…
Step 2 The web server signs (RSA-encrypts) the random data using its private key
and returns the signed random data to the client (response).
Step 3 The client verifies (RSA-decrypts) the signed random data using the public
key of the server from the verified certificate and compares it against the
random data that the client generated previously.
Step 4 If the signature is authentic, the web server really possesses the private key,
corresponding to the public key in the certificate of the web server
(www.amazon.com), and is, therefore, authentic.
NOTE In this example, the web server was authenticated by the client; the web server,
however, has no idea about the identity of the client. Authentication of the client is optional in
SSL and TLS and, if used, would use exactly the reverse procedure to authenticate the client,
provided that the client also possesses a private and public RSA key pair and a certificate
recognized by the web server. However, most servers choose to authenticate the client using a
simple username/password mechanism over the secure SSL/TLS session because this method is
easier to deploy than client-side certificates.
Using the authentic certificate of the web server, the client can now generate and send session
keys, which are used by the symmetric encryption algorithm of SSL or TLS to communicate
securely with the web server. As shown in Figure 25-10, the exchange of session keys is done in
the following way:
Step 1 The client generates symmetric session keys for SSL or TLS Hash-based
Message Authentication Code (HMAC) and encryption algorithms.
PKI Example 567
Step 2 The client encrypts these keys using the www.amazon.com public key of
the web server and sends them to the server.
Step 3 The www.amazon.com web server (only) can decrypt the such-encrypted
session keys using its private key.
Figure 25-10 Exchanging Session Keys
Web Browser Generate Web Server

Session Keys
2
ke4P6d23Le… RSA
Session Keys Internet Private Key of
Web Server
3
RSA
Public Key of
Session Keys
Web Server
Now, after the client and the server have shared the secret session keys, they can use them to
exchange authenticated (signed using the HMAC algorithm) and encrypted (using a symmetric
encryption algorithm, such as Triple Data Encryption Standard [3DES], Advanced Encryption
Standard [AES], or RC4) messages, as shown in Figure 25-11.
Figure 25-11 Session Encryption
Web Browser Data Web Server

Data
3DES
Ss199le4… 3DES
Session Keys
Session Keys Internet

Text 3DES
Text 3DES dV46ax7…

Summary
This chapter discussed the concepts behind the Public Key Infrastructure (PKI). PKI is needed for
secure, scalable key exchange over larger corporate networks and public networks, such as the
Internet. PKI uses the concept of a single “trusted introducer” to eliminate the need for any-to-any
authentication. These trusted introducers are known in PKI as Certificate Authorities (CAs), which
issue certificates, through a process known as enrollment, to its associated users. PKI revocation
allows these certificates issued to the clients to be announced as invalid before the lifetime of the
certificate has expired. This process is necessary when private keys become compromised and,
therefore, cannot be trusted anymore. SSL, TLS, and S/MIME are common examples of PKI-
enabled protocols.
Review Questions
1. What problem does PKI solve?

a. The lack of a common encryption standard for Internet applications
b. The problem that asymmetric encryption techniques do not work without a PKI
c. The fact that Diffie-Hellman is not secure
d. The problem of scalable, secure key exchange
e. The problem of manually issuing bulk certificates
f. The performance problem when using RSA
2. Which two statements about symmetric encryption are true?
a. Symmetric encryption is a good choice for real-time encryption of bulk data.
b. Symmetric encryption is commonly used to sign asymmetric keys.
c. Symmetric encryption uses asymmetric keys.
d. RSA is an example of a symmetric encryption algorithm.
e. ASE is an example of a symmetric encryption algorithm.
f. With symmetric encryption, the encryption key equals the decryption key.
3. Which two statements about trusted introducers are incorrect?
a. The trusted introducer has to be trusted by all other members of the system.
b. The trusted introducer has to trust all other members of the system.
c. The trusted introducer guarantees the authenticity of entities it is introducing to others.
d. Only the trusted introducer has to trust the root of the system.
e. The trusted introducer is the root of a system.
f. Any entity of the system can guarantee the authenticity of any other member.
4. Which statement about a certificate is true?

a. A certificate includes the identity of the owner of the certificate and the symmetric key
of the owner.
b. A certificate includes the public key of the issuer.
c. A certificate includes the identity of the owner of the certificate and the private key of
the owner.
d. A certificate includes the identity of the issuer of the certificate, the identity of the owner
of the certificate, and the public key of the owner.
e. A certificate does not include any keys in cleartext.
f. A certificate includes an encrypted private key of the owner and a cleartext public key of
the issuer.
5. Which of the following are the two valid options to secure enrollment in a PKI?
a. Perform the enrollment from a trusted device only.
b. Perform the enrollment in both directions.
c. Perform the enrollment over a trusted network.
d. Use self-signed certificates on all devices.
e. Do not send the private key in the enrollment.
f. Perform mutual out-of-band authentication between the PKI user and CA.
6. Which of the following is true about certificate revocation?
a. Any entity of a PKI system that receives an untrusted certificate can request revocation
of that certificate.
b. The CA periodically revokes all expired certificates.
c. Certificate revocation is needed when the public key has been transferred without a cer-
tificate.
d. Certificate revocation is needed whenever the private key is not trustworthy anymore.
e. Certificate revocation is needed whenever the public key is not trustworthy anymore.
f. Certificate revocation is the process of adding a user to the PKI.
7. What is the certificate of a web server used for when you are using SSL?
a. It is used to authenticate the client.
b. The public key of the server is used by the client when encrypting the data sent to the
server.
c. The private key of the server is used by the client when encrypting the data sent to the
server.
d. It is used to authenticate the server and to protect the challenge response traffic during
client authentication.
e. It is used to authenticate the server and to encrypt the symmetric session keys used for
the asymmetric encryption of the data stream.
f. It is used to authenticate the server and to encrypt the symmetric session keys used for
the authentication and encryption of the data stream.
8. Which of the following is an asymmetric encryption algorithm?
a. AES
b. Diffie-Hellman
c. DES
d. Blowfish
9. When using asymmetric encryption, which key is transmitted from the sending to the
receiving host?
a. Public key
b. Private key
c. Shared secret key
d. No keys are transmitted.
10. A nontrusted user has obtained the certificate of one of your public web servers. What should
be done to ratify this situation? (Select all that apply.)
a. Nothing
b. The certificate should be immediately revoked and published to a CRL.
c. The certificate should be immediately revoked. Clients can immediately find this revo-
cation if they are using OCSP.
d. The private key should be regenerated on the web server.
following topics:
■ Explaining how file manipulation, tampering

with call-processing signaling, man-in-the-
middle attacks, eavesdropping, and IP phone
and server identity theft can compromise a
Cisco CallManager system
■ Explaining how the authentication and
encryption mechanisms in a Cisco
CallManager system protect against security
threats
■ Explaining the role of CAPF, external CAs,
MIC and LSC, CTLs, and Cisco CTL client
■ Explaining the PKI enrollment process in a
Cisco IP telephony environment
■ Explaining where keys and certificates are
stored in a Cisco IP telephony environment
■ Describing the processes of image
authentication, device authentication, file
authentication, and signaling authentication
■ Describing the processes and protocols used
for signaling encryption and media
encryption
CHAPTER 26
Understanding Cisco IP
Telephony Authentication and
Encryption Fundamentals
Cisco IP telephony systems are subject to several threats, including eavesdropping, identity
spoofing, and denial of service (DoS) attacks. In Cisco CallManager Release 4.0 and later, the
Cisco IP telephony solution can be secured against these threats by enabling authentication and
encryption features. This chapter explains how authentication and encryption can be applied in
a Cisco IP telephony environment.
Threats Targeting the IP Telephony System

The main threats targeting the IP telephony system are as follows:
■ Loss of privacy—If calls can be sniffed, conversations can be replayed or eavesdropped.

Because of the open nature of IP networks and, especially, the low trust level in the Internet,
privacy is a commonly raised concern when comparing IP telephony solutions against
traditional telephony solutions.
■ Loss of integrity—If voice call signaling or media messages can be intercepted, they can
be modified. Although this issue might not arise for human conversations very often, do not
forget that calls such as telebanking, e-mail or voice-mail access over a telephone, and
similar applications using interactive voice response (IVR) might require secure
communication.
■ Impersonation—Identity spoofing is not limited to humans (for instance, the person

behind a telephone) but can also extend to devices, such as Cisco CallManager, a voice
gateway, or an IP phone.
■ Denial of Service (DoS)—These attacks cause loss of functionality. They can be directed
against IP telephony components, such as voice gateways, Cisco CallManager nodes, or IP
phones, or against the underlying infrastructure. An example is replacing IP phone
configuration files or images stored on a TFTP server with invalid files that cause the IP
phone to malfunction.
574 Chapter 26: Understanding Cisco IP Telephony Authentication and Encryption Fundamentals
How CallManager Protects Against Threats

Cisco designed Cisco CallManager 4.0 with security largely in mind. A Cisco IP telephony
network can now be protected by using cryptographic services. These services are used to provide
the following:
■ Secure signaling—For authentication of devices and authentication and encryption of

signaling messages. This precaution stops all kinds of signaling attacks.
■ Secure media transfer—For authentication and encryption of media streams, preventing

eavesdropping on conversations.
■ Authentication of phone images—To stop attacks against phone images by ensuring the
integrity of the image file.
■ Authentication of phone configuration files—To stop attacks against phone configuration

files, again by ensuring the integrity of the file.
Secure Signaling and Media Transfer

Secure signaling in Cisco IP telephony provides authentication and authorization of
communicating devices (Cisco IP Phones and Cisco CallManager) and authentication of the
signaling messages exchanged between them. It can also provide encryption of the signaling
messages. Securing the call signaling is mandatory if you plan to secure the media transfer as well.
The reason for this precaution is that the keys used for securing the media channels are exchanged
inside signaling messages.
Secure signaling is achieved by using Transport Layer Security (TLS) and is based on the Cisco
IP telephony Public Key Infrastructure (PKI) solution. The secure signaling encapsulates the
Skinny Client Control Protocol (SCCP, or Skinny) messages in TLS. TLS provides transport-layer
protection and is similar to Secure Sockets Layer (SSL), used for secure web browsing.
Secure media transfer in Cisco IP telephony provides confidentiality by encrypting the media
stream. If a hacker captures the media streams, the hacker cannot interpret them or play them back.
Secure media transfer also provides integrity and authenticity so that the packets cannot be altered
while in transit. If an attacker modifies, removes, or adds Real-Time Transport Protocol (RTP)
packets, the receiver detects this manipulation because of the missing or incorrect authentication
data. Secure media transfer requires encrypted call signaling because the media encryption keys
are exchanged over signaling channels. After you encrypt the media stream, the call is considered
a Secure RTP (SRTP) session.
Figure 26-1 illustrates that for secure media transfer, SRTP is used instead of the insecure RTP to
exchange voice packets between IP phones. Encapsulating the Skinny protocol inside of TLS
encryption ensures secure communication between the IP phone and the CallManager. SRTP is a
How CallManager Protects Against Threats 575
standard-based (RFC 3711, The Secure Real-Time Transport Protocol) and an application-layer
encryption that performs inside-payload encryption where the protocol headers do not change.
Because the headers in RTP and SRTP are the same, an attacker who sniffs the conversation does
not know whether the RTP stream has been encrypted when examining the packet header only.
Only when further analyzing the sniffed packets and trying to play them back can the attacker
recognize that the audio has been encrypted.
Figure 26-1 Secure Signaling and Media Transfer
SCCP in TLS SCCP in TLS
SRTP
IP IP
Authentication of Phone Images

To ensure the integrity of Cisco IP Phone images that are loaded from a TFTP server, authenticated
images are used. Cisco IP Phones support image authentication on all Cisco IP Phone models.
With image authentication, Cisco manufacturing signs the images (using a private key) and
appends the signature to the actual firmware. This signature ensures the firmware is from Cisco
Systems. Most modern Cisco IP Phones already include the Cisco Systems public key to verify
that the signature is accurate, as shown in Figure 26-2. In addition, this feature also allows phones
to check the image device type so that incorrect images (those for other phone models) are not
loaded.
Figure 26-2 Phone Image Verification
Binary
Executable
TFTP Server File
Cisco IP Phone
Image Signature Public Key
of Cisco
CTL
Image.bin.sgn
Image.bin.sgn
Config1.xml.sgn TFTP
Config2.xml.sgn IP
Config3.xml.sgn
IP phone image authentication was introduced with Cisco CallManager Release 3.3(3). In this and
later versions, phone images include the public key that corresponds to the private key used by
Cisco manufacturing to sign phone images. In addition, the firmware accepts new images only if
their signature is authentic.
IP phone image authentication does not need any additional configuration and is totally
independent of the Cisco IP telephony PKI that is used for other features.
TIP If you need to downgrade to an IP phone image that does not yet support IP phone image
authentication (earlier than Cisco CallManager Release 3.3(3)), a special “breakout” image can
be obtained from the Cisco Technical Assistance Center (TAC). Simply trying to load an older
image does not work because the current image will accept only signed images.
Authentication of Phone Configuration Files

In addition to IP phone images, IP phone configuration files can be signed as well. This eliminates
man-in-the-middle attacks on the Cisco IP Phone configuration files, which would attempt to
direct the IP phone to an alternate (rogue) CallManager server.
Signed IP phone configuration files are implemented differently from signed images. The
configuration files are signed by the Cisco TFTP server (with its private key). An IP phone loading
a new configuration verifies the configuration file before applying it. The IP phone needs the
public key of the TFTP server to do so. Except for the Cisco development public key, the public
key of the TFTP server is different for every installation and, therefore, cannot be embedded in the
firmware of the IP phone. Therefore, verification must use the Cisco IP telephony PKI.
Authenticated IP phone configuration files prevent tampering with the files on the TFTP server or
in transit.
NOTE Because authenticated IP phone configuration files depend on the existence of a Cisco
IP telephony PKI, the deployment of this feature is far more complex than signed IP phone
images. On the other hand, when you enable your cluster for security, authentication of phone
configuration files is automatic for all IP phones that are configured for secure operation.
PKI Topologies in Cisco IP Telephony

Unlike classic enterprise PKI deployments, the PKI topology in Cisco IP telephony is not a single
PKI system. Instead of having a single Certificate Authority (CA) that issues all certificates, there
are several instances issuing certificates:
■ Self-signed certificates—Cisco CallManager and other servers issue their certificates on

their own.
■ Certificates signed by the Cisco manufacturing CA—Cisco IP Phone 7970 models (and
later) have manufacturing installed certificates (MICs).
■ Certificates signed by Cisco CallManager Certificate Authority Proxy Function (CAPF)

or by an external CA—One of these two options is used to issue locally significant
certificates (LSCs) to Cisco IP Phone models that support SRTP.
Self-Signed Certificates PKI Topology

Figure 26-3 illustrates several IP telephony services with self-signed certificates. This design
requires each server (or server component) have its own self-signed certificate. In the example
shown, the Cisco CallManager, TFTP, and CAPF servers all have created their own self-signed
certificates. Each one of them act as their own PKI root (they establish their own authority).
Figure 26-3 Self-Signed Certificate PKI Topology

PUB TFTP
TFTP
PUB
Private Key Private Key
of PUB of TFTP
of PUB of TFTP
of PUB of TFTP
SUB1 CAPF
CAPF
SUB1
of SUB1 of CAPF
of SUB1 of CAPF
of SUB1 of CAPF
Manufacturing Installed Certificates PKI Topology

Currently, the Cisco 7911, 7941, 7961, 7970, and 7971 IP Phones have a certificate installed by
Cisco in nonvolatile memory. This allows the IP phone to be capable of communicating securely
with the CallManager out of the box. Cisco highly recommends that you replace these MICs with
your own certificate as soon as possible. Figure 26-4 illustrates the keys used when an IP phone
comes with an MIC preinstalled.
Figure 26-4 MIC PKI Topology
Phone Cisco CA
of Phone Issue Certificate of Cisco CA
During Production
Public Key
of Phone IP
of Phone Cisco CA Cisco CA of Cisco CA

of Cisco CA of Cisco CA
The IP phone with the MIC has its own public and private key pair and a Cisco manufacturing CA
certificate installed. The certificate of the IP phone is signed by the Cisco manufacturing CA,
which makes the Cisco manufacturing CA the PKI root for all MICs.
Locally Significant Certificates PKI Topology

LSCs are certificates that you install to the IP phone. These can be installed from the CAPF
CallManager function or from another CA. When deploying the LSCs to the IP phones, you have
a major choice to make:
■ Deploy LSCs through a centralized, automated process to all IP phones attached to the
network.
■ Deploy LSCs manually on a phone-by-phone basis.
Of course, the factor that this decision hinges on is risk. By automatically deploying LSCs to the
IP phones, you assume that all the phones that are currently plugged into the network are
“friendly.” The CallManager will automatically deploy LSCs to all phones. If a hacker has a phone
plugged into the network, it will also get an LSC and become a trusted device. The alternate,
manual method creates a huge amount of administrative overhead because you must visit each
device and manually enter a sequence of digits provided by the CAPF. However, by following this
method, you ensure that each IP phone be physically verified before it is considered a trusted
device.
As shown in Figure 26-5, the CAPF can either generate LSCs itself, or it can act as a proxy
between the IP phone and some other CA.
Figure 26-5 LSC PKI Topology

CAPF Acting CAPF Acting
as a CA as a Proxy
PUB PUB Enterprise CA
CAPF CAPF
Enroll
Enroll Enroll
IP IP
Independent, Separated PKI Topology

As illustrated in Figure 26-6, your IP telephony network might have several, coexisting PKI
topologies. There is no single root; instead, there are multiple independent PKI topologies. So far,
there is no trust relationship among these different PKIs. A trusted introducer is needed, bringing
all the different PKI topologies together under one trusted source. The Certificate Trust List (CTL)
client provides that function.
Figure 26-6 Multiple PKI Topologies
PUB TFTP
TFTP
PUB Private Key Private Key
of PUB of TFTP
of PUB of TFTP
of PUB of TFTP
MIC LSC 7940/

Cisco CA 7970 CAPF
7960
IP IP
Cisco CA CAPF

of 7970 of 7940/7960
CTL Client
The Cisco CTL client itself has a certificate issued by the Cisco manufacturing CA. The goal of
the Cisco CTL client is to obtain the certificates of all entities that issue certificates (self-signed
certificates only or certificates for other devices). Then, the Cisco CTL client signs the list of these
certificates, the CTL, using its private key. The public and private keys of the Cisco CTL client are
stored on a smart token called a security token, which is just a USB key plugged into the PC
running the CTL client.
Now there is a single, trusted introducer in the system again: The Cisco CTL client “introduces”
trusted devices, not by signing their certificates, but by signing a list of trusted certificates signed
by several PKI roots, as shown in Figure 26-7. The CTL can be compared to the root certificate
store of Microsoft Internet Explorer. Both are a list of trusted certificate-issuing entities.
Figure 26-7 CTL Client Creating a List of Trusted Certificates

Signed List of Certificate Issuers
PUB CTL Certificate Trust List (CTL)
PUB
Client
TFTP PUB CAPF
Public Key
of PUB
Public Key Public Key Public Key
TFTP TFTP Cisco CA of TFTP of PUB of CAPF
Private Key
Cisco CA Cisco CA
Public Key of CTL
of TFTP Client Public Key
of CTL
CAPF Client
CAPF of Cisco of CTL
Public Key CA Client
of CTL
Public Key Client
of CAPF
The CTL usually includes these certificates:
■ Cisco CallManager certificates—Each Cisco CallManager has a self-signed certificate. It

allows the Cisco CallManager to authenticate to a device (IP phone) during device
registration.
■ TFTP server certificate—The TFTP server that provides the IP phone with files, such as the
IP phone image or the IP phone configuration file, is trusted by the IP phone only if the TFTP
server is listed in the CTL of the IP phone.
■ CAPF certificate—When you are using LSCs, the CAPF issues certificates to the IP phones.
The certificate of the CAPF allows the CAPF to authenticate to an IP phone during the
enrollment.
■ Cisco certificate—MICs and the certificate of the security tokens (storing the keys used by
the Cisco CTL client) are issued by Cisco manufacturing CA. To allow the phone to verify
certificates issued by this Cisco CA, the phone needs the certificate of the Cisco
manufacturing CA.
■ Cisco CTL client certificate—The Cisco CTL client signs the CTL using one of the security
tokens. The certificates of the Cisco CTL client (one per security token) have to be known to
the IP phone to allow verification of the signature of the CTL.
As shown in Figure 26-8, When an IP phone boots, the CTL is downloaded from the TFTP server
to the IP phone. It contains all certificates of the entities that issued self-signed certificates. By
having this list, the IP phone knows which PKI roots to trust and can trust all certificates that have
been issued by any PKI root contained in the CTL.
Figure 26-8 Obtaining the CTL
CTL Client Cisco CA
TFTP PUB
Public Key
of Phone
TFTP
of TFTP of PUB
Cisco CA IP Private Key

of Phone
Public Key of
CTL Client Public Key
of Phone
CTL Client Application

The Cisco CTL client software, available as a plug-in application on Cisco CallManager
Administration, is used to create or update the CTL. When the list is accurate, the Cisco CTL client
will ensure that the CTL is signed by the keys of the Cisco CTL client. These keys are stored on
an external Universal Serial Bus (USB) device, which is called the security token. When the CTL
needs to be signed, the Cisco CTL client passes the CTL to the security token, and the security
token signs it and then returns the signed CTL to the Cisco CTL client application. The Cisco CTL
client itself does not have access to the private key stored on the security token. Therefore, it is not
the CTL client application that actually signs the CTL; the CTL client only interacts with the
security token requesting the security token to create the signature. The public key of a security
token is signed by the Cisco manufacturing CA during production, and the appropriate certificate
is also stored on the security token itself.
The CTL file needs to be updated after configuration changes, such as changing or adding IP
telephony servers or security tokens to the system.
The CTL also acts as an authorization list because it specifies which certificates belong to which
IP telephony function. A TFTP server, for instance, is not allowed to sign a CTL, only IP phone
configuration files. The CAPF, as another example, is allowed to sign the LSCs of other IP phones
only but not the CTL or any TFTP files.
CTL Verification on the IP Phone

Every time that an IP phone receives a new CTL, the new CTL is verified. It is accepted by the IP
phone only if it was signed by the Cisco CTL client using one of the administrator tokens. The
phone can verify the signature using the public key (the certificate) of the Cisco CTL client (in
fact, the certificate of the appropriate administrator token), which must be included in the currently
installed (“old”) CTL.
This certificate of the administrator token is signed by the Cisco manufacturing CA. This signature
is also validated by using the certificate of the Cisco manufacturing CA, which also must be
included in the currently installed CTL. This concept works well as long as the phone already has
a CTL.
Initial Deployment Issue

For the first download of a CTL to an IP phone, there is an issue: How does an IP phone know
which administrator tokens are trusted without already having a CTL? This problem occurs only
at initial deployment, when the phone does not yet have a local CTL. In this case, any
administrator token could pretend to be a valid token for the IP telephony system in question. An
attacker could either replace the CTL file on the TFTP server with a falsified file or change the
CTL file in the path between the IP phone and the TFTP server.
The problem can be solved by downloading the initial CTL over a trusted network to ensure that no
falsified initial CTL is loaded to the phone. When the phone has a valid CTL, it will trust new CTLs
only if they are signed using a security token that is already known to the IP phone. If the CTL file
in the IP phone is erased, the same problem occurs. Again, you must ensure that the next CTL
download is done over a trusted network path because the IP phone will blindly accept any CTL.
After the IP phone is deployed, it is usually difficult to trust the network path between the phone
and Cisco CallManager. Therefore, a user should not be able to erase the initially installed CTL.
There are two ways to remove a CTL from an IP phone: a factory reset or the IP phone Settings
menu. A factory reset is not simple, but using the Settings menu is rather easy. To prevent users
from using the Settings menu to remove the CTL, you should disable settings access at the phone.
PKI Enrollment in Cisco IP Telephony 583
When using authentication strings as the authentication method during CAPF phone certificate
operations, you have to enable settings access during the enrollment. After successful enrollment,
you should then disable settings access again.
PKI Enrollment in Cisco IP Telephony

To obtain a signed certificate, an IP phone needs to enroll with the entity that will issue (sign) the
certificate. During enrollment, the phone will get the certificate of the issuer and then send its data
to the issuer asking for a (signed) certificate. IP phone enrollment depends on the type of
certificate.
With MICs, enrollment was already done by Cisco manufacturing during production. When the IP
phone is shipped to the customer, it already has its public and private keys, a certificate issued by
the Cisco manufacturing CA, and the certificate of the Cisco manufacturing CA installed. No other
PKI provisioning tasks are required. MICs always remain on the phone, even if an LSC is added.
With LSCs, enrollment has to be done by the customer.
NOTE If the IP phone has both a MIC and an LSC, the LSC has priority.
CAPF Acting as a CA
To obtain an LSC from the CAPF acting as a CA, an IP phone has to enroll with the CAPF, as
Figure 26-9 CAPF Enrollment Process
CAPF CAPF
Public Key Public Key 5

4
of CAPF of Phone
PUB
Public Key
CAPF of Phone
Enroll 3
IP
2
CAPF 1
Private Key Public Key Private Key Public Key
of CAPF of CAPF of Phone of Phone
Public Key
of CAPF
The CAPF enrollment process is as follows:
1. The IP phone generates its public and private key pairs.

2. The IP phone downloads the certificate of the CAPF and uses it to establish a TLS session
with the CAPF.
3. The IP phone enrolls with the CAPF, sending its identity, its public key, and an optional
authentication string.
4. The CAPF issues a certificate for the IP phone signed with its private key.
5. The CAPF sends the signed certificate to the IP phone.
CAPF Acting as a Proxy to an External CA

If an IP phone should obtain an LSC from an external CA using the CAPF as a proxy, the IP phone
has to enroll with the external CA, as shown in Figure 26-10.
Figure 26-10 CAPF External CA Enrollment Process
CA CA CA
Public Key Public Key 6 Public Key 7

5
of CA of Phone of Phone
PUB Public Key

Enterprise of Phone
CA CAPF 3 Enroll
IP
4 CAPF
1
2
of Phone of Phone
of CA of CA Public Key
of CAPF
The external CA enrollment process occurs as follows:
1. The IP phone generates its public and private key pairs.

2. The IP phone downloads the certificate of the CAPF and uses it to establish a TLS session
with the CAPF.
3. The IP phone sends an enrollment request to the CAPF, including its identity, its public key,
and an optional authentication string.
4. The CAPF forwards the request to the external CA.

5. The external CA issues a certificate for the IP phone signed with the private key of the CA.
6. The external CA sends the signed IP phone certificate to the CAPF.
7. The CAPF sends the signed IP phone certificate to the phone.
Keys and Certificate Storage in Cisco IP Telephony

Key storage is a major part of key management because an improperly stored key can enable an
attacker to compromise parts of the PKI or the whole PKI. The IP phone stores its public and
private RSA keys and its certificate in its nonvolatile memory. This information is preserved across
phone reboots and resets. The keys cannot be extracted from the IP phone unless the phone is taken
apart and the nonvolatile memory is then physically analyzed.
The IP telephony servers (Cisco CallManager, CAPF, and TFTP server) store certificates on the
local hard disk, in a special area called the Microsoft certificate store. The private key of the server
is stored in the private-key storage. The private-key storage is protected by the periodically
changed master key. The master key itself is encrypted with Triple Data Encryption Standard
(3DES) using a key derived from the password of the user.
Microsoft Windows XP stores a certificate locally on the computer or device that requested it or,
in the case of a user, on the computer or device that the user used to request it. The storage location
is called the certificate store.
The Cisco CTL client stores its public and private RSA keys on the security tokens supplied by
Cisco. The keys are embedded on the token during production, and the token is designed never to
leak these keys from its memory.
Authentication and Integrity

Cisco CallManager allows authentication of calls. When you are configuring devices for
authenticated calls, two services are provided:
■ Device authentication for the IP phone and the server—Achieved by using device
certificates and digital signatures
■ Authentication and integrity of signaling messages—Achieved by using TLS Secure Hash

Algorithm 1 (SHA-1) Hash-based Message Authentication Code (HMAC) (with symmetric
keys)
Certificate Exchange in TLS

At the beginning of a TLS session, the Cisco CallManager server and the IP phone exchange
certificates using the messages shown in Figure 26-11.
Figure 26-11 Certificate Exchange Process

Phone Hello
Cisco CallManager Hello
Cisco CallManager Cert.
Cert. Req.
Phone Cert.
IP
The certificate exchange process occurs as follows:
1. The IP phone and the Cisco CallManager server negotiate the cryptographic algorithms in the
IP phone Hello and Cisco CallManager Hello messages.
2. The server sends its (self-signed) certificate to the IP phone.
3. The server requests a certificate from the IP phone.
4. The IP phone sends its certificate to the server.
At this point, both the IP phone and the server validate the certificates they just received over the
network:
■ The IP phone simply looks up the certificate of the server in its local certificate store. The
received certificate must be found locally because it must have been sent in the CTL. If it is
not included in the CTL, the session is dropped. If it is found, the public key of the server is
extracted from the certificate.
■ The server looks up the IP phone in the local device database to see if this IP phone is known
and authorized to connect via TLS. Then, the certificate of the IP phone is validated using the
locally available CAPF public key (from the CAPF certificate), and if valid, the public key of
the IP phone is extracted from the IP phone certificate.
Server-to-Phone Authentication
The next stage of the TLS handshake is authentication of the server by the IP phone. A simplified
version of the authentication steps is shown in Figure 26-12.
Figure 26-12 Server-to-Phone Authentication

Phone Hello
Cert. Req.
Phone Cert.
Challenge1
IP
Response1
The CallManager-to-Phone authentication occurs as follows:
1. The IP phone generates a random challenge string and sends it to the server, requesting that
the server sign it with the private RSA key of the server.
2. The server signs the message with its private RSA key and returns the result (response) to the
IP phone.
3. The IP phone verifies the signature using the public key of the server.
Phone-to-Server Authentication
After the server has authenticated to the IP phone, the IP phone needs to authenticate to the server.
A simplified version of the authentication steps is shown in Figure 26-13.
Figure 26-13 Phone-to-Server Authentication

Phone Hello
Cert. Req.
Phone Cert.
Challenge1
IP
Response1
Challenge2
Response2
The Phone-to-CallManager authentication occurs as follows:
1. The server generates a random challenge string and sends it to the IP phone, requesting that
the IP phone sign it with the private RSA key of the IP phone.
2. The IP phone signs the message with its private RSA key and returns the result (response) to
the server.
3. The server verifies the signature with the public key of the IP phone.
NOTE In the certificate of the IP phone, the public key of the IP phone is tied to the identity
of the IP phone. Because Cisco CallManager identifies an IP phone by MAC address and not by
IP address or name, the MAC address of the phone is used as the identifier in the certificate of
the IP phone.
TLS SHA-1 Session Key Exchange

After the bidirectional authentication, a SHA-1 session key is exchanged using these steps:
1. The IP phone generates a session key for SHA-1 hashing.

2. The IP phone encrypts it using the public RSA key of the server and sends it to the server.
3. The server decrypts the message and thus also knows which key to use for SHA-1 hashing of
the TLS packets.
The IP phone and the server can now exchange signaling messages over authenticated TLS
packets, ensuring the integrity and authenticity of each signaling message exchanged between the
two.
Encryption
In addition to authentication and integrity, Cisco CallManager also provides confidentiality of
calls by using encryption. When configuring devices for encrypted calls, signaling messages and
media streams are encrypted as follows:
■ Signaling messages are encrypted using TLS encryption with Advanced Encryption Standard
(AES) 128-bit encryption.
■ Media streams are encrypted using SRTP with AES 128-bit encryption.
Encryption 589
To ensure the authenticity of encrypted packets, in Cisco CallManager, encryption is supported

only if combined with authentication. This limitation applies to both protocols, TLS and SRTP.
TIP It is a general rule in cryptography to always complement packet encryption with packet
authentication. If encryption is used without authentication, the receiver of an encrypted packet
has no guarantee that the packet comes from the expected source. Assuming that an attacker does
not know the key to be used for the encryption, the attacker might not be able to send valid data
but could send arbitrary data to keep the receiver busy with decrypting the packets. Because this
decryption performed at the receiver can cause considerable processing overhead, an attacker
could launch a DoS attack just by flooding a system with packets that will be decrypted by the
receiver. In some situations, the attacker could even inject incorrect data into the application.
This is possible when the sent data does not have any special format but when any bit patterns
are considered to be valid data and are accepted by the receiver. An example is encrypted
digitized voice samples. An example where the receiver can detect invalid data is the transfer of
an encrypted Microsoft Word file. In this case, after decrypting the received arbitrary data (or a
valid file that has been encrypted with an incorrect key), the receiver would not recognize the
file as a valid Word document.
TLS AES Encryption

If an IP phone is configured for encryption, it will not only create a SHA-1 key but also an AES
key after the two-way authentication in TLS. The IP phone encrypts both keys using the public
RSA key of the server and then sends them to the server. The server decrypts the message so that
the IP phone and the server can exchange signaling messages over authenticated and encrypted
TLS packets. In Cisco IP telephony, TLS encryption requires TLS authentication so that the
authenticity of the encrypted TLS packets is always guaranteed.
SRTP Media Encryption

Media streams are encrypted by using SRTP. Cisco CallManager generates the SRTP session keys
(for media authentication and media encryption) and sends them to the IP phones inside signaling
messages. If the signaling messages are not protected, an attacker could easily learn the SRTP keys
just by sniffing the signaling messages. To ensure protection of the key distribution, encrypted
signaling is mandatory in Cisco CallManager when media streams are encrypted.
As stated before, in Cisco CallManager, encrypted packets always have to be signed to ensure the
authenticity of the source and the content of the packet.
To summarize, these rules apply to Cisco CallManager authentication and encryption:
■ Signaling encryption requires signaling authentication.

■ Media encryption requires media authentication and signaling encryption (hence also
signaling authentication).
■ Media authentication requires media encryption.
■ Signaling encryption requires media encryption.
As a consequence of these rules, you can configure one of the following secure operation modes
in Cisco CallManager:
■ Authenticated—This mode provides authenticated signaling only (TLS SHA-1).
■ Encrypted—This mode provides authenticated and encrypted signaling (TLS SHA-1 and
TLS AES) and authenticated and encrypted media transfer (SRTP SHA-1 and SRTP AES).
SRTP Packet Format

As shown in Figure 26-14, the SRTP packet header does not differ from an RTP packet header.
The RTP payload differs only in the sense that it is not clear text voice but encrypted voice. In
addition to the encrypted payload, a 32-bit SHA-1 authentication tag is added to the packet. The
authentication tag holds the first 32 bits of the 160-bit SHA-1 hash digest computed from the RTP
header and the encrypted voice payload (“truncated fingerprint”). As you can see from the figure,
the RTP packet header and the RTP payload (encrypted voice) are also authenticated. Therefore,
RTP encryption is performed before RTP authentication. RTP Header Compression (cRTP) is
supported in conjunction with SRTP.
Figure 26-14 SRTP Packet Format
V P X CC M PT Sequence Number
Time Stamp
Synchronization Source (SSRC) Identifier
Contributing Sources (CSRC) Identifier

…
RTP Extension (Optional)
RTP Payload
SHA-1 Authentication Tag (Truncated Fingerprint)
Encrypted Data Authenticated Data

Summary 591
Secure Call Flow Summary

When a call is placed between two IP phones when encryption is enabled, the following sequence
occurs:
1. The IP phones and Cisco CallManager exchange certificates.

2. The IP phones and Cisco CallManager authenticate each other by requesting some random
data to be signed. When this process is finished, Cisco CallManager and the IP phones know
that the other devices are authentic.
3. Each IP phone creates TLS session keys. One key will be used for TLS SHA-1 authentication;
the other key will be used for TLS AES encryption.
4. Each IP phone encrypts the generated keys with the public key of the Cisco CallManager and
sends the encrypted keys to Cisco CallManager.
5. Now each IP phone shares its session keys with Cisco CallManager. At this stage, each phone
can exchange signaling messages with Cisco CallManager over an authenticated and
encrypted TLS session.
6. When the call is established between the two IP phones, Cisco CallManager creates SRTP
session keys. One key is used for SRTP SHA-1 authentication; the other key is used for SRTP
AES encryption.
7. Cisco CallManager sends the generated SRTP session keys to both IP phones over the secured
TLS session.
8. The IP phones now share the session keys for authenticating and encrypting their RTP
packets. At this stage, the two IP phones can start secure media exchange.
Summary
This chapter discussed the concepts behind securing the IP telephony network. The threats targeting
Cisco IP telephony include eavesdropping, IP phone image and configuration file tampering, and
DoS attacks. To combat this, Cisco IP telephony uses authentication and encryption techniques.
Because there is no single PKI topology in Cisco IP telephony, the CTL client acts as a trusted
introducer for the different PKI systems. The IP phones can use preinstalled certificates from Cisco
(called MICs) or use LSCs issued by the CallManager CAPF or a company CA.
Cisco CallManager stores all self-signed certificates in the operating system private key storage.
The keys used by the CTL client are stored on security tokens, and the keys used with IP phones
are stored in protected nonvolatile memory.
Cisco CallManager supports device authentication, authenticated signaling (using TLS SHA-1),
and authenticated media (using SRTP SHA-1). It also supports encryption of signaling messages
using TLS AES and encryption of media using SRTP AES.
Review Questions
1. Which of the following are not security threats to an IP telephony system? (Choose two.)
a. Loss of privacy
b. Impersonation
c. Integrity
d. Loss of integrity
e. Loss of control
f. DoS
2. Which of the following represent correct mappings of application—protocol—security
features? (Choose two.)
a. Secure signaling—SRTP—device authentication, integrity
b. Secure signaling—TLS—device authentication, integrity, privacy
c. Secure media—SRTP—privacy, confidentiality, security
d. Secure media—TLS—privacy, confidentiality, security
e. Secure media—TLS—privacy, integrity
f. Secure media—SRTP—privacy, integrity
3. Which two statements about trusted introducing are incorrect?
a. The trusted introducer has to be trusted by all other members of the system.
b. The trusted introducer has to trust all other members of the system.
c. The trusted introducer guarantees the authenticity of entities it is introducing to others.
d. Only the trusted introducer has to trust the root of the system.
e. The trusted introducer is the root of a system.
f. Any entity of the system can guarantee the authenticity of any other member.
4. Which two statements about PKI topologies in Cisco IP telephony are true?
a. MICs are self-signed by the IP phone.
b. Cisco IP Phone 7940, 7960, and 7970 models can have MICs and LSCs.
c. The CAPF has a self-signed certificate.
d. Only Cisco IP Phone 7940, 7960, and 7970 (and subsequent) models can have LSCs.
e. The CTL is signed by the Cisco manufacturing CA.
f. MICs are signed by CAPF.
5. Which are the two valid options to secure enrollment in a PKI?
a. Perform the enrollment from a trusted device only.
b. Perform the enrollment in both directions.
c. Perform the enrollment over a trusted network.
d. Use self-signed certificates on all devices.
e. Do not send the private key in the enrollment.
f. Perform mutual out-of-band authentication between the PKI user and CA.
6. Which statement about enrollment in the IP telephony PKI is true?
a. MICs are issued by CAPF itself or by an external CA.
b. LSCs are issued by the Cisco CTL client or by CAPF.
c. CAPF enrollment supports the use of authentication strings.
d. CAPF itself has to enroll with the Cisco CTL client.
e. Enrollment of IP phones occurs automatically if the cluster is in secure-only mode.
f. LSCs can be issued by an external CA when using the CTL client as a proxy.
7. Which of the following entities uses a smart token for key storage?
a. CTL
b. CTL client
c. CAPF in proxy mode
d. CAPF in CA mode
e. Cisco IP Phone 7940 and 7960
f. Cisco IP Phone 7970
8. What are the authentication features of TLS in Cisco IP telephony?

a. Two-way device authentication
b. Two-way device authentication and signed media messages
c. One-way device authentication and signed signaling message
d. Two-way device authentication and signed signaling messages
e. One-way device authentication and signed media messages
f. Signed signaling messages
9. During an encrypted call between two IP phones, which two of the following does not
happen?
a. Mutual certificate exchange between Cisco CallManager and each IP phone
b. Mutual certificate exchange between the IP phones
c. SRTP packet authentication and encryption
d. Encrypted transmission of SRTP session keys between the IP phones
e. TLS packet authentication and encryption
f. Encrypted transmission of TLS session keys between Cisco CallManager and the IP
phones
10. Which is the most accurate list of tasks required to configure a Cisco CallManager cluster for
security?
a. Enable services, set cluster to mixed mode, create a signed CTL, and deploy certificates
to the IP phones.
b. Enable services, set cluster to secure-only mode, create a signed CTL, and deploy certif-
icates to the IP phones.
c. Enable extended services, set cluster to authenticated or encrypted mode, create a
signed CTL, and deploy certificates to the IP phones.
d. Disable extended services, set cluster to mixed mode, create a signed CTL, and deploy
certificates to the IP phones.
e. Enable services, set cluster to mixed mode, create a signed CTL, deploy certificates to
the IP phones, and set the device security mode.
f. Run the auto-secure feature.
following topics:
■ Identifying the steps to configure a Cisco

CallManager system for authentication and
encryption
■ Activating the Cisco CTL Provider service
and the CAPF service in Cisco CallManager
Serviceability
■ Installing the Cisco CTL client on a Windows
2000 server or workstation with a USB port
■ Configuring the Cisco CTL client to create a
CTL file and setting the cluster security mode
■ Using the CAPF settings in the Phone
Configuration window to install, upgrade,
delete, and troubleshoot LSCs
■ Configuring the default security mode for
supported IP Phone models and the device
security mode for a single device
■ Generating a CAPF report in Cisco
CallManager Administration to view
authentication strings and authentication
modes
■ Finding IP Phones in the network that support
authentication, support encryption, or use
CAPF for LSC operations
CHAPTER 27
Configuring Cisco IP Telephony
Authentication and Encryption
Cisco IP telephony systems are subject to several threats, including eavesdropping, identity
spoofing, and denial of service (DoS) attacks. In Cisco CallManager Release 4.0 and later, you
can secure the Cisco IP telephony solution against these threats by enabling authentication and
encryption features. This chapter explains how to configure the authentication and encryption
that you can apply in a Cisco IP telephony environment.
Authentication and Encryption Configuration Overview

Cisco CallManager Release 4.0 and later releases support authentication and encryption in a
Cisco CallManager cluster. By using these features, you can secure the following
communication methods:
■ Signaling messages between a supported Cisco IP Phone and Cisco CallManager—

Cisco IP Phone 7970, 7960, and 7940 models can be configured to use Transport Layer
Security (TLS) for authenticated and encrypted signaling.
■ Media exchange between two supported IP Phones within a Cisco CallManager

cluster—Cisco IP Phone 7970, 7960, and 7940 models can be configured to use Secure
Real-Time Transport Protocol (SRTP) for authenticated and encrypted media exchange.
Secure media exchange was introduced for the 7970 in Cisco CallManager 4.0. Support for
the additional IP Phones was added in CallManager 4.1.
NOTE Cisco CallManager-to-Cisco CallManager intercluster communication is not

secured. If two Cisco IP Phones are configured to use SRTP and are registered to different
Cisco CallManager servers within the cluster, there is a security risk because the SRTP
session keys need to be exchanged between the Cisco CallManager nodes (in cleartext).
Therefore, if the communication paths between Cisco CallManager nodes within a cluster are
not trusted, the recommendation is to use IPsec between the Cisco CallManager nodes.
598 Chapter 27: Configuring Cisco IP Telephony Authentication and Encryption
■ Media exchange between a supported Cisco IP Phone and a supported Media Gateway
Control Protocol (MGCP) and H.323 gateways—Cisco IP Phone 7970, 7960, and 7940
models and Cisco IOS MGCP gateways (running Cisco IOS Software Release 12.3(11)T2 or
later) can be configured to use SRTP for authenticated and encrypted media exchange. H.323
support for SRTP was added in Cisco IOS Software Release 12.4(6T).
NOTE When using SRTP with an MGCP gateway, the SRTP session keys by default are
exchanged in cleartext between Cisco CallManager and the MGCP gateway. Therefore, if the
communication path between Cisco CallManager and the MGCP gateway is not trusted, the
recommendation is to use IPsec between Cisco CallManager and the MGCP gateway.
■ Signaling messages between a supported IP Phone and a supported Cisco Survivable

Remote Site Telephony (SRST) device—Cisco IP Phone 7970, 7960, and 7940 models and
Cisco IOS SRST Version 3.3 or later devices (running Cisco IOS Software Release 12.3(14)T
or later) can be configured to use TLS for authenticated and encrypted signaling.
NOTE The Cisco SRST device can also provide SRTP session keys to the Cisco IP Phones so
that the IP Phones that are in fallback mode can still use both signaling message and media
exchange protection.
With the current release of Cisco CallManager, authenticated and encrypted calls are not possible
in any other situation than listed, including the following:
■ Calls to other Cisco CallManager clusters using intercluster trunks—Secure signaling

and media exchange are supported only for calls within a Cisco CallManager cluster;
intercluster trunk calls are not supported.
■ Calls that are connected to any media resources, such as conferences, transcoders, or
music on hold (MoH)—Secure media exchange is supported only between supported
endpoints (Cisco IP Phones and Cisco IOS MGCP gateways); conference bridges,
transcoders, or MOH servers are not supported endpoints.
To enable authentication and encryption support in your Cisco CallManager cluster, you need to
complete these tasks:
Step 1 Enable security services—You need to enable the Cisco Certificate Trust
List (CTL) Provider service and the Cisco Certificate Authority Proxy
Function (CAPF) service.
Step 2 Use the Cisco CTL client to activate security options—You need to
configure mixed mode and create a signed CTL.
Enabling Services Required for Security 599
Step 3 Configure devices for security—IP Phones need to have certificates

(either manufacturing installed certificates [MICs] or locally significant
certificates [LSCs]), they have to be configured for a security mode
(authenticated or encrypted), and the CAPF parameters have to be set if
LSCs are used.
Enabling Services Required for Security

When enabling security in your Cisco CallManager cluster, you have to activate these services:
■ Cisco CTL Provider—This service has to be activated on all Cisco CallManager servers and
Cisco TFTP servers of your cluster.
■ Cisco Certificate Authority Proxy Function—This service has to be activated on the

publisher server.
Activate Cisco CallManager services from the Cisco CallManager Serviceability Service
Activation window, shown in Figure 27-1.
Figure 27-1 Enabling Services Required for Security

Using the CTL Client

The Cisco CTL client software, available as a plug-in application on Cisco CallManager
Administration, is used to create or update the Certificate Trust List (CTL). The CTL is a list of
the trusted certificates in the CallManager cluster. When the list is accurate, the Cisco CTL client
will ensure that the CTL is signed by the keys of the Cisco CTL client. These keys are stored on
an external Universal Serial Bus (USB) device—the security token. When the CTL needs to be
signed, the Cisco CTL client passes the CTL to the security token, and the security token signs it
and then returns the signed CTL to the Cisco CTL client application. The Cisco CTL client is
needed in these situations:
■ For the initial activation of security in your cluster
■ For the deactivation or reactivation of security in your cluster
■ After modifying Cisco CallManager or Cisco TFTP server configuration (which includes
adding, removing, renaming, or restoring a server or changing the IP address or hostname of
a server)
■ After adding or removing a security token (due to theft or loss)
■ After replacing or restoring a Cisco CallManager or Cisco TFTP server
In all the situations listed, the Cisco CTL client creates a new CTL and signs it by using a security
token. The Cisco IP Phones load the new CTL and are then aware of the changes to the IP
telephony system. Any changes that are not reflected in the CTL (for instance, if you change the
IP address of a server but do not create a new CTL using the Cisco CTL client application) cause
the Cisco IP Phones to treat the corresponding device as untrusted. From this perspective, the CTL
can be seen as the certificate root store of your browser (listing all trusted certificate-issuing
entities). If any device that was previously trusted is not trustworthy anymore (for instance, when
a security token is lost), there is no need for a certificate revocation list (CRL). Instead, you will
use the Cisco CTL client and update the CRL by removing the untrusted entry (for instance, a lost
security token) from the list.
Installing the CTL Client

The Cisco CTL client application can be installed on any PC running Microsoft Windows 2000 or
XP Workstation or Microsoft Windows 2000 or 2003 Server, as long as the PC has at least one
Universal Serial Bus (USB) port. This device can be any Cisco CallManager server in your cluster
or any client PC.
The Cisco CTL client application is installed from the Cisco CallManager Administration Install
Plugins window. You can accomplish the installation just by walking through a simple wizard, as
shown in Figure 27-2. During installation, you are prompted for the destination folder; you can set
any directory of your choice or simply accept the default.
Using the CTL Client 601
Figure 27-2 Installing the CTL Client
The Smart Card service has to be activated on the PC. To activate the Smart Card service under
Microsoft Windows 2000, choose Start > Settings > Control Panel > Administrative Tools >
Services to launch the Microsoft services administration tool. Then use the tool to verify the status
of the Smart Card service. The service should have the startup type of Automatic and the Current
Status should be Running.
After you have installed the CTL Client, you can access it from the icon automatically placed on
your desktop. Initially, it will ask for the CallManager server information for the cluster, as shown
in Figure 27-3.
Figure 27-3 Configuring the CTL Client

After entering the CallManager server information and successfully authenticating, you can either
set the cluster security mode or update the CTL file. A Cisco CallManager cluster supports two
security modes:
■ Mixed mode—This mode allows secure calls between two security-enabled devices and
allows nonsecure calls between devices where at least one of the devices is not security-
enabled.
■ Nonsecure mode—This is the default configuration, in which all calls are nonsecure.
NOTE There is no secure-only mode. This setting would prevent Cisco IP Phones without
security enabled from placing calls. Many Cisco IP Phones do not support security features and
would not be able to operate in a secure-only environment.
In addition to setting the cluster security mode, you use the Cisco CTL client to update the CTL
file. This update is needed after adding or removing components, such as servers or security
tokens. After changing the list of CTL entries, you need to sign the new CTL using a security
token.
Working with Locally Significant Certificates

Cisco IP Phone 7940 and 7960 models do not have MICs; they only work with LSCs. The Cisco
IP Phone 7970 can use either MICs or LSCs. If an LSC is installed in a Cisco IP Phone 7970, the
LSC has higher priority than the MIC.
CallManager uses the CAPF to issue LSCs. CAPF can act as a Certificate Authority (CA) itself,
signing the LSCs, or it can act as a proxy to an external CA, having the external CA signing the
LSCs. You can configure the CAPF service at the CAPF service parameter web page shown in
Figure 27-4. To access this page, choose Cisco CallManager Administration > Service >
Service Parameter > Cisco Certificate Authority Proxy Function.
You can set the certificate issuer (CAPF itself or an external CA) and IP address of the external
CA (if used). You can also modify some default values, such as the Rivest, Shamir, and Adleman
(RSA) key size or the certificate lifetime.
When you want to install or upgrade LSCs for Cisco IP Phones that you are configuring, use the
relevant CAPF settings at the Phone Configuration window by choosing Cisco CallManager
Administration > Device > Phone. All possible settings are found in the Certificate Authority
Proxy Function (CAPF) Information area.
Working with Locally Significant Certificates 603
Figure 27-4 Working with Locally Significant Certificates
There are four operations options in the Certificate Operation field (as shown in Figure 27-5):
■ Install/Upgrade—This operation allows the installation of an LSC (if the IP Phone does not
already have an LSC) and the upgrade (replacement) of an existing LSC (if the IP Phone
already has an LSC).
■ Delete—This operation allows the removal of an existing LSC from a Cisco IP Phone.
■ Troubleshoot—This operation retrieves all existing IP Phone certificates from the IP Phone
and stores them in CAPF trace files. There are separate CAPF trace files for MICs and for
LSCs. The CAPF trace files are located in C:\Program Files\Cisco\Trace\CAPF.
■ No Pending Operation—This is the default value. You can also change back to this value
when you want to cancel a previously configured operation that has not yet been executed.
Figure 27-5 Selecting a Certificate Operation
In the Authentication Mode field (as shown in Figure 27-6), you can choose one of four possible
authentication modes:
■ By Authentication String—This authentication mode is the default and requires the Cisco
IP Phone user to manually initiate the installation of an LSC. The user must authenticate to
Cisco CallManager by the authentication string that has been set by the administrator in the
Authentication String field. To enable the user to enter the correct authentication string, the
administrator has to communicate the configured authentication string to the user.
■ By Null String—This authentication mode disables Cisco IP Phone authentication for the
download of the IP Phone certificate (enrollment). The enrollment of the IP Phone should be
done over a trusted network only when this setting is used. Because no user intervention is
needed, the enrollment is done automatically the next time the Cisco IP Phone boots or is
reset.
■ By Existing Certificate (Precedence to LSC)—This authentication mode uses an existing

certificate (with precedence to the LSC if both a MIC and an LSC are present in the IP Phone)
for IP Phone authentication. Because no user intervention is needed, the enrollment is done
automatically the next time that the IP Phone boots or is reset.
Working with Locally Significant Certificates 605
■ By Existing Certificate (Precedence to MIC)—This authentication mode uses an existing

certificate (with precedence to MIC if both a MIC and an LSC are present in the IP Phone)
for IP Phone authentication. Because no user intervention is needed, the enrollment is done
automatically the next time that the IP Phone boots or is reset.
Figure 27-6 Selecting the IP Phone Authentication Method
NOTE Some authentication options will only appear under specific phone models. For
example, the “By Existing Certificate (Precedence to MIC)” option is unavailable on older Cisco
IP Phones such as the 7940 and 7960.
Issuing a Phone Certificate Using an Authentication String

Figure 27-7 illustrates an example for a first-time installation of a certificate with a manually
entered authentication string. For such a scenario, set the Certificate Operation field to Install/
Upgrade and the Authentication Mode to By Authentication String. You can manually enter a
string of four to ten digits, or click the Generate String button to create an authentication string
(and populate the Authentication String field). After you click Update and reset the IP Phone,
the IP Phone is ready for enrollment. However, enrollment is not automatically triggered; it has to
be initiated by the user (from the Settings menu of the Cisco IP Phone).
Figure 27-7 Issuing a Phone Certificate Using an Authentication String
NOTE The Settings menu can also be used to gain information about the IP telephony system
or remove the CTL. Usually, you do not want IP Phone users to have access to such options, and,
therefore, access to the settings on the IP Phone is often restricted or disabled. LSC enrollment
with authentication by authentication string is not possible if settings access is not (fully)
enabled. If access to settings is restricted or disabled, you have to enable it for the enrollment
and then return it to its previous value.
When a user starts the enrollment procedure, the user has to enter the authentication string
configured, and if the process is successful, the certificate is issued to the IP Phone.
On a Cisco IP Phone 7940, the user would complete these steps:
Step 1 Press the Settings button to access the Settings menu.

Step 2 Scroll to the Security Configuration option and press the Select softkey to
display the Security Configuration menu.
Step 3 Press **# to unlock the IP Phone configuration.
Configuring the Device Security Mode 607
Step 4 Scroll to LSC and press the Update softkey to start the enrollment.
Step 5 Enter the authentication string and press the Submit softkey to authenticate
the IP Phone to the CAPF when prompted to do so.
Step 6 The IP Phone generates its RSA keys and requests a certificate signed by
the CAPF. When the signed certificate is installed, the message “Success”
appears at the lower-left corner of the Cisco IP Phone display.
Issuing a Phone Certificate Using the CAPF

You might use the CallManager CAPF for a certificate upgrade using an existing LSC to
authenticate the communication. A reason for such an upgrade could be that an LSC will soon
reach its expiration date. By issuing a new LSC shortly before the expiration of the existing LSC,
the IP Phone can use the existing LSC for the upgrade (which avoids entering a manual
authentication string at the IP Phone).
For such a scenario, set the Certificate Operation field to Install/Upgrade and the
Authentication Mode to By Existing Certificate (Precedence to LSC). After you click Update
and reset the Cisco IP Phone, the IP Phone automatically contacts the CAPF for the download of
the new certificate. The existing certificate is used to authenticate the new enrollment, and there is
no need for a manually entered authentication string.
Configuring the Device Security Mode

After you have configured the Cisco CallManager for mixed mode using the CTL Client and the
Cisco IP Phones have certificates, you must configure the IP Phones to support authenticated or
encrypted calls. You can use the device security mode to configure a Cisco IP Phone for one of
three security modes:
■ Non Secure—The IP Phone will not support authenticated or encrypted calls.
■ Authenticated—The IP Phone will support authenticated calls.
■ Encrypted—The IP Phone will support encrypted calls.
The default device security mode is configured in the Cisco CallManager Enterprise Parameters
window; choose Cisco CallManager Administration > System > Enterprise Parameters. The
default mode is Non Secure.
In addition to setting the default value, you can configure each individual IP Phone with the device
security mode. Choose Cisco CallManager Administration > Device > Phone to display the
Phone Configuration window, as shown in Figure 27-8. The default mode is Use System Default.
Figure 27-8 Configuring IP Phone Security Options
NOTE In several situations, you should not use cryptographic services for Cisco IP Phones at
all. With some Cisco IP Contact Center (IPCC) applications, for instance, cleartext signaling
messages or media packets have to be seen by other devices (for instance, attached PCs).
Another example is the use of Network Address Translation (NAT) or Port Address Translation
(PAT). Because the translating device has to see cleartext signaling messages to be able to
dynamically allow the negotiated UDP ports that will be used for Real-Time Transport Protocol
(RTP), encryption cannot be used.
Negotiating Device Security Mode

The actual security mode used for a call depends on the configuration of both IP Phones
participating in the call. As shown in Figure 27-9, these rules apply:
■ If either device in a conversation is set to Non Secure, a nonsecure call (that is, a call without
authentication and without encryption) is placed.
■ If both devices are set to Encrypted, an encrypted call (that is, a call with authentication and
encryption) is placed.
Generating a CAPF Report 609
■ If one device is set to authenticated and the other phone is set to either authenticated or
encrypted, the resulting call will be authenticated.
Figure 27-9 Device Security Negotiation
Phone 2
Non Secure Authenticated Encrypted
Non Secure Non Secure Non Secure Non Secure
Phone 1 Authenticated Non Secure Authenticated Authenticated
Encrypted Non Secure Authenticated Encrypted
Generating a CAPF Report

CAPF reports can be quite useful because you can use them to search for IP Phones matching
selected CAPF criteria:
■ Certificate Operation Status
NOTE The Certificate Operation Status displays the result of the last CAPF activity for each
IP Phone. Possible values include None (typically shown on IP Phones with device security
mode Non Secure), Operation Pending (when CAPF waits for a user to manually retrieve a
[new] certificate), and result information (success, failure) after upgrade, delete, or
troubleshooting operations.
■ Device Security Mode
■ Authentication Mode
■ Authentication String
You can click entries from the results list to display the configuration window of the corresponding
IP Phone. You can also save the results list of a search to a file in Comma-Separated Values (CSV)
format.
You can create CAPF reports from Cisco CallManager Administration. Figure 27-10 shows a
CAPF report where the administrator accessed the CAPF Report window by choosing Cisco
CallManager Administration > Device > Device Settings > CAPF Report and started a report
to search for all IP Phones where the authentication string matched a specific value. This allowed
the administrator to quickly locate the status for a specific IP phone.
Figure 27-10 Generating CAPF Reports
In addition to using CAPF reports to locate devices, you can use the Find and List Phones window
to search on a variety of other criteria. For example, Figure 27-11 shows a query to search for IP
Phones where the device security mode is Encrypted. Two IP Phones have been found matching
the search criteria.
Summary 611
Figure 27-11 Searching Using the Find and List Phones Query Tool
Summary
This chapter discussed the configuration of encryption and authentication in the Cisco IP
telephony network. To enable either of these features, you must activate the Cisco CTL Provider
and the Cisco CAPF services. From there, you must install the CTL Client on at least one machine
in the cluster to sign the CTL using a security token, which is manually plugged in to the USB port
of the machine. This CTL Client must be rerun each time the CTL entries change.
Cisco allows the use of LSCs on all security-enabled IP Phones. You can configure the Cisco IP
Phones for nonsecure calls, authenticated calls only, or authenticated and encrypted calls. When
the phones communicate, they will agree on a lowest-common-denominator security standard.
You can locate the phones that have successfully enabled security functions using the CAPF report
utility.
Review Questions
1. Which is the most accurate list of tasks required to configure a Cisco CallManager cluster for
security?
a. Enable services, set cluster to mixed mode, create a signed CTL, and deploy certificates
to the IP Phones
b. Enable services, set cluster to secure-only mode, create a signed CTL, and deploy certif-
icates to the IP Phones
c. Enable extended services, set cluster to authenticated or encrypted mode, create a
signed CTL, and deploy certificates to the IP Phones
d. Disable extended services, set cluster to mixed mode, create a signed CTL, and deploy
certificates to the IP Phones
e. Enable services, set cluster to mixed mode, create a signed CTL, deploy certificates to
the IP Phones, and set the device security mode
f. Run the auto-secure feature
2. Which two services must be enabled when configuring a Cisco CallManager cluster for
security?
a. Cisco CAPF
b. Cisco Authority Provider Function
c. Cisco CTL Provider
d. Cisco CTL Proxy
e. Cisco CTL Client Provider
f. Cisco Extended Functions
3. What are the minimum needs for the PC when you are installing Cisco CTL Client?
a. Windows 2000 or XP operating system, at least one USB port, Smart Card service
enabled
b. Windows XP operating system, at least two USB ports, Smart Card service enabled
c. Windows 2000 operating system, at least two USB ports, Smart Card service disabled
d. Installation on Cisco CallManager publisher server only, two available USB ports,
Smart Card service enabled
e. Windows 2000 operating system, PCMCIA slot, Smart Card service enabled
f. Windows 2000 operating system, at least one USB port, Cisco VT Advantage camera,
Smart Card service enabled
4. When is update of the configuration using the Cisco CTL client not needed?
a. When an LSC of the IP Phone is upgraded
b. When a security token is added to the system
c. When a Cisco CallManager has been removed
d. When an IP address of the Cisco TFTP server has been changed
5. Which two statements about LSCs are correct?
a. On a Cisco IP Phone 7970, a MIC has priority over an LSC.
b. On a Cisco IP Phone 7960, an LSC has priority over a MIC.
c. The CAPF issues LSCs if used as a CA and MICs if used as a proxy.
d. The certificate operation of the CAPF can be set to install, upgrade, or delete.
e. The certificate operation of the CAPF can be set to install/upgrade, delete, or trouble-
shoot.
f. CAPF authentication can be configured to be done by authentication string, null string,
or existing certificates.
6. Which two combinations of device security modes and resulting call type are correct?
a. Phone 1: Non Secure, phone 2: Authenticated; call: authenticated
b. Phone 1: Non Secure, phone 2: Encrypted; call: authenticated
c. Phone 1: Authenticated, phone 2: Non Secure; call: nonsecure
d. Phone 1: Encrypted, phone 2: Authenticated; call: authenticated
e. Phone 1: Authenticated, phone 2: Encrypted; call: encrypted
f. Phone 1: Encrypted, phone 2: Non Secure; call: encrypted
7. Which two of the following options are not search criteria when you are creating a CAPF
report?
a. Authentication mode
b. Certificate operation status
c. Device name
d. Certificate lifetime
e. Authentication string
f. Device security mode
8. Which two of the following options are not search criteria when you are searching for IP
Phones using the Find and List Phones window?
a. Authentication mode
b. LSC status
c. Device name
d. Directory number
e. Authentication string
f. Device security mode
9. Which phone model does not support phone security options?
a. Cisco 7920
b. Cisco 7970
c. Cisco 7961
d. Cisco 7960
10. Which of the following types of communication cannot be encrypted? (Choose three.)
a. Signaling messages between an IP Phone and an SRST gateway
b. Calls to other clusters using intercluster trunks
c. Calls communicating using a conference bridge
d. Calls using a transcoder
Part VI: IP Video
Chapter 28 Introducing IP Video Telephony
Chapter 29 Configuring Cisco VT Advantage

following topics:
■ Classifying the functions and components of

the Cisco IP video telephony solution
■ Characterizing a video call in terms of the
RTP stream types and codecs, and describing
audio and video bandwidth requirements
■ Comparing an H.323 video call in a Cisco
CallManager environment with an SCCP
video call
■ Describing the two main factors that
determine the bandwidth requirement for
video calls and calculating bandwidth
requirements
■ Implementing call admission control within a
cluster using the locations and regions of
Cisco CallManager Administration
■ Describing call admission control between
clusters using a gatekeeper
CHAPTER 28
Introducing IP Video Telephony
Companies that want to enable video calls can install and use products from the Cisco enterprise
video products portfolio. Video communication capabilities are integrated into Cisco
CallManager Release 4.0. These capabilities extend several video features to benefit end users,
network administrators, and enterprises as a whole. A common IP infrastructure for all
communications not only provides an enterprise with reduced cost of ownership but with a
faster return on investment (ROI) as users more readily and easily adapt to a system that can be
deployed to the desktop. This chapter introduces the benefits, terms, and concepts of video
telephony.
IP Video Telephony Solution Components

Cisco introduced its IP video telephony solution in Cisco CallManager Release 4.0. With this
solution, video is fully integrated into Cisco CallManager, and new endpoints are available from
Cisco and its strategic partners. Video is now just as easy to deploy, manage, and use as a Cisco
IP Phone.
The Cisco IP video telephony solution consists of products and solutions:
■ Cisco CallManager Release 4.0 or later. Cisco CallManager provides call routing and a
centralized dial plan for voice and video devices.
■ Cisco IP/VC 3500 Series Multipoint Control Units (MCUs) for both H.323 and Skinny
Client Control Protocol (SCCP) videoconference calls. The MCUs are responsible for
mixing the various video and voice streams in a videoconference. In a videoconference, all
devices send their streams to the MCU; the MCU mixes these streams into a single picture
and sends the mixed stream back to the endpoints.
■ Cisco IP/VC 3500 Series H.320 gateways interconnect the IP world with the ISDN world.
To interconnect the H.323 with the H.320 (ISDN) world, a video gateway is required.
Unlike a voice gateway, a video gateway is able to bond bearer channels (B channels). An
H.263 video call needs at least 128 kbps, at lowest quality, in each direction; the VT
Advantage default for a video call is an H.263 384-kbps video call. For a 384-kbps call, the
620 Chapter 28: Introducing IP Video Telephony
video gateway needs to open six ISDN B channels at the same time. The IP/VC 3500 Series
H.320 gateways support B-channel bonding. The IP/VC 3500 Series H.320 gateways require
an H.323 gatekeeper to register.
■ A Cisco IOS H.323 gatekeeper is required to register third-party H.323 devices and the 35XX
MCU. The main tasks of an H.323 gatekeeper are endpoint registration, bandwidth
management, and directory number (DN) resolution. Many video devices, such as H.323
MCUs or H.320 video gateways, require an H.323 gatekeeper for registration.
■ Cisco Video Telephony (VT) Advantage is a video telephony solution comprising the Cisco
VT Advantage software application and Cisco VT Camera, a video telephony Universal Serial
Bus (USB) camera. With the Cisco VT Camera attached to a PC collocated with a Cisco IP
Phone, users can place and receive video calls on the enterprise IP telephony network. Users
make calls from Cisco IP Phones using familiar telephone interfaces, but calls are enhanced
with video on a PC, without requiring any extra button-pushing or mouse-clicking.
■ TANDBERG SCCP endpoints are developed by the videoconference equipment vendor

TANDBERG under Cisco license.
■ The existing range of H.323-compliant products from vendors such as Polycom,

TANDBERG, Sony, VCON, and VTEL Products. (Most H.323 video devices require a
gatekeeper to register.)
■ The Cisco 7985 IP Phone is one of the newest IP Phones added to the Cisco fleet. The 7985
supports full videoconference capabilities out of the box.
■ Cisco IP Communicator 2.0 and later allows you to integrate the VT Advantage solution with
the on-screen IP Communicator softphone.
Cisco CallManager Release 4.0 and later adds support for video in both the SCCP and H.323
protocols. Cisco CallManager can now manage H.323 endpoints, MCUs, and gateways, providing
the system administrator with PBX-style control over all call routing and bandwidth management
for those devices.
The video telephony solution consists of end-user devices, infrastructure, and applications. The
end-user devices include video phones, soft video phones for remote workers (such as Cisco IP
Communicator 2.0), and desktop environments that incorporate existing telephones and desktop
computers. Additionally, the existing H.323 endpoints that the customers already own will become
part of the video telephony environment.
Cisco CallManager is the center of the IP telephony infrastructure and supports both voice and
video telephony. Cisco CallManager provides a single dial plan for all endpoints; there is no need
for an additional video dial plan. The same applications are still supported: voice mail,
conferencing, and scheduling for audio and video resources are possible.
The video stream mixing is accomplished by the MCU. In a Cisco CallManager environment, the
MCU can be SCCP or H.323 or SCCP- and H.323-controlled. The main difference is that the
H.323-controlled MCU needs an H.323 gatekeeper to register.
To allow external H.320 parties to participate in a videoconference, a video gateway, such as the
Cisco IP/VC 3521 BRI Videoconferencing Gateway or IP/VC 3526 PRI Videoconferencing
Gateway, is required. The normal voice gateways used in the Cisco CallManager environment do
not support ISDN B-channel bonding and cannot route video calls from and to the public switched
telephone network (PSTN).
Video Call Concepts

Using video endpoints in the Cisco CallManager system is, from the perspective of a user, as easy
as placing a telephone call. Video terminals can be either H.323- or SCCP-controlled and are able
to communicate with each other regardless of the protocol controlling the device. All video
devices use the same dial plan as all other devices in the Cisco CallManager system. For Cisco
CallManager, video devices are treated like voice devices, with the same configuration options.
For the Cisco VT Camera, the same DN as the associated IP Phone is used. There is no additional
number to remember to be able to reach someone over the video system. The video signaling is
transparent to the user, and the devices negotiate their video capabilities, such as codec, format,
and bit rate used for the video call, with no additional user action.
Cisco CallManager Involvement in Video Calls

A video telephony call can be established between any two video-enabled endpoints. If one of the
endpoints is not video-enabled, the call is set up as a normal voice call. When both endpoints are
video-enabled, Cisco CallManager signals the voice and video capabilities to both endpoints, as
shown in Figure 28-1, and the endpoints directly build two RTP streams, one for voice and one for
video. Call control for video calls operates the same way as the call control that governs voice
calls.
Figure 28-1 CallManager Video Involvement
Video Call Setup
Voice RTP Stream

User A User B
DN: 1234 Video RTP Stream DN: 4567
TIP The term “video call” is sometimes confused with the term “videoconferencing.” A
videoconference is a video call with at least three participants.
For videoconferences in the Cisco CallManager system, extra hardware is required. The device
that mixes the video streams is an MCU.
Video Call Flow

The typical video call includes two RTP streams in each direction. A basic video call includes two
unidirectional RTP streams for voice and two unidirectional RTP streams for video. The call can
include these stream types:
■ Audio—These are the same codecs as used in audio-only calls, G.711 and G.729, plus
additional codecs, G.722 and G.728. These additional codecs can be used when
communicating to third-party equipment. Because Cisco IP Phones do not use these codecs,
transcoding is necessary.
■ Video—The video codecs used are H.261, H.263, H.264, and Cisco Wideband codecs. The
video codec is a software module that enables the use of compression for digital video. There
is a complex balance between the quality of the video, the video call bit rate, the complexity
of the encoding and decoding algorithms, the robustness to data losses and errors, the ease of
editing, the random access, the state of the art of compression algorithm design, the end-to-
end delay, and a number of other factors.
■ Far-end camera control (FECC)—FECC is used only for H.323 devices and is optional.
FECC enables a user to control the camera of the far side during an active video call. This
feature must be supported by both video-enabled H.323 devices. When FECC is used, two
more unidirectional RTP streams are sent between the video devices. The two additional RTP
streams are used for the camera control parameters. This functionality is typically used for
security cameras.
In the example shown in Figure 28-2, the video-enabled endpoints report their video and audio
capabilities to Cisco CallManager. Cisco CallManager now treats the endpoints simply as video
phone devices. When a call is placed or received between two video-enabled devices, Cisco
CallManager signals for both audio and video streams. First the audio capabilities, such as audio
codec information and audio bit rate, are signaled and negotiated, and then the audio stream is set
up. After the audio stream is set up, the video capabilities, such as video codec information and
video channel bit rate, are negotiated, and the devices exchange their video streams separately
from the audio. In this example, the video call has four RTP streams—two unidirectional RTP
streams for voice and two unidirectional RTP streams for video transmission.
Figure 28-2 Video Call Flows
Audio + Video Capabilities
Audio + Video Capabilities
Request Call
Answer Call
Audio Capability Negotiation
Audio Streaming
Video Capability Negotiation
Video Streaming
Video Codecs Supported by Cisco CallManager

Cisco CallManager supports several standard video codecs and a Cisco proprietary video codec.
H.263 is a video codec specified by the International Telecommunication Union
Telecommunication Standardization Sector (ITU-T) as a low-bit-rate encoding solution for
videoconferencing. It was first designed to be used in H.324-based systems (PSTN and other
circuit-switched network video environments) but has since found use in these other solutions as
well:
■ H.323 (IP-based videoconferencing)
■ H.320 (ISDN-based videoconferencing)

■ Real-Time Streaming Protocol (RTSP)
■ Session Initiation Protocol (SIP)
H.263 was developed as an evolutionary improvement based on experience with H.261, the
previous ITU-T standard for video compression, and the Moving Picture Experts Group-1
(MPEG-1) and Moving Picture Experts Group-2 (MPEG-2) standards. The first version of H.263
was completed in 1995, and it provided a suitable replacement for H.261 at all bit rates. H.263 was
further enhanced in H.263 version 2 (H.263v2, also known as H.263+ or H.263 1998) and H.263
version 3 (H.263v3, also known as H.263++ or H.263 2000).
The next enhanced codec specified by the ITU-T after H.263 is the H.264 standard. Because H.264
provides a significant improvement in capability beyond H.263, the H.263 standard is now
considered primarily a legacy design (although this is a recent development). Most new
videoconferencing products include H.261, H.263, and H.264 capabilities.
The video codecs supported by Cisco CallManager Release 4.1 include H.261, H.263, and H.264.
These codecs exhibit the parameters and typical values listed in Table 28-1.
Table 28-1 Video Codec Parameters
Parameter Values
Video call speed 128 kbps, 384 kbps, 768 kbps, and 1.544 Mbps
Resolution • Common Intermediate Format (CIF); resolution of 352 x 288 pixels
• Quarter CIF (QCIF); resolution of 176 x 144 pixels
• 4CIF; resolution of 704 x 576 pixels
• Sub QCIF (SQCIF); resolution of 128 x 96 pixels
• 16CIF; resolution of 1408 x 1152 pixels
Frame rate 15 frames per second (fps)
30 fps
The Cisco Wideband codec is a proprietary codec that is a fixed-bit-rate codec and runs on a PC
that is linked to a phone. It enables the PC to associate with a call that the phone receives and can
only be used by the Cisco VT Camera.
Video Protocols Supported in Cisco CallManager

Cisco CallManager Release 4.0 added support for video in both the SCCP and H.323 protocol.
You can make calls between SCCP clients, between H.323 clients, and between SCCP and H.323
clients.
H.323 and SCCP endpoints have these characteristics:
■ H.323 devices typically register with an H.323 gatekeeper (such as the Cisco IOS gatekeeper).
The H.323 gatekeeper maintains the registration state of each endpoint, but it directs all call
requests to Cisco CallManager. SCCP devices register directly with Cisco CallManager.
■ H.323 devices send the complete dialed number all at once in their H.225 setup messages.
SCCP devices send each dialed digit one-by-one as they are entered on the keypad. The
difference is subtle but worth noting because it affects the experience of the user. Dialing on
an H.323 device is much like dialing on a cell phone—the user enters the entire number and
then presses the Call or Dial button to initiate the call. SCCP devices are more like a
traditional telephone, in which the user goes off-hook, receives a dial tone from Cisco
CallManager, and then starts dialing digits.
■ H.323 devices are configured through the user interface of each endpoint. Changes to the
configuration or software or firmware loads must be done locally on each endpoint (or, in
some cases, through Simple Network Management Protocol [SNMP] or other vendor-specific
management applications). SCCP devices are centrally controlled and configured in Cisco
CallManager Administration. The configuration and software or firmware loads are then
pushed to the endpoints via the TFTP. TANDBERG SCCP devices that receive their
configurations via TFTP are the exception; however, software or firmware upgrades must be
done manually (or through TANDBERG management applications).
■ H.323 devices advertise their capabilities to Cisco CallManager on a call-by-call basis using
H.245. SCCP devices advertise their capabilities when they register with Cisco CallManager
and whenever their capabilities change. Cisco CallManager then decides, on a call-by-call
basis, which types of media channels are negotiated between the endpoints.
■ H.323 video devices offer basic call capabilities in interaction with Cisco CallManager. SCCP
devices offer PBX-style features, such as hold, transfer, conference, park, pickup, and group
pickup.
■ For both types of endpoints, the signaling channels are routed through Cisco CallManager,
but the media streams (audio and video channels) flow directly between the endpoints using
RTP.
■ H.323 and SCCP endpoints can call one another. Cisco CallManager provides the signaling
translation between the two protocols and negotiates common media capabilities (common
codecs).
■ SCCP endpoints can invoke supplementary services, such as placing the call on hold,
transferring the call, and conferencing with another party. H.323v2 devices support receiving
Empty Capability Set (ECS) messages. ECS allows the devices to be the recipients of such
features (that is, they can be placed on hold, transferred, or conferenced), but they cannot
invoke those features.
■ H.323-to-SCCP calls are not the only types of calls allowed by Cisco CallManager. Any
device can call any other device, but video calls are supported only on SCCP and H.323
devices. Specifically, video is not supported in these protocols in Cisco CallManager Release
4.1:
— Computer telephony integration (CTI) applications (Telephony Application

Programming Interface [TAPI] and Java TAPI [JTAPI])
— Media Gateway Control Protocol (MGCP)
— SIP
Table 28-2 summarizes the key differences between H.323 and SCCP endpoints.
Table 28-2 SCCP and H.323 Comparison
SCCP Clients H.323 Clients

Register with Cisco CallManager Register with H.323 gatekeeper
Send dialed number digit-by-digit Send entire dialed number at once
Are configured in Cisco CallManager Are configured through user interface
Administration
Advertise their capabilities when Advertise their capabilities on a call-by-call basis
registering
Offer PBX-style features Offer basic call capabilities in interaction with Cisco
CallManager
SCCP Video Call Characteristics

If a call is placed from an SCCP phone that reported video capabilities to Cisco CallManager and
the other end supports video as well (another SCCP phone with video capabilities or an H.323
device), Cisco CallManager automatically signals the call as a video call. The amount of
bandwidth the call can consume is determined by your region configuration. The system does not
ask users for bit rate.
In Cisco CallManager Release 4.1, H.264 support was added for SCCP video devices. H.264
contains a number of features that allow it to compress video much more effectively than older
codecs. This capability allows H.264 to deliver better video call quality over less bandwidth. Cisco
CallManager supports the H.261 and H.263+ codecs as well. In addition, Cisco CallManager
Release 4.1 supports the audio codecs G.711, G.722, G.723.1, G.728, G.729, Cisco Wideband
codec, and Global System for Mobile Communications (GSM). The wideband video codec of
Cisco VT Camera reduces PC CPU utilization, unlike the resource-intensive H.263 and H.264
codecs. However, you will sacrifice a significant amount of bandwidth when using the wideband
codec. For this reason, you will use this codec primarily in LANs, not over WAN links.
Cisco CallManager uses out-of-band H.245 alphanumeric dual-tone multifrequency (DTMF).

DTMF might not work between SCCP and third-party H.323 devices because many H.323 devices
pass DTMF in-band.
H.323 Video Call Characteristics

Call forwarding, dial plan, and other call routing–related features work with H.323 endpoints.
During all this, the Cisco CallManager remains the central call-routing intelligence. In Cisco
CallManager, H.323 endpoints can be configured as H.323 terminals, H.323 gateways, or H.323
trunks. Many H.323 devices request a gatekeeper for registration.
Some vendors implement call setup such that they cannot increase the bandwidth of a call when
the call is transferred or redirected. In such cases, if the initial call is audio, users might not receive
video when they are transferred to a video endpoint.
Video H.323 clients that use trunks to other Cisco CallManager systems or other H.323 systems
can use the same features as audio-only H.323 clients. Currently, neither video media termination
points (MTPs) nor video transcoders exist. If an audio transcoder or MTP is required for a call,
that call will be audio only. H.323 video endpoints cannot initiate hold, resume, transfer, park, or
offer other similar features. Only if an H.323 endpoint supports ECS can the endpoint be held,
parked, and so on.
Dynamic H.323 addressing within Cisco CallManager provides a facility to register H.323 video
terminals on Cisco CallManager when the video terminal receives its IP address through a
Dynamic Host Configuration Protocol (DHCP) server. Endpoints are tracked based on their E.164
address registration with an adjacent video gatekeeper. The E.164 address is a static identifier that
remains constant from the perspective of both gatekeeper and Cisco CallManager. The feature
became available with Cisco CallManager Release 4.1. To move to a converged voice and video
dial plan, it is highly desirable that Cisco CallManager become the entity that manages call routing
and digit manipulation for both the voice and video endpoints, regardless of call-signaling
protocol. Earlier releases of Cisco CallManager required that an H.323 video terminal be
configured on Cisco CallManager based on static IP address information. As a support and
mobility issue, this design could not facilitate a scalable method for endpoint management
because configuration information was accurate only as long as the DHCP lease did not expire
with the endpoint in question.
Cisco CallManager Release 4.1 supports H.261 and H.263 when using H.323-based video
endpoints; H.264 is supported only for SCCP endpoints. H.323 video clients support the voice
codecs G.711, G.722, G.723.1, G.728, and G.729. H.323 endpoints can also support far-end
camera control (FECC). FECC enables a user to control the camera of the far side during an active
video call. For FECC, two separate, unidirectional RTP streams are set up. This feature was
introduced in H.323 version 5 (H.323v5) as Annex Q.
Cisco CallManager uses out-of-band H.245 alphanumeric DTMF. DTMF might not work because
many H.323 devices pass DTMF in-band. If an H.323 device uses in-band DTMF signaling, Cisco
CallManager will not convert it to out-of-band signaling, and the DTMF signaling will fail. Both
sides need to use a common scheme; either both devices need to use out-of-band signaling or the
H.323 device needs to support both methods and autodetect the method it should use.
SCCP and H.323 in Cisco CallManager

The call-related features of the Cisco IP video telephony integration strongly depend on the
protocol in use. FECC, for example, is an H.323 feature supported only by H.323 devices and is
not available on SCCP devices. Videoconferencing with three or more parties on a call requires an
MCU. The Cisco IP/VC 3540 MCU supports both protocol stacks: H.323 and SCCP. This can be
configured in the MCU Administration window.
The Cisco IP/VC 3521 and 3526 videoconferencing gateways bridge the gap between the installed
base of ISDN videoconferencing group and room systems and IP-based H.323 systems. The
gateways connect H.320 video systems on ISDN to H.323 systems on IP by translating calls
initiated from the PSTN to their equivalent on the packet network, and vice versa. To enable SCCP
to call H.320 endpoints, an H.323-based video gateway is necessary as well.
Beginning in Cisco CallManager Release 4.1, mid-call video is supported. Mid-call video allows
an active voice call to become a video call if the video capabilities are added during the active call
(for instance, if the Cisco VT Advantage software is turned on). Cisco VT Advantage will then
associate with the phone and try to set up a video stream. If both parties are SCCP video endpoints,
the call immediately becomes a video call. If the other party is an H.323 endpoint, the SCCP
endpoint tries to request a video channel. If the H.323 endpoint rejects the incoming channel or
does not open a channel, the call becomes either one-way video or audio only.
Bandwidth Management
Bandwidth management with Cisco CallManager call admission control enables you to control the
audio and video quality of calls over a WAN link by limiting the number of calls that are allowed
on that link simultaneously. In a packet-switched network, audio and video quality can begin to
degrade when a link carries too many active calls and the bandwidth is oversubscribed. Call
admission control regulates audio and video quality by limiting the number of calls that can be
active on a particular link at the same time. Call admission control does not guarantee a particular
level of audio or video quality on the link (this is the job of QoS), but it does allow you to regulate
the amount of bandwidth that active calls on the link consume.
The actual bandwidth required is more than just the speed of the video call. The speed of the video
call is just the payload, but the final packet also includes some amount of overhead for header
information that encapsulates the payload into RTP segments, User Datagram Protocol (UDP)
frames, IP packets, and finally a Layer 2 transport medium (such as Ethernet frames, ATM cells,
or Frame Relay frames). References to the video call bandwidth should include the sum of the
video call speed and all packetization overhead (RTP, UDP, IP, and Layer 2).
Video Call Bandwidth Requirement

A typical video call consists of two media channels: one for the video stream and one for the audio
stream. These channels are referred to as logical channels in the H.323 protocol, and each logical
channel is negotiated separately. For the call to succeed, Cisco CallManager checks that audio is
successfully signaled; if video cannot be negotiated, the call will be an audio-only call. The audio
channel consists of the actual audio bit rate. This bit rate is dictated by the audio codec in use. In
the case of a G.711 codec, the bit rate is 64 kbps, whereas in the case of a G.729 codec, it is 8 kbps.
For an audio channel, only the pure audio data is considered, not the packetization overhead.
The bit rate available for the video channel depends on the negotiated audio codec and the video
call speed. The video channel bit rate is the speed of the video call minus the bit rate of the codec
used for the audio channel. In the case of a 384-kbps video call with an audio channel that uses
G.711, the bit rate left for the video channel is 320 kbps (384 kbps minus 64 kbps). In the case of
an audio channel using the G.729 codec, the payload of the same video call (384 kbps) leaves 376
kbps. Table 28-3 lists possible video call speeds, their possible audio channel codecs, and the
associated video channel bit rates.
Table 28-3 Video Call Speeds and the Associated Audio and Video Codecs
Video Call Speed Audio Codec and Rate Video Codec and Rate
128 kbps G.711 at 64 kbps H.261 or H.263 at 64 kbps
1.472 Mbps G.729 at 8 kbps H.261 or H.263 at 1.464 Mbps
continues
Table 28-3 Video Call Speeds and the Associated Audio and Video Codecs (Continued)
Video Call Speed Audio Codec and Rate Video Codec and Rate
1.472 Mbps G.711 at 64 kbps H.261 or H.263 at 1.408 Mbps
7 Mbps G.729 at 8 kbps Wideband at 7 Mbps (minus 8 kbps for
the audio stream)
7 Mbps G.711 at 64 kbps Wideband at 7 Mbps (minus 64 kbps
for the audio stream)
For example, as shown in Figure 28-3, a 384-kbps video call may be G.711 at 64 kbps (for audio)
plus 320 kbps (for video). If the audio codec for a video call is G.729 (at 8 kbps), the video rate
increases to maintain a total bandwidth of 384 kbps. If the call involves an H.323 endpoint, the
H.323 endpoint might use less than the total video bandwidth that is available. An H.323 endpoint
can always choose to send at less than the maximum bit rate for the call.
Figure 28-3 Video Call Bandwidth Requirement Examples

Video Call Speed 384 kbps with G.711
320 kbps 64 kbps
Video Audio
Video Call Speed 384 kbps with G.729
376 kbps 8 kbps
NOTE None of these values include packetization overhead.
Calculating the Total Bandwidth

The two main factors that influence bandwidth requirements for video calls are the media channels
and the bandwidth used per call.
Actual Bandwidth Used Per Video Call

To calculate the exact overhead ratio for video, it is recommended that you add about 20 percent
to the video call speed regardless of which type of Layer 2 medium the packets are traversing. The
additional 20 percent gives plenty of headroom to allow for the differences among Ethernet, ATM,
Frame Relay, PPP, High-Level Data Link Control (HDLC), and other transport protocols and also
some cushion for the bursty nature of video traffic. Keep in mind that the amount of bandwidth
consumed by the video correlates directly to the amount of movement on the video. Video streams
with constant motion of the person speaking and the background (such as speaking in front of an
ocean backdrop) can potentially double the amount of bandwidth required. The 20 percent rule is
a general guide, but your actual results might vary.
Table 28-4 shows the recommended bandwidth values to use for some of the more popular video
call speeds, incorporating this 20 percent margin.
Table 28-4 Video Call Speeds and the Associated Audio and Video Codecs
Video Call Speed Requested by Endpoint Actual Bandwidth Required on the Link
128 kbps 153.6 kbps
256 kbps 307.2 kbps
384 kbps 460.8 kbps
512 kbps 614.4 kbps
768 kbps 921.6 kbps
1.5 Mbps 1.766 Mbps
7 Mbps 8.4 Mbps
Call Admission Control in Cisco CallManager

As shown in Figure 28-4, the location configuration settings for Cisco CallManager–based call
admission control have also been enhanced, compared to earlier Cisco CallManager releases not
supporting video, to provide for accounting of video bandwidth on a per-call and aggregate basis.
The location setting for call admission control defines the overall bandwidth allowed for all video
calls to a certain location. That is the video call speed for all video calls. To allow five video calls
with a bandwidth of 384 kbps for each video call (defined in the Cisco CallManager regions), the
value to enter in the Cisco CallManager location is 1920 kbps.
For video calls, the negotiated bandwidth for a video-enabled device typically includes both audio
and video; for example, a 384-kbps video call comprises 64-kbps audio and 320-kbps video
channels. For voice-only calls, the region uses the same setting that is used for the audio channel
in video calls. The negotiated bandwidth for an IP telephony device includes the “real” audio
bandwidth including IP overhead; for example, a G.711 64-kbps audio call uses 80 kbps, and this
value has to be entered in the Cisco CallManager location configuration.
Figure 28-4 Cisco CallManager Enhanced Location Configuration
H.323 gatekeepers have slightly different bandwidth measurements. The H.323 specification
dictates that the bandwidth values must be entered as twice the codec bit rate. For example, a 384-
kbps video call would be entered as 768 kbps in the gatekeeper. A G.711 audio-only call would be
entered as 128 kbps in the gatekeeper.
NOTE Call admission control behavior changed in Cisco CallManager Release 3.2(2)c and
Cisco IOS Software Release 12.2(2)XA. Before that release, Cisco CallManager asked for bit
rate plus Layer 3 overhead, and Cisco IOS gateways asked for 64 kbps, regardless of the type of
call.
Call Admission Control Within a Cluster

Why do you need to configure call admission control for video calls? The answer to this question
is quite simple. If you do not control the number of video and voice calls over your WAN links in
a centralized deployment (shown in Figure 28-5), periodically you will see decreased video and
voice throughout the whole system. With call admission control, you control the number of calls
on the WAN link and ensure that all calls, video and audio-only, are processed throughout the
network with acceptable quality.
Figure 28-5 Typical Centralized Call Control Environment
IP WAN
IP
Cisco CallManager uses regions and locations to implement call admission control:
■ Regions define the codec used and maximum video bandwidth allowed per call. This value is
configurable for calls within each region and for calls between any pair of regions.
■ Locations define the maximum bandwidth allowed for all calls to and from a location.
In Cisco CallManager, devices derive their region setting with the associated device pool
configuration. You can assign locations on a per-device basis. If you assign locations at the device
level, it overrides the setting inherited from the device pool.
Calls between devices at the same location do not need call admission control because the
assumption is that these devices reside on the same LAN that has “unlimited” available bandwidth.
However, for calls between devices at different locations, the assumption is that there is an IP
WAN link in between that has limited available bandwidth.
Region Configuration
When you are configuring a region, you set two fields in Cisco CallManager Administration: the
Audio Codec and the Video Call Bandwidth fields, as shown in Figure 28-6. Note that the audio
setting specifies a codec type, whereas the video setting specifies the bandwidth that you want to
allow. However, even though the notation is different, the Audio Codec and Video Call
Bandwidth fields actually perform similar functions. The Audio Codec value defines the
maximum bit rate allowed for audio-only calls and for the audio channel in video calls.
Figure 28-6 Cisco CallManager Region Configuration

For instance, if you set the Audio Codec value for a region to G.711, Cisco CallManager allocates
64 kbps as the maximum bandwidth allowed for the audio channel for that region. In this case,
Cisco CallManager permits calls using G.711, G.728, or G.729. However, if you set the Audio
Codec value to G.729, Cisco CallManager allocates only 8 kbps as the maximum bandwidth
allowed for the audio channel; in addition, Cisco CallManager permits calls using only G.729,
because G.711, G.722, and G.728 all require more than 8 kbps.
The Video Call Bandwidth value defines the maximum bit rate for the video call, that is, the bit
rate of the voice and video channels. For instance, if you want to allow video calls at a speed of
384 kbps using G.711 audio, you would set the Video Call Bandwidth value to 384 kbps and the
Audio Codec value to G.711. The bit rate that would be used by the video channel thus would be
320 kbps. If the Video Call Bandwidth value is set to None for the region, Cisco CallManager
either terminates the call or allows the call to pass as an audio-only call, depending on whether the
calling device has the Retry Video Call as Audio option enabled.
In summary, the Audio Codec field defines the maximum bit rate used for the audio channel of
audio-only calls and for the audio channel of video calls, whereas the Video Call Bandwidth field
defines the maximum bit rate allowed for video calls and includes the audio portion of the video
call.
NOTE Video endpoints typically support only G.711 and G.722, whereas audio-only
endpoints typically support only G.711 and G.729. Because you cannot configure the audio
codec for audio and video calls separately, often the only common audio codec for mixed
environments (including third-party equipment) is G.711.
Location Configuration
When configuring locations, you also set two fields in Cisco CallManager Administration: the
Audio Bandwidth and the Video Bandwidth values, shown in Figure 28-7. Unlike regions,
however, audio bandwidth for locations applies only to audio-only calls, whereas video bandwidth
again applies to the video call (that is, audio and video channels).
Figure 28-7 Cisco CallManager Location Configuration
The audio and video bandwidth are kept separate because if both types of calls shared a single
allocation of bandwidth, it is very likely that audio calls would take all of the available bandwidth
and leave no room for any video calls, or vice versa.
Both the Audio Bandwidth and the Video Bandwidth fields offer three options: None,
Unlimited, or a field that accepts numeric values. However, the values entered in these fields use
two different calculation models:
■ For the Audio Bandwidth field, the value entered has to include the Layer 3 and 4 overhead
required for the call. For instance, if you want to permit a single G.729 call to or from a
location, you would enter the value 24 kbps. For a G.711 call, you would enter the value 80
kbps. The payload of a G.711 voice call is 64 kbps; the 80-kbps value is the 64-kbps payload
plus 16-kbps RPT plus UDP.
■ The value in the Video Bandwidth field, by contrast, is entered without any overhead. For
instance, for a 128-kbps call, enter the value 128 kbps; for a 384-kbps call, enter the value 384
kbps. As with the values used in the Video Bandwidth field for regions, it is recommended
that you always use increments of 56 kbps or 64 kbps for the Video Bandwidth field for
locations.
NOTE The value None in the Video Bandwidth field indicates that video calls are not
allowed between this location and other locations. Video calls can, however, be placed within
this location.
Call Admission Control Example

Figure 28-8 illustrates an example of audio and video bandwidth requirements for a company with
a three-site network. The San Francisco location has a 1.544-Mbps T1 circuit connecting it to the
San Jose main campus. The system administrator wants to allow four G.729 voice calls and one
384-kbps video call to or from that location.
Figure 28-8 Video Call Bandwidth Example
IP IP
San Jose
1
2
IP
IP IP IP IP
San Francisco Dallas

San Francisco Dallas
Audio Bandwidth 96 kbps 640 kbps
Video Bandwidth 384 kbps 768 kbps
The Dallas location has two 1.544-Mbps T1 circuits connecting it to the San Jose main campus,
and the administrator wants to allow eight G.711 voice calls and two 384-kbps video calls to or
from that location.
For this example, the administrator might set the San Francisco and Dallas locations to the values
in Table 28-5.
Table 28-5 Bandwidth Example
Location Number of Audio Number of Video

Audio Calls Bandwidth Video Calls Bandwidth
Desired Field Value Desired Field Value
San Francisco Four using G.729 96 kbps (4 * 24 One at 384 kbps 384 kbps
kbps)
Dallas Eight using 640 kbps (8 * 80 Two at 384 kbps 768 kbps
G.711 kbps)
First, a video device in San Francisco calls a video device in San Jose. Call admission control
allows exactly one 384-kbps video call between San Francisco and any other location. The video
call is active.
Next, another video device from San Francisco tries to set up a video call to a video device in
Dallas. Call admission control does not allow a second video call from San Francisco to any other
location and denies the video call.
If the call fails because of insufficient location bandwidth, it will not be retried with lower-bit-rate
codecs. In this scenario, with no further configuration of the Cisco CallManager, the call (both
audio and video portions) will be rejected.
Retry Video Call as Audio

The Retry Video Call as Audio setting appears as a check box in the IP Phone configuration in
Cisco CallManager Administration, as shown in Figure 28-9.
Figure 28-9 Configuring the Retry Video Call as Audio Setting
This setting is enabled by default on all device types and applies to these scenarios only:
■ The region is configured not to allow video.
■ The location is configured not to allow video, or the requested video speed exceeds the
available video bandwidth for that location.
■ The requested video speed exceeds the zone bandwidth limits of the gatekeeper.
When this option is activated (checked), if there is not enough bandwidth for a video call (for
example, if the Cisco CallManager regions or locations do not allow video for that call), Cisco
CallManager will retry the call as an audio-only call. When this option is deactivated (unchecked),
Cisco CallManager will not retry the video call as audio only but will instead either reject the call
or reroute the video call by whatever Automated Alternate Routing (AAR) path is configured.
The Retry Video Call as Audio option takes effect only on the terminating (called) device,
allowing flexibility for the calling device to have different options (retry or AAR) for different
destinations.
NOTE The called device determines the result. In other words, when one device calls another
and any of the discussed insufficient bandwidth conditions applies, Cisco CallManager looks at
the destination device to see whether the Retry Video Call as Audio option is enabled.
Call Admission Control Between Clusters

Calls between Cisco CallManager clusters typically use intercluster trunks. Cisco CallManager
Release 4.1 supports these types of trunks:
■ Nongatekeeper-controlled intercluster trunks—This trunk type is specifically designed for

communications between Cisco CallManager clusters and should not be used with other types
of H.323 devices. The name implies that there is no gatekeeper between the clusters to
regulate the bandwidth used between them. Therefore, the only way to provide call admission
control for this type of trunk is by using Cisco CallManager locations. To have Cisco
CallManager control call admission control, the intercluster trunk has to be in a separate
Cisco CallManager location than the devices that use the intercluster trunk.
■ Gatekeeper-controlled intercluster trunks—This trunk type is specifically designed for

communications between Cisco CallManager clusters and should not be used to register other
H.323 endpoints or devices. The name implies that there is a gatekeeper between the clusters
to regulate the bandwidth used between them. To provide call admission control, you
configure an H.323 zone in the gatekeeper for each of the Cisco CallManager clusters and
assign zone bandwidth limits to each zone. If the gatekeeper-controlled intercluster trunk is
not in the same location as the devices, the location settings need to be reconsidered. A device
will need to get its call admission control permissions first from Cisco CallManager, because
the device is in a different location than the trunk, and call admission control will be carried
out at the gatekeeper.
■ H.225 gatekeeper-controlled trunks—The H.225 gatekeeper-controlled trunk is designed

for use with any H.323 device other than a Cisco CallManager cluster. In an H.225
gatekeeper-controlled trunk scenario, only the gatekeeper has control over call admission.
■ SIP trunks—SIP trunks are designed for use with any SIP device, including other Cisco
CallManager clusters, either directly or via a Cisco SIP proxy server.
NOTE Because Cisco CallManager Release 4.1 does not support video over the SIP protocol,
this chapter does not cover SIP trunks.
Call Admission Control Between Clusters 641
Gatekeeper Call Admission Control Options

H.323 gatekeepers are hierarchical in nature. A gatekeeper can have one or more local zones.
Using multiple zones allows you to group the devices that use the gatekeeper and gives better call
admission control options (intrazone and interzone limitations). Zones that are served by another
gatekeeper are called remote zones, shown in Figure 28-10. These zones can be configured with
different call admission control settings as well.
Figure 28-10 Sample Gatekeeper Design
Zone A
Zone C
Gatekeeper Gatekeeper
Interzone
Zone B Remote, Interzone
Total, Session
The Cisco gatekeeper might reject calls from an endpoint because of bandwidth limitations.
Rejection can occur if the gatekeeper determines that the bandwidth available on the network is
not sufficient to support the call. This function also operates during an active call when a terminal
requests additional bandwidth or reports a change in bandwidth used for the call. The Cisco
gatekeeper maintains a record of all active calls so that it can manage the bandwidth resources in
its zones.
When an endpoint or gateway is configured to use an H.323 gatekeeper, it first sends an admission
request (ARQ) to the gatekeeper. The gatekeeper checks whether there is bandwidth available. If
the available bandwidth is sufficient for the call, an admission confirmation (ACF) is returned;
otherwise an admission rejection (ARJ) is returned.
As of Cisco IOS Software Release 12.3(1), you can configure the following types of zone
bandwidth limitations on the Cisco gatekeeper:
■ Interzone—The maximum bandwidth of all calls from a local zone to all other zones (local
or remote)
■ Total—The maximum bandwidth of all calls within a local zone

■ Session—The maximum bandwidth allowed for a single session within a local zone
■ Remote—The maximum bandwidth for all calls from all local zones to all remote zones
The first three limitations can be configured individually for each local zone. However, you can
also specify the default configuration for all local zones that are not configured explicitly.
Gatekeeper Call Admission Control Example

The intercluster trunk gatekeeper provides address resolution and bandwidth control between
Cisco CallManager clusters. For example, suppose that your network has two Cisco CallManager
clusters connected via a digital signal level 3 (DS-3) WAN link, as shown in Figure 28-11. Also
assume that you want to allow a maximum of twenty G.711 audio calls and five 384-kbps video
calls between the two clusters.
Figure 28-11 Gatekeeper Call Admission Control Example

Cluster 1 Cluster 2
14085551… 19725551…
Gatekeeper
In this case, configure one zone per cluster using the bandwidth interzone command to restrict
the bandwidth between the two zones to the total of twenty G.711 calls plus five 384-kbps video
calls. If you want to set the maximum bandwidth per call in a local zone, use the bandwidth
session command. The following is a partial H.323 gatekeeper code snippet:
zone local cluster-1 domain.com

zone local cluster-2 domain.com
zone prefix cluster-1 14085551...
zone prefix cluster-2 19725551...
bandwidth interzone cluster-1 6400
bandwidth session cluster-1 768
bandwidth interzone cluster-2 6400
bandwidth session cluster-2 768
Table 28-6 shows a sample scenario using this environment.
Table 28-6 Example for Twenty G.711 Audio Calls and Five 384-kbps Video Calls
Desired Calls Gatekeeper Bandwidth Value

20 * G.711 audio calls 128 kbps * 20 = 2560 kbps
5 * 384-kbps video calls 768 kbps * 5 = 3840 kbps
20 * G.711 audio + 5 * 384-kbps video calls 2560 kbps + 3840 kbps = 6400 kbps
Summary 643
The bandwidth value that you have to enter is twice the bit rate of the call. For example, a G.711
audio call that uses 64 kbps is denoted as 128 kbps in the gatekeeper, and a 384-kbps video call is
denoted as 768 kbps. Table 28-7 shows the bandwidth values for some of the more popular call
speeds.
Table 28-7 Bandwidth Values for Frequently Used Call Speeds
Call Speed Gatekeeper Bandwidth Value

G.711 audio call (64 kbps) 128 kbps
G.729 audio call (8 kbps) 16 kbps
128-kbps video call 256 kbps
NOTE You cannot avoid having more G.711 calls active than desired because the bandwidth
for voice and video is not managed separately. It can happen that more G.711 calls are active
than desired and that they consume the configured bandwidth actually reserved for video calls,
leaving no bandwidth for video calls.
Summary
This chapter discussed the concepts behind a Cisco video telephony solution. Cisco video
telephony supports SCCP and H.323 devices communicating in any combination. For
videoconferences, an MCU is required. The typical video call includes two or three RTP streams
in each direction. Using SCCP clients allows you to offer end users major PBX features, whereas
using H.323 clients limits your end users to only supporting basic call functionality.
When determining bandwidth requirements for the audio and video calls, adding 20 percent to the
payload bandwidth requirement for video calls provides plenty of buffer for overhead information.
Bandwidth call admission control within a cluster is done using Cisco CallManager locations and
regions. Between clusters, you must use H.323 gatekeeper control.
Review Questions
1. What is the video payload in a video call using 384 kbps and the G.711 codec?
a. 300 kbps
b. 320 kbps
c. 376 kbps
d. 384 kbps
2. To enable video calls with a requested speed of 256 kbps, what would the recommended
bandwidth setting be on Cisco CallManager locations?
a. 307.2 kbps
b. 256 kbps
c. 512 kbps
d. 281.6 kbps
3. Which video codec will be used when an SCCP and H.323 device communicate?
a. H.263
b. H.264
c. Cisco wideband codec
d. H.261
4. How is call admission control performed between Cisco CallManager clusters with a
nongatekeeper-controlled intercluster trunk?
a. Gateway
b. Cisco CallManager locations
c. Gatekeeper
d. MCU
e. Intercluster trunk
5. Which device performs call admission control within a cluster?
a. Gateway
c. Gatekeeper
d. MCU
e. Intercluster trunk
6. Which device is needed to integrate H.320 into the Cisco video solution?
a. MCU
b. Video gateway
c. MGCP gateway
d. H.323 gatekeeper
7. Which of the following are required to support videoconferencing capabilities?
a. Gateway
c. Gatekeeper
d. MCU
e. SCCP Clients support videoconferencing natively.
8. Which of the following best defines FECC?
a. The ability to negotiate audio codecs between H.323 and SCCP devices
b. The ability to negotiate video codecs between H.323 and SCCP devices
c. The ability to change the position of a remote camera
d. Converting between the H.320 and H.263 video codecs
9. How much bandwidth is consumed by the Cisco proprietary wideband codec?
a. 384 kbps
b. 768 kbps
c. 1.544 Mbps
d. 7 Mbps
10. Which Cisco CallManager release first added video support?
a. Cisco CallManager 3.2(3)
b. Cisco CallManager 3.3(3c)
c. Cisco CallManager 4.0(1)
d. Cisco CallManager 4.1(1)
following topics:
■ Describing the features and functions of

Cisco VT Advantage
■ Explaining how placing and receiving calls
works with Cisco VT Advantage
■ Configuring Cisco CallManager for video
■ Configuring feature settings in Cisco
CallManager to support video on Cisco IP
Phones
■ Installing the Cisco VT Advantage
application and identifying the hardware and
software required to run it
CHAPTER 29
Configuring Cisco VT Advantage
Cisco Video Telephony (VT) Advantage is a video telephony solution comprising the Cisco VT
Advantage software application and Cisco VT Camera, a video telephony Universal Serial Bus
(USB) camera. With the Cisco VT Camera attached to a PC wired through a Cisco IP Phone,
users can place and receive video calls on the enterprise IP telephony network. By using the VT
Advantage solution, you can take full advantage of your existing IP network to deliver
communications that extend voice and video to every user in the organization. This chapter
discusses the concepts and configuration behind Cisco VT Advantage.
NOTE As with most products in the Cisco IP telephony line, the Cisco VT Advantage has
recently been renamed to Cisco Unified Video Advantage. For brevity, we have decided to use
the shorter VT Advantage name throughout this chapter.
Cisco VT Advantage Overview

Cisco VT Advantage brings video telephony functionality to the Cisco IP Phone 7940, 7960,
and 7970 (and later) models. Cisco VT Advantage software coupled with the Cisco VT Camera
(a USB camera) allows a PC connected to a Cisco IP Phone to add video to telephone calls
without requiring any extra button-pushing or mouse-clicking. When registered to Cisco
CallManager, the Cisco VT Advantage-enabled Cisco IP Phone has the features and
functionality of a full-featured IP video phone. Supplementary services, such as call forward,
transfer, hold, and mute, are also available for video calls and are all initiated through the Cisco
IP Phone. Cisco VT Advantage is intended for desktop-to-desktop IP video telephony
environments, not as a general-purpose videoconferencing solution for use in conference rooms.
Cisco VT Advantage Components

To deploy Cisco VT Advantage, the minimum requirement is Cisco CallManager Release 4.0(1)
with Service Release 2 or higher. Currently, you can enable video on Cisco IP Phone 7940G,
7960G, and the 7970G (and later) models. The Cisco VT Camera is connected to a PC (via USB)
648 Chapter 29: Configuring Cisco VT Advantage
where Cisco VT Advantage software is installed. Cisco VT Advantage software works only with
the Cisco VT Camera.
NOTE Cisco VT Advantage and the Cisco IP Communicator (Softphone running on a PC) can
run on the same PC; however, Cisco VT Advantage is not supported to interconnect with Cisco
IP Communicator.
Cisco VT Advantage software provides the user with an easy-to-use graphical interface, with these
options:
■ Receive Only Mode—Users can choose to view incoming video only and not transmit video.
■ Video Check—Users can check their video before calls are placed or received.
■ Mute Video on Audio Mute—When users mute the audio on the IP phone, video is
automatically paused until the audio on the IP phone is restored.
■ Video Signal Indicators—The quality of the incoming and outgoing video is graphically
displayed.
■ Connectivity Indicator—Graphics are used to indicate the state of the connections from the
PC to its associated Cisco IP Phone and Cisco VT Camera.
Users can operate the Cisco IP Phone as they normally do. The Cisco VT Advantage software is
controlled from the PC connected directly to the access port labeled “PC” on the back of the Cisco
IP Phone. The PC and the Cisco IP Phone that is registered in Cisco CallManager as a video-
enabled device build an association. The voice Real-Time Transport Protocol (RTP) streams flow
between the two IP phones, as in a normal voice call. The video streams flow between the two PCs
where the Cisco VT Advantage software is installed.
Cisco VT Advantage Supported Standards

Cisco VT Advantage, like any other application that runs on a PC, has an impact on system
performance that should be taken into consideration. Cisco VT Advantage supports two types of
video codecs: H.263 and the Cisco VT Camera wideband video codec. Of these two types, the
Cisco VT Camera wideband video codec places the least demand on the PC. Therefore, if your
network has plenty of available bandwidth, you can use the Cisco VT Camera wideband video
codec and save on PC CPU and memory resources.
Cisco VT Advantage Overview 649
When you are using a codec that performs compression, more CPU power is needed. The H.263
codec is more demanding of PC system resources, but it requires less bandwidth. Therefore, if you
want to use H.263 compressed video to conserve bandwidth on the network, you should ensure
that your PCs have enough CPU and memory resources available. The Cisco VT Advantage H.263
codec supports a range of speeds up to 1.5 Mbps.
Regardless of the video standard used, the VT Advantage supports video formats with up to 30
frames per second (fps) at the following resolutions:
■ VGA (640x480)
■ CIF (352 x 288)
■ SIF (320 x 240)
■ QCIF (176 x 144)
■ QSIF (160 x 120)
Table 29-1 lists the video codecs that Cisco VT Advantage supports.
Table 29-1 Video Codecs Supported by Cisco VT Advantage
Codec Parameters
H.263 • Bandwidth: 128 kbps to 1.5 Mbps
• Native Resolution: Common Intermediate Format (CIF) and Quarter CIF
(QCIF)
• Frame rate: Up to 30 frames per second (fps)
Cisco VT Camera • Bandwidth: 7 Mbps
wideband video • Native Resolution: 320 x 240
codec
• Frame rate: Up to 30 fps
Protocols Used by Cisco VT Advantage

Cisco VT Advantage supports several industry-standard and Cisco networking protocols required
for video communication. Table 29-2 displays an overview of the supported networking protocols.
Table 29-2 Overview of the Supported Networking Protocols
Networking Description Usage Notes

Protocol
Cisco Audio • Allows communication between the • Cisco Audio Session Tunnel is used between
Session Tunnel Cisco IP Phone and associated Cisco VT Advantage and the IP Phone
software, such as Cisco VT — To build an association (after the
Advantage. PC discovers the IP phone using Cisco
• Uses source and destination port 4224. Discovery Protocol)
• Uses TCP. — To send signaling information for video
• Cisco proprietary protocol. streams from the IP phone to Cisco VT
Advantage (after the IP phone receives
the signaling messages for both audio
and video from Cisco CallManager)
• Cisco Audio Session Tunnel signaling
messages include these:
— Call video stream start and stop
— Call hold and resume
Cisco Discovery • A device-discovery protocol that runs • Cisco VT Advantage uses Cisco Discovery
Protocol on all Cisco manufactured equipment. Protocol to communicate its capabilities to
• A Layer 2 protocol. the Cisco IP Phone, and the Cisco IP Phone
uses Cisco Discovery Protocol to
• Works only between directly communicate information, such as its IP
connected neighbors. address, to Cisco VT Advantage.
• Using Cisco Discovery Protocol, a
device can advertise its existence to
other devices and receive information
about other devices in the network.
• Cisco proprietary protocol.
RTP • A standard for using User Datagram • The RTP protocol is used to encapsulate and
Protocol (UDP) to transport real-time stream the audio (between Cisco IP Phones)
data, such as interactive voice and and video (between Cisco VT Advantage
video, over data networks. endpoints).
Skinny Client • A Cisco protocol using low-bandwidth • Cisco VT Advantage does not use SCCP
Control Protocol messages that allows the exchange of itself. It uses Cisco Audio Session Tunnel to
(SCCP, or Skinny) signaling messages between IP send signaling messages to the Cisco IP
devices and the Cisco CallManager. Phone, which acts as a proxy and passes the
• Works on TCP port 2000. signaling messages to Cisco CallManager
using SCCP.
• Cisco proprietary protocol.
How Calls Work with Cisco VT Advantage 651
How Calls Work with Cisco VT Advantage

When a Windows PC has Cisco VT Advantage installed, it should be connected to the secondary
Ethernet port (that is, PC port) of a Cisco IP Phone 7940G, 7960G, or 7970G (or later) model. In
most configurations, the PC will be in a different VLAN than the Cisco IP Phone (located in the
voice or auxiliary VLAN). In such configurations, all IP-based communication between Cisco VT
Advantage and the Cisco IP Phone has to be routed between the VLANs, and only Cisco
Discovery Protocol is exchanged directly. The following list explains the call process shown in
Figure 29-1:
1. Cisco Discovery Protocol exchange takes place so that Cisco VT Advantage and the Cisco IP
Phone can discover one another. A Cisco Discovery Protocol driver is installed on the PC
during the installation of Cisco VT Advantage. This allows the Cisco VT Advantage
application to dynamically learn the IP address of the Cisco IP Phone during the Cisco
Discovery Protocol exchange, and associate with it. This serves as both an ease-of-use feature
for the end user and for security. The use of Cisco Discovery Protocol to facilitate the
association process allows it to occur automatically, without the user having to configure the
Cisco VT Advantage application. This allows for mobility of the application between different
IP phones on the network. The user can plug into the PC port of any supported Cisco IP Phone
on the network (if permitted by the administrator) and begin making video telephony calls.
Cisco Discovery Protocol also provides a measure of security in that the IP phone will
respond only to association messages from a Cisco VT Advantage client that matches the IP
address of the device that is connected to its PC port (that is, its Cisco Discovery Protocol
neighbor), minimizing the risk of someone else associating with your Cisco IP Phone over the
network and receiving video when calls are placed on your IP phone. The Cisco IP Phone
begins listening for Cisco Audio Session Tunnel messages on TCP port 4224.
2. After Cisco Discovery Protocol discovery, Cisco VT Advantage and the IP phone exchange
Cisco Audio Session Tunnel protocol messages over TCP port 4224. Cisco VT Advantage
sends a Cisco Audio Session Tunnel message to the IP phone, which is in a different IP
network (VLAN). The packet first travels through the PC VLAN to the default gateway, where
it is routed toward the IP phone (using the voice VLAN). The Cisco Audio Session Tunnel
protocol allows Cisco VT Advantage to associate with the IP phone and receive event
messages from the IP phone when calls are placed or received. After this association process
occurs between the Cisco VT Advantage client and the IP phone, the IP phone updates its
registration status with Cisco CallManager, advising Cisco CallManager of its video
capabilities.
3. When the Cisco IP Phone receives signaling information for video calls, it acts as a proxy
toward Cisco VT Advantage for the setup of the video streams. Only the signaling is proxied,
but when the RTP endpoints (IP addresses and UDP RTP port numbers) are negotiated, the IP
phone specifies the IP address of the PC for the video stream and its own IP address for the
audio stream. When Cisco CallManager tells the Cisco IP Phone to open the video channel,
(communicating to the IP phone using the voice VLAN) the IP phone proxies those messages
to Cisco VT Advantage using the Cisco Audio Session Tunnel protocol. These Cisco Audio
Session Tunnel messages have to be routed between the voice and the PC VLAN again.
4. After the voice and video channels have been successfully set up, the audio stream is sent to
the IP address of the IP phone (to the voice VLAN), whereas the video stream is sent directly
to the PC IP address (to the PC VLAN).
Figure 29-1 Cisco VT Advantage Call Process
Phone VLAN PC VLAN

Si
IP
802.1q
Cisco Discovery Protocol

1
Inter-VLAN 2
Routing “Cisco Audio Session Tunnel: Association”
“SCCP: Call Signaling (Video and Voice)” Signaling

Proxy 3
for Video
Inter-VLAN
Routing “Cisco Audio Session Tunnel : Call Signaling (Video Only)”
Audio Packets 4
Video Packets
NOTE Firewalls or access control lists (ACLs) must permit TCP port 4224 to allow the
exchange of Cisco Audio Session Tunnel messages.
Cisco VT Advantage Video Modes

For privacy, the participants can switch to a mode called receive-only mode to prevent the camera
from sending a picture to the other end of the call. Table 29-3 shows various scenarios of the
calling or the called party activating or disabling receive-only mode.
Configuring Cisco CallManager for Video 653
Table 29-3 Cisco VT Advantage Video Modes
Cisco VT Advantage Cisco VT Advantage Result

Mode on Your PC Mode on the PC of the
Other User
Enabled Enabled When you place or answer a call, two
video windows open on your PC—you
will see yourself in the Local Video
window and the other party in the
Remote Video window.
Receive-only Enabled When you place or answer a call, you
will see the other party in the Remote
Video window. The Local Video
window will not display. The other
party will see a blank image in the
Remote Video window.
Enabled Receive-only When you place or answer a call, you
will see yourself in the Local Video
window and a blank image in the
Remote Video window. The other party
will not see a Local Video window.
Receive-only Receive-only No one will see the other party; this
mode is similar to a telephone call.
NOTE When Cisco VT Advantage is not running on your PC or on the PC of the remote
peer, the call functions as a regular telephone call without video.
Configuring Cisco CallManager for Video

When you are setting up a video IP telephony environment, you first have to configure video in
Cisco CallManager. Cisco VT Advantage requires Cisco CallManager to handle video call
processing. Before enabling video on Cisco CallManager, you must update your locations and
regions settings to adjust your bandwidth settings. Media Resource Group Lists (MRGLs) are
used to control the access to multipoint control units (MCUs). Only devices that are allowed to use
an MRGL are able to use the resources in the MRGL. Using this feature, you should restrict access
to the video conference MCU resources to only the video-capable endpoints. Otherwise, the MCU
resources could be depleted by managing audio conferences best handled by other hardware.
Further, you have to reconsider the call-routing configuration when using, for example, Automated
Alternate Routing (AAR). Another important point that arises during the consideration of video is
the Differentiated Services Code Point (DSCP) settings for quality of service (QoS).
In large Cisco CallManager environments, you have to consider whether it makes sense to use the
Cisco VT Advantage Deployment Tool to make Cisco VT Advantage software available for
download in Cisco CallManager.
Call-Routing Considerations
One of the advantages of using Cisco VT Advantage is that you can use the existing dial plan. If
the bandwidth needed by an endpoint for a video call is not available, by default the call is retried
as an audio call. To use route or hunt lists or AAR groups to try different paths for such video calls
instead of retrying them as audio calls, uncheck the Retry Video Call as Audio check box in the
configuration settings for the applicable gateways, trunks, and IP phones.
Video-enabled IP phones should have a separate MRGL with the videoconference bridge as the
first choice. If the nonvideo-enabled IP phones use the videoconference bridge as a first choice,
you run the risk of having no videoconference resources available for videoconference calls
because all the videoconference resources are occupied by audio-only conferences. It is
recommended that you create two separate MRGLs—one for video-enabled IP phones and one for
nonvideo-enabled IP phones.
Video is as time-critical as voice. In a voice- and video-enabled network, you have to prioritize
voice and video packets so that you do not experience quality issues. Both voice and video must
be of higher priority than data, for example.
DSCP packet marking includes these characteristics:
■ Audio streams in audio-only calls default to the Expedited Forwarding (EF) class.
■ Video streams and associated audio streams in video calls default to Assured Forwarding,
Class 4, with low drop precedence (AF41).
You can change these defaults using the Cisco CallManager Enterprise Parameters.
These service parameters affect DSCP packet marking:
■ DSCP for Audio Calls (for Media RTP Streams)—This parameter specifies the DSCP
value for audio calls.
■ DSCP for Video Calls (for Media RTP Streams)—This parameter specifies the DSCP value
for video calls.
Configuring Cisco CallManager for Video 655
VT Advantage Deployment Tool

For simplified and scalable deployment of Cisco VT Advantage client installation software, you
can use the Cisco VT Advantage Deployment Tool. Administrators can use this tool to make the
Cisco VT Advantage installer program available on a Cisco CallManager publisher server. The
installer program will then reside in the CCMPluginsClient website that is mapped to the
C:\CiscoPlugins\Client\CVTA directory. This website is set up with the correct permissions to
allow anonymous access to the Cisco VT Advantage Installer executable file to facilitate
installation for your technicians and users.
The Cisco VT Advantage Deployment Tool lets you set these options for the installation:
■ AutoUpdate—Sets the AutoUpdate option so that users are notified automatically about
updates to Cisco VT Advantage
■ Proxy—Sets proxy server information for users needing to use a proxy server to reach the
Cisco CallManager publisher server
■ Error Reporting Tool—Sets the e-mail or FTP addresses where users send reports generated
by the Error Reporting Tool
To publish Cisco VT Advantage to Cisco CallManager, the administrator must download the latest
available Cisco VT Advantage Deployment Tool from Cisco.com. Run the DeployMan.exe file to
set up Cisco VT Advantage for end users. The Cisco VT Advantage installer program is now stored
in Cisco CallManager and a download link will be made available on Cisco CallManager user-
accessible installation pages.
To install Cisco VT Advantage on a PC, users must complete these steps:
Step 1 Access the Cisco CallManager user installation page and download the
Cisco VT Advantage software: https://<CCM Publisher Server name or IP
address>/CCMUser. (This URL is found in the Services Help screen on
Cisco IP Phone models that support Cisco VT Advantage.)
Step 2 Install Cisco VT Advantage software on the PC.
When you execute the DeployMan.exe file, the DeployMan window opens, as shown in Figure 29-2.
When you click the Use Defaults button in the DeployMan main window, the Cisco VT Advantage
Deployment Tool must be running on the Cisco CallManager publisher server. If this is not the
case, change the path in the CVTAInstall.exe Destination field by referring to the
C:\CiscoPlugins\Client\CVTA directory at the publisher via a network file share. The Choose Host
Name dialog box opens. Enter the hostname (or IP address) of the Cisco CallManager publisher
server. This value populates the Update URL field for the AutoUpdate feature.
Figure 29-2 Using the VT Advantage Deployment Tool
These fields are automatically populated with default values:
■ Cisco VT Advantage Version
■ CVTAInstall.exe Destination
■ Update URL
■ Comma-Delimited E-Mail/FTP Addresses
Make sure that the Update URL field in the AutoUpdate Options area contains the hostname (or
IP address) of the Cisco CallManager publisher server. If you do not want to use AutoUpdate,
uncheck the Deploy AutoUpdate check box. The Precertification Options area will be usable only
in later versions of the deployment tool, even though it is shown in the GUI of the current version
(2.0).
If users in the network need to use a proxy server to reach the Cisco CallManager publisher server,
check the Use HTTP Proxy check box and fill in the Proxy URL and Proxy Port fields with the
appropriate values.
Configuring Cisco IP Phones for Cisco VT Advantage 657
In the Comma-Delimited E-Mail/FTP Addresses field of the Error Reporting Tool Options,
enter the e-mail or FTP addresses to which error reports generated by users can be sent. You can
enter multiple addresses separated by commas. The default e-mail address is [email protected].
This e-mail address is used by the Cisco Technical Assistance Center (TAC) to pick up files sent
by customers.
When you have filled in all necessary options, click OK to make the Cisco VT Advantage
installation program available to users.
Configuring Cisco IP Phones for Cisco VT Advantage

In addition to the global Cisco CallManager parameters that are needed for a video-enabled IP
telephony network, you have to configure the individual devices (IP phones) to support video calls.
The Cisco IP Phone requires Cisco CallManager for call processing and the appropriate phone
load to support video on the IP phone. When you are configuring IP phones for video in Cisco
CallManager, you will need to ensure you have configured these settings:
■ Make sure that a phone load that supports video is installed on each Cisco IP Phone that will
be video-enabled.
■ The port labeled “PC” on the back of the Cisco IP Phone connects a PC or a workstation to
the IP phone so that they can share a single network connection. Make sure that this feature
is enabled on Cisco IP Phones that operate with Cisco VT Advantage.
■ The Cisco IP Phone has to be configured to support video calls.
■ Check or uncheck the Retry Video Call as Audio check box. When the Retry Video Call as
Audio check box is checked, a Cisco IP Phone that cannot obtain the bandwidth that it needs
for a video call will retry the call as an audio call.
■ Verify that the Cisco IP Phone is video-enabled after configuring the IP phone in Cisco
CallManager configuration.
Verifying Phone Loads

As shown in Figure 29-3, CallManager assigns each device type a phone load that it will use from
the TFTP server. Devices must use phone load 6.0(3) or later for the Cisco IP Phone 7940 and 7960
models to support the Cisco VT Advantage. For the Cisco IP Phone 7970, use the phone load
6.0(1) or later. To verify the phone loads, from the Cisco CallManager Administration, choose
System > Device Defaults.
Figure 29-3 Configuring Phone Load Defaults
NOTE You can always download the latest phone loads from Cisco.com.
Configuring IP Phones to Support Video

The IP phone configuration settings that are required for video support can be found in the IP
Phone Configuration window (shown in Figure 29-4) of Cisco CallManager Administration.
Configuring Cisco IP Phones for Cisco VT Advantage 659
Figure 29-4 Configuring IP Phones to Support Video
TIP The IP phone settings need not be configured before Cisco VT Advantage can be loaded
on the client PC. But the preferred sequence is to configure the IP phone first and then install
the Cisco VT Advantage software.
Because the PC that has Cisco VT Advantage installed needs to be physically connected to a PC
port of the IP phone, ensure that the PC port of the IP phone is not disabled in the Cisco
CallManager IP phone configuration. By default, the PC port is enabled.
When the Video Capabilities field is set to Enabled, the phone will participate in video calls
when connected to an appropriately equipped PC. Make sure that this feature is enabled on Cisco
IP Phones that operate with Cisco VT Advantage. Video capability is disabled by default.
In addition, a video-enabled Cisco IP Phone can be configured to retry video calls as audio calls
if the video call cannot be set up. The Retry Video Call as Audio check box is located in the Phone
Configuration window, shown in Figure 29-5, and this feature is activated by default. To enable
the Retry Video Call as Audio feature, choose Cisco CallManager Administration > Device >
Phone.
Figure 29-5 Enabling Retry Video Call as Audio
Figure 29-6 Verification of Phone Configuration

Camera Icon
Verification of Phone Configuration

An IP phone enabled for video displays a video camera icon in the lower-right corner of its liquid
crystal display (LCD) screen, as shown in Figure 29-6. A PC with Cisco VT Advantage installed
does not have to be connected to the IP phone to produce the video camera icon. The camera icon
is displayed as soon as video is enabled for the IP phone in Cisco CallManager configuration.
Installing Cisco VT Advantage on a Client

After configuring Cisco CallManager and the Cisco IP Phone, you need to install Cisco VT
Advantage on the PC. When installing VT Advantage, use this setup procedure:
Step 1 Consider the hardware and software requirements to install Cisco VT

Advantage.
Step 2 Verify the preparation checklist to ensure that all necessary preinstallation
tasks have been completed successfully. After the verification, install the
Cisco VT Advantage software and hardware.
Step 3 Verify the Cisco VT Advantage installation using Cisco VT Advantage
tools.
Cisco VT Advantage Hardware and Software Requirements

To run the Cisco VT Advantage, you must be using one of the Cisco IP Phone 7940, 7960, or 7970
(or later) models. In addition, Table 29-4 describes the hardware requirements for the PC where
the Cisco VT Advantage software is installed.
Table 29-4 Cisco VT Advantage Hardware Requirements
PC Feature Minimum Requirements

CPU 1.0-GHz or higher Pentium III or compatible processor
(Streaming Single Instruction Stream, Multiple Data Stream
[SIMD] Extensions support required)
1.4-GHz or higher Pentium III or compatible processor

recommended
System memory 256 MB minimum
Free disk space 40 MB
Universal Serial Bus (USB) port At least one free USB (1.1 or 2.0 compliant) port
Video display Video-capable graphics card at 800 x 600 x 16 bits or better
Network 10/100-Mbps Ethernet network interface card (NIC)
Of course, the Cisco VT Camera will be connected to the PC when prompted during the
installation process of the Cisco VT Advantage software on the PC.
NOTE Cisco VT Advantage supports only the Cisco VT Camera, and the Cisco VT Camera
works only with the Cisco VT Advantage software.
The PC must be using either Microsoft Windows 2000 Professional with Service Pack 3.0 or later
or Windows XP Professional with Service Pack 1.0 or later.
The Cisco VT Advantage Software can be downloaded in two ways:
■ From the Cisco CallManager publisher (if the administrator used the deployment tool):
The default URL for the software download when Cisco VT Advantage is deployed
with the deployment tool is https://<CCM Publisher Server name or IP address>/
CCMUser.
■ From Cisco.com:
— To download the Cisco VT Advantage software from Cisco.com, users need a valid
Cisco.com account with the necessary software download privileges.
— Alternatively, the administrator can download the Cisco VT Advantage software and
provide it to users via CD, file server, or any other means.
Cisco VT Advantage Installation Preparation Checklist

Check these points before installing Cisco VT Advantage software on a PC:
■ Ensure that the Cisco IP Phone is properly connected to the corporate telephony network. You
could do a short audio test call to ensure that the IP phone is working correctly.
■ Ensure that the Cisco IP Phone is video-enabled. If the LCD screen on the Cisco IP Phone
displays the video camera icon on the status line, the phone is video-enabled.
■ Ensure that the Ethernet port of the PC is connected to the PC port of the IP phone because
this connection is mandatory for Cisco VT Advantage.
■ For the Cisco Audio Session Tunnel protocol to operate between Cisco VT Advantage and the
IP phone, the PC must be capable of reaching the IP phone over TCP/IP. Typical IP telephony
designs use separate voice and data VLANs, as well as ACLs or firewalls between those
VLANs to secure the IP telephony network.
NOTE Do not forget to enable port 4224 on your firewall.

Installing Cisco VT Advantage

To install the Cisco VT Advantage software on a client PC, complete the following steps:
Step 1 Open your web browser, enter the Cisco VT Advantage download URL in
the address field, and press Enter.
NOTE The default URL, when Cisco VT Advantage is deployed with the deployment tool
is https://<CCM Publisher Server name or IP address>/CCMUser.
Step 2 On the Cisco CallManager Client Install Plugins page, click the Cisco VT
Advantage Installer Plug-In icon, shown in Figure 29-7. The camera
should not be connected at this time.
Figure 29-7 VT Advantage Plug-In Installation
Step 3 Follow the instructions presented in the dialog boxes to complete the
installation of Cisco VT Advantage:
• In the Welcome window shown in Figure 29-8, click Next to start the
installation wizard.
Figure 29-8 VT Advantage Installation Wizard
• In the License Agreement window, select I Accept the Terms in the

License Agreement, and click Next.
• In the Customer Information window, enter the user information, select
the desired option, and click Next.
• In the Destination Folder window, accept the default installation folder
or click Change to enter a different installation folder. When you are
done, click Next.
— In the Ready to Install window, click Install. Depending on the setup
of your PC, you might see messages for the installation of the Cisco
Media Termination Driver and the Cisco VT Camera software that
differ, based on the operating system of the PC:
— Windows 2000—If a “Digital Signature Not Found” message
appears, the driver software has no Microsoft digital signature that
would affirm that the software has been tested with Windows and that
the software has not been altered since it was tested. Click Yes to
continue.
— Windows XP—If a “Hardware Installation” message appears, the
software that you are installing has not passed Windows Logo testing
to verify its compatibility with Windows XP. To continue the
installation, click Continue Anyway.
Step 4 Plug the Cisco VT Camera into an available USB port on the PC. As soon
as you plug in the camera, the operating system finds it automatically. A
window appears offering the creation of a shortcut to the application.
Choose the desired option and then click Next.
Step 5 A window appears indicating that the installation has completed. Confirm
the message by clicking Finish. If you are prompted to restart the PC, click
Yes to restart the PC.
Cisco VT Advantage Installation Verification

When the Cisco VT Advantage application is started, verify its connectivity; if a connection to the
Cisco IP Phone or to the Cisco VT Camera is not working, a red X is displayed over the
corresponding connecting line, as shown in Figure 29-9. If the connections are functioning, the
corresponding lines are shown without the red X and the lines themselves are green.
Figure 29-9 Verifying VT Advantage Connections
Camera Not Connected Phone Not Connected
When you have ensured that the camera connection is functioning, you should run a video check.
To start the video check, choose Video > Start Video Check. While the video check is being
performed, two green bars are displayed (one for the sending video signal and the other for the
receiving video signal) when the system is working. In addition, two popup windows show the
local and remote views, as shown in Figure 29-10. During the video check, you will see the same
image in both windows.
Figure 29-10 Cisco VT Advantage Video Check
After a successful video check, you should be able to place and receive calls. Place a test call to
another IP phone with Cisco VT Advantage enabled, and verify the local and remote views. While
in a call, you can launch the diagnostic tool, shown in Figure 29-11. The diagnostic tool provides
some technical details about the current state of the Cisco VT Advantage software that is running
on the PC, as well as some indications about the Cisco VT Camera frame rate and light level.
Figure 29-11 Cisco VT Advantage Diagnostic Tool

Summary 667
To use the diagnostic tool, open the Cisco VT Advantage main window and double-click the video
signal quality bars. The Diagnostics window opens. When users report seeing a low signal quality
bar, you can check the Cisco VT Camera frame rate and light level. Look near the center of the
Diagnostics window for the Camera area. It contains two fields:
■ Frame Rate—Indicates the frames per second reported by the camera. The values are 5, 10,
15, 20, 25, or 30. The LED colors correspond to the frame rate: Green is 30 fps, yellow is 5,
10, 15, 20, or 25 fps, and black is blank or off.
■ Light Level—Indicates the lighting conditions reported by the camera. The values are Off,
Normal, and Low Light. The LED colors correspond to the light levels: Black is off, green
is normal, and red is low light. Increasing the lighting conditions near the camera should result
in a higher frame rate and a normal light level. When you are troubleshooting some Cisco VT
Advantage problems with the assistance of Cisco TAC representatives, they might ask you to
provide them with the information displayed in the Diagnostics window.
A low frame rate can be caused by low light conditions. The Cisco VT Camera is normally set for
automatic exposure. When the lighting is low, the camera has to expose each frame for a longer
time, resulting in a lower frame rate. To test for this issue with the diagnostic tool, double-click
the signal quality bars. The Diagnostics window opens. The Video Signal area at the middle left
of the window contains fields showing the current number of frames per second being processed.
At this point, all the data is coming from the camera, so if the value is less than 15 fps, the problem
most likely is the lighting conditions. Improving the lighting near the camera should result in a
higher frame rate and a normal light level.
Summary
This chapter discussed the concepts and configuration behind Cisco VT Advantage. This software
associates with the Cisco IP Phone, which registers as a video-capable phone with Cisco
CallManager. The IP phone then acts as a SCCP proxy between the Cisco VT Advantage software
and Cisco CallManager.
To properly configure video telephony, you should first configure Cisco CallManager locations
and regions to support video bandwidth levels. You should then configure the Cisco IP Phones for
video using the CallManager Administration utility. Finally, you must install the VT Advantage
software on the end-user PCs. Be sure to wait until the installation wizard prompts you to plug in
the Cisco VT Camera.
Review Questions
1. What does the icon of a video camera on the Cisco IP Phone status line mean?
a. The Cisco VT Camera is connected to the PC, and the PC is connected to the Cisco IP
Phone.
b. The Cisco IP Phone has web access.
c. The Cisco IP Phone is configured with video capabilities.
d. The Cisco IP Phone is receiving a video call.
2. Into which port should the PC with Cisco VT Advantage client software installed be plugged?
a. Cisco Catalyst switch port
b. Cisco IP Phone access port
c. MCU port
d. Videoconferencing port
3. Which protocols are supported by Cisco VT Advantage?
a. Cisco Audio Session Tunnel
b. FTP
c. RTP
d. SSH
e. TFTP
4. Which SCCP Cisco endpoints are video-capable? (Choose three.)
d. Cisco IP Phone 7920
e. Cisco IP Phone 7940
f. Cisco IP Phone 7960
g. Cisco IP Phone 7970
5. Which of the following IP phone settings are enabled by default in Cisco CallManager?
(Choose two.)
a. Video capabilities
b. PC port
c. Retry Video Call as Audio
d. Video calling search space
e. AAR
6. Video calls using 384 kbps need to be supported across a gatekeeper-controlled trunk. What
value should be entered into the gatekeeper to support this bandwidth?
a. 192 kbps
b. 384 kbps
c. 512 kbps
d. 768 kbps
7. Which of the following web camera(s) is VT Advantage compatible with?
(Select all that apply.)
a. Cisco VT Camera
b. Logitech E905
c. Creative Labs ClearStream
d. Mindworks v11
8. How much bandwidth is consumed by the Cisco proprietary wideband codec?
a. 384 kbps
b. 768 kbps
c. 1.544 Mbps
d. 7 Mbps
9. The VT Advantage software can stream up to ______ fps.
a. 10
b. 15
c. 20
d. 30
Part VII: IPT Management
Chapter 30 Introducing Database Tools and Unified

CallManager Serviceability
Chapter 31 Monitoring Performance
Chapter 32 Configuring Alarms and Traces
Chapter 33 Configuring CAR
Chapter 34 Using Additional Management and Monitoring Tools

following topics:
■ Examining the database structure and

replication status and reactivating broken
database connections
■ Identifying the purpose of the major services
that Cisco CallManager Serviceability
provides
■ Explaining how to use the Control Center to
start and stop services
■ Explaining how to use the Service Activation
window tools to enable and disable services
■ Classifying the tools used to monitor Cisco
CallManager by their function
CHAPTER 30
Introducing Database
Tools and Cisco Unified
CallManager Serviceability
Phone communication is one of the most critical services for businesses today. It is necessary to
maintain and troubleshoot Cisco CallManager installations as fast as possible to reduce the risk
of system outages. Administrators need tools to easily gather information about system behavior
and troubleshoot database replication. This chapter introduces database tools that allow
administrators to manage the Microsoft Structured Query Language (SQL) databases used by
Cisco CallManager. It gives an overview of how to use Cisco CallManager Serviceability and
other tools that you can use to maintain the Cisco CallManager system.
Database Management Tools

On Cisco CallManager systems, Microsoft SQL 2000 Enterprise databases are used to store
system and device configuration as well as call detail records (CDRs). To maintain data
consistency throughout the cluster, the publisher database server uses one-way, or
unidirectional, replication to each subscriber. All information is stored on one server (the
publisher server) and replicated to the other servers (subscriber servers) in the cluster.
Replicating the SQL database is a core function inside Cisco CallManager clusters.
Configuration changes on Cisco CallManager systems are possible only while the publisher
server is available. To make sure that CDRs are written even if the publisher server is offline,
every server stores CDR information locally in flat files. Cisco CallManager writes information
to the publisher database only if Cisco CallManager web components are installed or Cisco
CallManager Administration is performed directly on the publisher server. All entries that are
made using the Cisco CallManager Administration page of the subscriber are written to the
database on the publisher server. If the publisher is down, no updates can be made in the Cisco
CallManager Administration page of the subscriber server.
CDR records are not written into a database of the subscriber, but they are written locally in
transaction files at the subscribers (but not directly in the CDR database). Those files are
periodically processed and inserted in the Microsoft SQL Server 2000 database by the CDR
Insert service on the publisher server. If the publisher server is down, transaction files reside on
the subscriber server to be processed as soon as the publisher is available again. After CDRs are
processed, they are replicated from the publisher to subscriber database.
674 Chapter 30: Introducing Database Tools and Cisco Unified CallManager Serviceability
When you are building a publisher server, Cisco CallManager software itself might or might not
be installed. If the publisher server does not have Cisco CallManager installed (and only has the
SQL database installed), Cisco refers to the server as a glass house. This configuration is the best
solution in large clusters. The publisher (either running Cisco CallManager or not [glass house])
holds the only copy of the database that is allowed to be written to. All subscribers replicate with
the database of the publisher only, so they are in read-only mode.
In addition, the publisher occasionally acts as a backup CallManager service for the configuration.
This configuration is typically used only in smaller clusters.
Database Management Tools Overview

Two tools are used to maintain the Cisco CallManager database services and are based on
Microsoft SQL Server 2000:
■ Microsoft SQL Server 2000 Enterprise Manager—Provided by Microsoft and included in

each Microsoft SQL Server 2000 installation
■ DBLHelper—Provided by Cisco and needs to be requested from Cisco Technical Assistance

Center (TAC) to resolve Cisco CallManager database issues
Both tools allow administrators to verify proper working of Cisco CallManager Microsoft SQL
Server 2000 databases. You can use the Microsoft SQL Server 2000 Enterprise Manager and
DBLHelper when administrators need to examine the database structure and replication. You can
use these tools to reactivate broken database connections.
In many cases, the reason for a broken database connection is publisher downtime. For example,
administrators might take the publisher off the network to perform a software upgrade and restore
it later. After reloading a Cisco CallManager publisher server, verify the database connection. A
broken connection would not cause any obvious problem in system activity. Usually, users become
aware of a broken connection only if the publisher becomes unresponsive again and any system
changes are lost. For example, call forwarding instructions that are months out of date are active
or newly added phones are unavailable when they failover to a subscriber server. Failures of the
Extension Mobility functionality might also point to a broken publisher connection.
When you are troubleshooting Cisco CallManager SQL databases, it is beneficial to use both tools
together.
Microsoft SQL Server 2000 Enterprise Manager

Microsoft SQL Server 2000 Enterprise Manager is included with the Microsoft SQL Server 2000
Enterprise software. To access it, choose Start > Programs > Microsoft SQL Server >
Enterprise Manager. This is the primary administrative tool for Microsoft SQL Server 2000 and
provides a Microsoft Management Console (MMC)–compliant user interface, shown in Figure 30-1,
which allows users to do the following:
■ Define groups of servers running SQL Server.
■ Register individual servers in a group.
■ Configure all SQL Server options for each registered server.
■ Create and administer all SQL Server databases, objects, logins, users, and permissions in
each registered server.
■ Define and execute all SQL Server administrative tasks on each registered server.
■ Design and test SQL statements, batches, and scripts interactively by invoking SQL Query
Analyzer.
■ Invoke the various wizards defined for SQL Server.
Figure 30-1 Microsoft SQL Server 2000 Enterprise Manager

NOTE Because the Cisco CallManager installation includes the database and database
structure needed for Cisco CallManager servers, it is not necessary to create new databases.
CAUTION Do not change the structure of the Cisco CallManager database. The Cisco
CallManager will malfunction or, in the worst case, stop responding.
The Microsoft SQL database name of Cisco CallManager is CCM03xx, where xx starts at 00 and
increases incrementally with each Cisco CallManager upgrade (for example, upgrading from
release 4.0 to 4.1).
You can use Microsoft SQL Server 2000 Enterprise Manager to determine whether a server is the
publisher or the subscriber. Expand the database hierarchy down to the database—Microsoft SQL
Servers\SQL Server Group\<Server_Name>\Databases\CCM03xx, as shown in Figure 30-2:
■ For a publisher, a Publications folder is displayed in the Database browse list.
■ For a subscriber, a Pull Subscriptions folder is displayed in the Database browse list.
Figure 30-2 Determining Publisher Server Function with SQL Server 2000 Enterprise Manager
TIP The second way to determine whether the server is the publisher or subscriber is to check
whether the Replication Monitor is present on the specific Microsoft SQL server. The
Replication Monitor is present only on the publisher and monitors the status of database
replication between the publisher and the subscribers.
DBLHelper
In the event that the subscriber stops replicating data from the publisher, you need to rebuild the
relationships between the publisher and subscriber. The DBLHelper utility, a tool provided by
Cisco, republishes or reinitializes a nonfunctioning subscription between the publisher and the
subscriber databases. For DBLHelper to work, the SQL account password and administrative
rights must be the same on the publisher and the subscriber. Because this tool is automated and
nonintrusive, you might make a schedule of running it once per week to ensure the subscriber and
publisher database links are active. If the link fails, you might not detect it for a considerable time
because the symptoms are so subtle.
NOTE If you are upgrading from an earlier version of Cisco CallManager, DBLHelper.exe is
usually located on the Cisco CallManager server in the C:\Program Files\Cisco Systems,
Inc\Cisco CallManager Upgrade Assistant\DbReplCheck folder. Otherwise, contact Cisco TAC
for the latest version of DBLHelper.
You can also use Microsoft SQL Server 2000 Enterprise Manager to re-create broken database
connections between Cisco CallManager servers in the cluster, but this process is far more
complex than using DBLHelper. The recommendation is to first try repairing the replication using
DBLHelper. If DBLHelper cannot fix the problem, use the Microsoft SQL Server 2000 Enterprise
Manager. For more information on how to use the Microsoft SQL Server 2000 Enterprise Manager
to repair replications, search for “Reestablishing a Broken Cisco CallManager Cluster SQL
Subscription” or “Broken SQL Replication” on Cisco.com. The documents for the 3.x version of
Cisco CallManager also apply to the 4.x CallManager versions.
In Figure 30-3, DBLHelper shows two different states of Cisco CallManager Microsoft SQL
Server 2000 databases. On the left, running DBLHelper.exe found that the replication was
working. This status is depicted by the two green smiley face icons. On the right, a red (lower) sad
face icon indicates that databases are out of synchronization. In this case, database connections
need to be reestablished. Clicking the Republish button deletes the current subscription and re-
creates it. Clicking the Reinitialize button reinitializes all subscriptions and starts the snapshot
agent. It also begins an attempt to rebuild the subscription with the current database. To get more
information about the current tab, click the Help button, which shows some additional
information.
NOTE If there is more than one subscriber and only one has a broken database connection,
use the Republish button to republish only the broken connection. If you use the Reinitialize
button, all subscriptions will be reinitialized, which could cause a long system outage.
Figure 30-3 Using DBLHelper to Restore Broken Links
Cisco CallManager Serviceability Overview

The Cisco CallManager Serviceability tool is available on each Cisco CallManager. It is the main
tool used by administrators to maintain the Cisco CallManager installation. To access the Cisco
CallManager Serviceability tool, choose the Application menu in Cisco CallManager
Administration, choose Start > Programs > Cisco CallManager 4.1 > Cisco Service
Configuration from the Cisco CallManager system, or use a browser to go directly to https://
<CallManager_IP>/CCMService.
Serviceability Components
Cisco CallManager Serviceability provides these menus, as shown in Figure 30-4:
■ Alarm
■ Trace
■ Tools
■ Application
■ Help
Figure 30-4 Cisco CallManager Serviceability Interface
Alarm
The Alarm service stores information about Cisco CallManager service events for troubleshooting
and provides alarm message definitions. Alarms can be forwarded to trace files, Microsoft
Windows 2000 Event Viewer, and a Syslog server for further analysis:
■ With the Configuration menu item shown in Figure 30-5, the Cisco CallManager
Serviceability alarms allow configuration of Cisco CallManager to write an event to a trace
file or the Windows 2000 Event Viewer when an incident occurs, such as the failure of a
telephone to register. Alarms for Cisco CallManager servers can be configured in a cluster or
for the services in each server.
■ The Definitions application contains alarm definitions and the recommended actions in a
Microsoft SQL Server 2000 database. The system administrator can search the database for
the definitions of all alarms. Definitions include the alarm name, description, recommended
action, severity, parameters, and monitors.
Figure 30-5 Serviceability Alarm Menu
Trace
The Trace service allows you to save detailed logs of Cisco CallManager events for
troubleshooting system problems. Trace data can be configured, collected, and analyzed using the
menu options shown in Figure 30-6:
■ Use the Configuration application to specify the trace parameters; for example, the Cisco
CallManager server within the cluster, the Cisco CallManager service on the server, the debug
level, and the specific trace fields.
■ Use the Analysis application to provide greater trace detail on a signal distribution layer
(SDL) trace, a system diagnostic interface (SDI) trace, a Cisco CallManager service type, or
the time and date of a trace. You can choose a specific log file from the list and choose
information from that log file, such as host address, IP address, trace type, and request a
device name.
■ The Q.931 Translator application filters incoming data from Cisco CallManager SDI logs and
translates the data into Cisco IOS messages. The Q.931 Translator application displays the
message in the message translator interface.
■ Use the Troubleshooting Trace Settings application to choose the services in Cisco
CallManager for which troubleshooting trace settings, which are predetermined in the alarms
configuration, need to be set.
Figure 30-6 Serviceability Trace Menu
Tools
The Tools service offers these applications, shown in Figure 30-7:
■ The CDR Analysis and Reporting (CAR) application supports analysis and reporting of
CDRs. CAR generates reports for quality of service (QoS), traffic, and billing information.
NOTE The CDR Analysis and Reporting menu item will only appear if you have installed the
CAR plug-in for Cisco CallManager. It is not installed by default.
■ The Service Activation application can activate and deactivate nearly all of the Cisco
CallManager services for all Cisco CallManager servers.
■ The Control Center application allows starting, stopping, restarting, and viewing the status of
Cisco CallManager services.
■ The Real-Time Monitoring Tool (RTMT) application monitors the real-time behavior of most
components in a Cisco CallManager cluster and displays on-screen feedback through a Java-
based application.
■ The QRT Viewer application allows filtering, formatting, and viewing problem reports. Cisco
IP Phones can be configured with a Quality Report Tool (QRT) softkey so that users can
report problems with IP Phone calls (for example, poor quality). When users press the QRT
softkey on the IP Phone, they are presented with a list of problem categories. Users can then
choose the appropriate problem category, and the system logs the event in an Extensible
Markup Language (XML) file.
■ The Serviceability Reports Archive application generates five daily reports in Cisco
CallManager Serviceability: Device Statistics, Server Statistics, Service Statistics, Call
Activities, and Alerts. Each report provides a summary that consists of various charts that
display the statistics for that particular report.
Figure 30-7 Serviceability Tools Menu

Application
The Application menu in Cisco CallManager Serviceability, shown in Figure 30-8, offers several
applications:
■ The Install Plugins application can help to extend the functionality of Cisco CallManager by
providing links to software plug-ins that are either installed in Cisco CallManager itself or on
a client PC.
■ The Cisco CallManager Administration menu item provides a convenient, direct link to the
Cisco CallManager Administration page.
■ The Bulk Administration Tool (BAT) application adds multiple telephones and users to Cisco
CallManager and performs bulk modifications.
NOTE The Bulk Administration Tool (BAT) menu item will only appear if you have installed
the BAT plug-in for Cisco CallManager. It is not installed by default.
Figure 30-8 Serviceability Application Menu

Help
The Help menu, shown in Figure 30-9, provides online assistance for every option in Cisco
CallManager Serviceability. Moreover, it can be used to display the latest installed component
version information for all Cisco CallManager servers in the cluster.
Figure 30-9 Serviceability Help Menu
Control Center
On the Control Center, accessed from the CallManager Serviceability Tools menu, Cisco
CallManager services can be started, stopped, or restarted. For example, restarting a Cisco
CallManager service could be necessary if you change central Cisco CallManager functionalities,
such as intercluster trunks or intersite bandwidth settings. To start, stop, or restart a service, first
select the service and then start, stop, or restart it by clicking the appropriate button, as shown in
Figure 30-10.
Figure 30-10 Using the Control Center
The Status column shows whether a service is started (a right arrow in the Status column) or
stopped (a square in the Status column). In addition, the Activation Status column tells how the
service is configured. A status of Activated or Deactivated indicates whether the service is
configured to be started at Windows startup.
NOTE Services with a status shown as Deactivated and Started are not activated in Service
Activation but are still started at Windows startup.
You cannot start Cisco Tomcat web services using the Control Center. To start them, you could use
the Microsoft Windows 2000 Server Services Management Console. In addition, with Cisco
CallManager Release 4.0 or later, Cisco Tomcat Web Application Manager allows you to start,
stop, or restart Tomcat services. To access Tomcat Web Application Manager, go to http://
<CallManager_IP>/manager/list.
Starting and stopping the Cisco CallManager service causes all Cisco IP Phones and gateways that
are currently registered to that Cisco CallManager service to failover to their secondary Cisco
CallManager service. In addition, starting and stopping the Cisco CallManager service causes
other installed applications (such as Conference Bridge or Cisco Messaging Interface) that are
homed to that Cisco CallManager to start and stop as well. Stopping the Cisco CallManager
service also stops call processing for all devices that are controlled by that service. When a Cisco
CallManager service is stopped, active calls from an IP Phone to another IP Phone continue; calls
in progress from an IP Phone to a Media Gateway Control Protocol (MGCP) gateway also
continue, whereas other types of calls, such as calls to an H.323 gateway, are dropped.
If you are upgrading Cisco CallManager, all services are stopped during the upgrade process.
Services that had been started before the upgrade began are started again afterward. Those that had
not been started remain deactivated. Service configurations are not lost during the upgrade
process.
Service Activation
Two tools allow you to manage services running on the Cisco CallManager system:
■ The Cisco CallManager Serviceability Service Activation tool, shown in Figure 30-11, is used
to activate or deactivate specific services on Cisco CallManager.
Figure 30-11 Managing Services Using the CallManager Service Activation Tool
■ The Services MMC, shown in Figure 30-12, is used to handle common Windows services.
The Services MMC can be found at Start > Settings > Control Panel > Administrative
Tools on the Cisco CallManager system.
Figure 30-12 Managing Services Using the Windows 2000 Service Utility
To optimize the performance of Cisco CallManager servers within a cluster, you can distribute
services to servers across the cluster. For example, place the TFTP server service on one server,
the Certificate Authority Proxy Function (CAPF) service on another server, and music on hold
(MoH) services on a third server where no Cisco CallManager service is running, and so on.
Distribution of services can also be used for security purposes. Some services (such as those that
provide IP Phone services with access to the Internet) must be exposed to the outside for proper
operation, whereas others, such as the Cisco CallManager service, should not be exposed. If the
exposed server is under attack, the attack would not have any impact on the other servers (for
instance, the server routing calls that is running the Cisco CallManager service).
NOTE If the Cisco CallManager and Cisco CTIManager services are deactivated in the
Service Activation window, the Cisco CallManager where the service was deactivated no longer
exists in the database. This means that the Cisco CallManager cannot be chosen for
configuration operations in Cisco CallManager Administration because it will not be displayed
in the GUI.
If the services are then reactivated on the same Cisco CallManager, the database re-creates the
Cisco CallManager and adds a “CM_” prefix to the server name or IP address; for example, if
the Cisco CallManager or CTIManager service is reactivated on a server with an IP address of
10.192.5.97, then “CM_10.192.5.97” is displayed in Cisco CallManager Administration. The
Cisco CallManager with the new “CM_” prefix can now be chosen in Cisco CallManager
Administration.
CAUTION When you deactivate the Cisco CallManager service, the server is removed from
Cisco CallManager groups but not added back after reactivation. So after activation or
reactivation, you must add the server to Cisco CallManager groups, otherwise no devices will
register to it.
Tools Overview
Several tools make administration easier when you are managing Cisco CallManager system
status or configuration. Some of those tools are provided by Cisco Systems, whereas others are
Microsoft tools included in the Windows 2000 Server operating system used by Cisco
CallManager. Table 30-1 provides an overview of these tools.
Table 30-1 CallManager Management Tools
Management Tool Function Device

Microsoft SQL Server 2000 Verify proper working of Any Microsoft SQL Server
Enterprise Manager databases 2000 server in the domain
DBLHelper Verify proper replication Cisco CallManager publisher
between databases server
Services MMC Manage services Actual Windows 2000 server
Microsoft Event Viewer Identify system-level events Actual Windows 2000 server
and errors
Microsoft Performance Monitor system and device Actual Windows 2000 server
Monitor statistics
RTMT Client Monitor Cisco CallManager Any PC on the network
system in real time
Password Changer Tool Change the CCMAdministrator Cisco CallManager publisher
password server
Tools Overview 689
Table 30-1 CallManager Management Tools (Continued)
Management Tool Function Device

CAR Tool Analyze Cisco CallManager Cisco CallManager Web page
CDRs via browser
Dialed Number Analyzer Analyze calls in a Cisco Cisco CallManager web page
CallManager dial plan via browser
QRT Create quality problem reports Cisco CallManager web page
via browser
The following is a brief description of the purpose of each of these tools:
■ Microsoft SQL Server 2000 Enterprise Manager can be used to verify the proper working of
the databases and to evaluate the Microsoft SQL environment. This tool can be used on any
Windows 2000 server in the domain to access the SQL database. By default, it is installed on
every server that has Microsoft SQL Server 2000 installed.
■ DBLHelper allows the verification and re-creation of the replication process between the
Cisco CallManager publisher and subscriber databases. This tool needs to be run on the Cisco
CallManager publisher server. Beginning with CallManager Release 4.0, this tool is available
only from Cisco TAC.
■ The Services MMC allows for management of services on Windows 2000 Server. Each
Windows 2000 server has its own Services MMC preinstalled. Even though administrators
can create their own MMC to manage services on any other system, it is recommended that
you use the preinstalled MMC on each server locally. Otherwise, you risk confusion, because
all the servers look the same.
■ Microsoft Event Viewer allows identification of system-level events and errors. Each event
and error of the Windows 2000 system is stored in Microsoft Event Viewer. Different kinds
of messages are grouped into a system log, an application log (where most information related
to Cisco CallManager is stored), and a security log. Because Event Viewer is also an MMC
snap-in, an MMC that includes Event Viewer can be created on any system but it is not
recommended to manage remote systems.
■ Microsoft Performance Monitor shows system and application statistics and is available on
every Microsoft Windows 2000 system. Although it is also an MMC snap-in for Microsoft
Windows 2000 Server, it can be used only on the server itself because it uses an ActiveX
control. To monitor performance statistics of remote systems, use the local snap-in as an
alternative and include remote counters.
■ The RTMT client allows you to monitor Cisco CallManager activities in real time. This Java-
based tool can be installed on any Windows PC. It is available in the Install Plugins window
of Cisco CallManager Serviceability.
■ The Password Changer tool available on the Cisco CallManager server allows changing the
administrator passwords on Cisco CallManager systems. This tool needs to be run on the
Cisco CallManager publisher server.
■ The CAR tool allows analysis of CDRs written on Cisco CallManager. CAR is available from
Cisco CallManager Serviceability and can be accessed using the Microsoft Internet Explorer
web browser on any PC on the network.
■ Dialed Number Analyzer analyzes how inbound and outbound calls are handled in the Cisco
CallManager call-routing configuration. It resides on the Cisco CallManager web server and
can be accessed using Internet Explorer on any PC on the network.
■ The QRT Viewer application allows you to filter, format, and view problem reports. It is
located on the Cisco CallManager web server and can be accessed using Internet Explorer on
any PC on the network.
NOTE Many of these applications are discussed in depth in Chapters 31 and 32, “Monitoring
Performance” and “Configuring Alarms and Traces,” respectively.
Summary
This chapter discussed the tools included with the Cisco CallManager software. You can use
Microsoft SQL Server 2000 Enterprise Manager and DBLHelper to verify proper working of the
SQL database and repair broken links, if necessary. You can use the tools in the Cisco CallManager
Serviceability system to control and manage the CallManager system. You can use the Control
Center to start, stop, or restart service related to the Cisco CallManager. The Service Activation
and Services Windows 2000 MMC allow you to manage all services on Cisco CallManager. Many
additional tools are provided by Cisco and Microsoft that allow management and troubleshooting
of the Cisco CallManager server.
Review Questions
You can find the solutions to these questions in Appendix A, “Answers to Review Questions."“
1. Which two tools are available on Cisco CallManager systems to manage the SQL database?
a. Microsoft SQL Server 2000 Enterprise Manager
b. Cisco SQL Server 2000 Enterprise Manager
c. SQL Server 2000 System Manager
d. DBAHelper
e. DBLHelper
2. Which of the following statements is true?
a. Cisco CallManager Serviceability needs to be installed from the Cisco CallManager
Install Plugins window.
b. Cisco CallManager Serviceability provides a configuration interface to configure
extension DNs.
c. Cisco CallManager Serviceability allows you to start, stop, or restart all Microsoft
Windows 2000 services.
d. Cisco CallManager Serviceability provides services to monitor alarms, generate
CARs, and collect and analyze traces.
3. Which two of the following services can be started, stopped, or restarted via the
Cisco CallManager Control Center?
a. Cisco CallManager
b. Cisco DHCP
c. Cisco TFTP
d. Cisco Extension Mobility
e. Cisco WebDialer
4. Where can Cisco CallManager Services be activated and deactivated? (Choose two.)
a. In DBLHelper
b. In Cisco CallManager Control Center
c. In the Windows 2000 Services MMC
d. In Microsoft Service Activation Tool
e. In Cisco CallManager Service Activation
5. Where should DBLHelper be run?

a. Only on the Cisco CallManager publisher server
b. Only on the Cisco CallManager subscriber server
c. On the Cisco CallManager publisher or subscriber server
d. On any PC in the network
6. If the publisher server is down, where are configuration changes stored in the CallManager
cluster?
a. Configuration changes will be lost.
b. Configuration changes are stored in the subscriber database file.
c. Configuration changes are stored in the subscriber transaction files.
d. The CallManager will prevent you from making configuration changes while the pub-
lisher is down.
7. If the publisher server is down, where are CDRs stored in the CallManager cluster?
a. CDRs will be lost.
b. CDRs are stored in the subscriber database file.
c. CDRs are stored in the subscriber transaction files.
d. The CallManager cluster will stop all calls while the publisher is down.
8. You are using SQL Server 2000 Enterprise Manager on one of your CallManager servers.
Which two items should you see if this server is the cluster publisher?
a. The existence of the Replication Monitor
b. A publisher image at the top of the database
c. A Publications folder under the database
d. The title bar of the SQL Enterprise Manager will show “Publisher” next to the title
9. Which utility does Cisco highly recommend you use to repair SQL database replication
issues?
a. Windows 2000 Services
b. DBLHelper
c. SQL Server 2000 Enterprise Manager
d. Serviceability Control Center
10. If replication has failed between a publisher and subscriber server, what does DBLHelper
display?
a. A replication failed notice
b. A broken link
c. A red sad face
d. A yellow exclamation point
following topics:
■ Defining a performance object and counter

and describing their interaction with
Microsoft Performance Monitor and RTMT
■ Using Microsoft Event Viewer to identify
system-level events and errors
■ Using Microsoft Performance Monitor to
monitor system and device statistics
■ Explaining the major monitoring categories
that RTMT provides
■ Using the default RTMT configuration and
creating customized configuration profiles
■ Identifying components of the RTMT
window
CHAPTER 31
Monitoring Performance
Growing demands to the telephony systems increase hardware usage on the telephony system
platform. If those demands rise too much, they could cause system overloads. Telephony
systems are among those that most directly affect businesses, and overloads that lead to system
outages could be extremely costly. Therefore, administrators must be able to monitor system
performance. This chapter covers tools that are used to monitor Cisco CallManager systems.
Further, this chapter describes Microsoft tools that are available on the Windows system, Real-
Time Monitoring Tool (RTMT) from Cisco, and how they work together.
Performance Counters
Performance counters reflect the performance data of Cisco CallManager. A counter is a
variable whose name is stored in the Registry. Each counter is related to a specific area of system
functionality. Examples include busy time of the processor, memory usage, and the number of
bytes received over a network connection. Each counter is uniquely identified through its name
and its path or location. In the same way that a file path includes drives, directories,
subdirectories, and filenames, a counter path consists of four elements: the machine, the object
(for example, processor or IP), the object instance (type of counter value; for example,
interrupt), and the counter name (special counter itself).
Performance counters are ideal for administrators for system maintenance, analysis, and
troubleshooting tasks:
■ An administrator needs to reset a voice gateway. With performance counters, it is possible

to watch the system until the last call is disconnected and then reset the gateway.
■ A user reports that using Cisco CallManager Extension Mobility to log in to the phone takes
a very long time. In analyzing the statistics produced with performance counter data, the
administrator discovers high processor usage and memory allocation due to problems with
processes on the system.
■ An administrator is dealing with a slow system. The system engineer has to decide whether
system expansion is necessary or whether the current extensive system usage is only a one-
time situation. To find out, the administrator should watch the system for a while.
696 Chapter 31: Monitoring Performance
NOTE All performance counter values are based on system events and utilization information
provided by the Microsoft Windows 2000 platform.
Performance Analysis
Microsoft Performance Monitor and Cisco Real-Time Monitoring Tool (RTMT) use Windows
2000 performance counters to monitor the system. Microsoft Performance Monitor, shown in
Figure 31-1, reports both general and specific information in real time, whereas RTMT, shown in
Figure 31-2, monitors focused Cisco CallManager performance by periodically polling Windows
2000 performance counter values.
Figure 31-1 Microsoft Windows 2000 Performance Monitor
RTMT provides optimized monitoring of performance objects and devices related just to the Cisco
CallManager. The device information includes device registration status, IP address, description,
and model type. RTMT provides clusterwide information that is stored in eight tables. The tables
include IP phones, gateway devices, media, H.323 devices, Session Initiation Protocol (SIP)
trunks, hunt lists, computer telephony integration (CTI), and voice messaging. RTMT also
displays object and counter information that is kept by each Cisco CallManager node in the
cluster. RTMT directly monitors the performance objects and counters.
Microsoft Event Viewer 697
Figure 31-2 Cisco Real-Time Monitoring Tool
Microsoft Event Viewer

Microsoft Event Viewer is a Microsoft Windows system tool. It is available by default on every
Windows NT-based system (Windows NT, Windows 2000, Windows XP, and Windows Server
2003). Beginning with the introduction of Windows 2000, Microsoft Event Viewer has been
managed through the Microsoft Management Console (MMC). It is possible to create an MMC
that includes the Event Viewer of each remote system. But keep in mind that using more than one
Event Viewer on one system often leads to confusion and is normally not recommended.
Windows stores information related to an event or error in Event Viewer. Event Viewer can be used
to gather, identify, and analyze Windows 2000 system events and information about hardware,
software, and system problems. Because Cisco CallManager runs as a Windows service,
information about Cisco CallManager events and errors can also be stored in Event Viewer.
Event Viewer, available at Start > Settings > Control Panel > Administrative Tools > Event
Viewer, can help identify problems at the system level. For example, to pinpoint a problem, use
Event Viewer to look for events involving Cisco CallManager Administration on the Internet
Information Server (IIS) server.
As shown in Figure 31-3, Event Viewer uses log types to group different kinds of logs:
■ Application logs—Contain events logged by applications or programs, such as Cisco

CallManager.
■ System logs—Report events logged by the Windows 2000 system components, such as the
failure of an operating system component or driver.
■ Security logs—Hold information records of security events. (Cisco CallManager does not
report events in this log.)
Figure 31-3 Microsoft Event Viewer
To classify events, Event Viewer provides three event types. Each is marked with its own icon. The
Microsoft Event Viewer event types are as follows:
■ Error—An indicator of a problem, such as the loss of data or failure to initialize properly;
represented by a red “X”
■ Warning—An event that might indicate a problem or a future problem, such as when a
service is stopped or started, which is not necessarily an error; represented by a yellow
exclamation point
■ Information—System information messages that might include hostnames, the version of

the database in use, or startup success; represented by a blue “i” symbol
Microsoft Performance Monitor

Microsoft Performance Monitor is a Windows 2000 Server application that uses Windows
2000 performance counters to create statistics. It displays the activities and status of the system.
Performance Monitor reports both general and specific information in real time. You can
use Performance Monitor to collect and display system and device statistics for any Cisco
CallManager installation.
Performance Monitor, like the Cisco CallManager Serviceability tool, monitors and logs resource
counters from the Cisco CallManager nodes in the network and displays the counters in real time.
The update interval could be set to a period between 1 second and 45 days.
Performance Monitor can collect data from multiple systems at once and store it in a single log
file. The monitor can then export this log file to a Tab-Separated Values (TSV) file or a Comma-
Separated Values (CSV) file that you can view in most spreadsheet applications.
To access Performance Monitor, choose Start > Settings > Control Panel > Administrative
Tools and choose Performance.
NOTE Extensive monitoring (monitoring a very large number of counters and using short
interval times) could cause rising processor usage. Therefore, it is recommended that you use
Performance Monitor only for special situations and not as a network management tool.
Microsoft Performance Monitor Report View

Microsoft Performance Monitor needs to be customized for the parameters related to monitoring
Cisco CallManager by choosing the objects, counters, and instances to include. As shown in
Figure 31-4, Performance Monitor supports a very concise report view. Many administrators
prefer this view because it gives exact figures rather than general estimates on a line chart.
Within Performance Monitor, alarms can be enabled to report certain value thresholds. For
example, the number of telephone devices active on Cisco CallManager can be set to a particular
level. If the number of devices exceeds that level, the monitor sends a network message alert to the
administrator or the person in charge. To create such an alert threshold, right-click Performance
Logs and Alerts > Alerts and choose New Alert Settings.
NOTE Data collection must be enabled in Cisco CallManager for Performance Monitor to
collect data. To verify that data collection is enabled, check the current settings in the Cisco
Real-Time Information Server (RIS) Data Collector Service Parameters Configuration window
in Cisco CallManager Administration. By default, the Data Collection Enabled parameter is
set to True.
Figure 31-4 Microsoft Performance Monitor Report View
Microsoft Performance Monitor Graph and Histogram View

Microsoft Performance Monitor provides graph and histogram graphical views for a visual
overview of how the system is currently in use and how it changes.
The graph view (shown in Figure 31-5) is ideal to see the current status of the counters and how
they have changed over the most recent period. You can specify the period for status updates and
include it in the graph. Using this tool is similar to using a heart-rate monitor. It displays the
heartbeat of the monitored performance objects.
In contrast to graphs, the histogram view (shown in Figure 31-6) shows only the current status of
the selected performance object counters. Because it uses only a single bar for each counter, the
histogram view is ideal for monitoring many performance counters. To switch between the graph
and histogram views, use the graph and histogram icons from the system monitor toolbar or press
Ctrl-G (graph) and Ctrl-B (histogram) alternately.
Figure 31-5 Microsoft Performance Monitor Graph View
Figure 31-6 Microsoft Performance Monitor Histogram View

Microsoft Windows Task Manager

In addition to Microsoft Performance Monitor, you can use Windows Task Manager (shown in
Figure 31-7) to get a fast view of the base performance values of CPU and memory usage. The
Performance tab of Task Manager shows the CPU and memory usage of the system as a graph
and histogram. Moreover, it provides some additional counter values, such as paged kernel
memory in plain text. It is an ideal tool to get a quick impression of system health and activity, for
example, if the system is working very slowly.
Figure 31-7 Microsoft Windows Task Manager
To access Windows Task Manager, right-click the taskbar and choose Task Manager from the
menu or press the Ctrl-Shift-Esc shortcut.
NOTE Windows Task Manager influences system performance because it also monitors all
system processes and running programs. Therefore, it is recommended that you use it only to
get a snapshot of the current situation and not for prolonged monitoring.
Real-Time Monitoring Tool Overview

With Cisco RTMT, the Cisco CallManager Serviceability tool provides a client-side, standalone
Java plug-in that monitors real-time behavior of the components in a Cisco CallManager cluster.
RTMT monitors a set of management objects by continuously polling the Windows 2000
performance counter values. It can generate various alerts in the form of e-mails for values that
exceed or fail to reach user-configured thresholds. RTMT can also generate daily reports for these
objects. The e-mail alerts work independently of the standalone Java front end. Therefore, it is not
necessary to run the RTMT window all the time to generate e-mail notifications.
To download RTMT, choose Cisco CallManager RTMT from the Cisco CallManager Install
Plugins page in the Application menu of Cisco CallManager Administration and Cisco
CallManager Serviceability. After successful installation, you can launch RTMT by double-
clicking the RTMT shortcut on the Windows desktop or by choosing Start > Programs > Cisco
CallManager Serviceability > RTMT.
NOTE To reduce the impact on the Cisco CallManager server, it is strongly recommended that
you run RTMT on an administrator PC and not locally on the Cisco CallManager server.
Moreover, RTMT should not run constantly.
As shown in Figure 31-8, The RTMT application is divided into four parts:
■ Menu bar—The RTMT menu bar, located at the top of the RTMT window, uses drop-down
menus that provide specific monitoring.
■ Controlling center pane—The controlling center pane, located on the left side of the
window, includes the View tab and the Tools tab. The View tab includes several different
monitoring categories, whereas the Tools tab offers only the Alert category. On the View tab
of the controlling pane, RTMT arranges the preconfigured monitoring objects into seven
major categories:
— Summary—The Summary category shows the activities of several predetermined

monitoring objects in the Cisco CallManager cluster, such as memory and CPU
usage, registered phones, calls in progress, and active gateway ports and channels.
— Device—The Device category monitors phones, gateways, and media devices on
each Cisco CallManager node and cluster. Monitored objects include the number of
registered phones, gateways, and media resources for each Cisco CallManager node
and cluster.
— CallProcess—The CallProcess category monitors all call activity for each Cisco
CallManager node and cluster. Monitored objects include CallsAttempted,
CallsCompleted, and CallsInProgress for each Cisco CallManager node and cluster.
— Server—The Server category monitors CPU usage, disk space usage, and critical
services on each Cisco CallManager server. Monitored objects include CPU and
memory usage on each server and the disk space usage for all logical drives on each
server.
— Service—The Service category monitors CTIManager information for each

CTIManager, Cisco TFTP server information, directory server information, and
heartbeat rate information. Monitored objects include TotalTftpRequests and
TotalTftpRequestsAborted.
— CTI—The CTI category provides CTI search and a CTIManager monitoring
window that displays the number of open devices, lines, and CTI connections.
— Performance—The Performance category provides performance monitoring
similar to Microsoft Performance Monitor.
■ Viewing pane—In the viewing pane, located to the right of the controlling center pane and
occupying most of the window area, RTMT displays the currently selected monitoring object.
■ Tab bar—The tab bar, located at the bottom of the window, allows switching among all
monitoring objects currently monitored in the viewing pane.
Figure 31-8 Cisco Real-Time Monitoring Tool
Real-Time Monitoring Tool Configuration Profiles

RTMT allows you to save monitoring configurations so that they do not need to be re-created every
time you start RTMT. When you start RTMT, you can select the desired profile. The CM-Default
profile is preconfigured. Its parameters include the number of registered phones, gateway status,
CPU and memory usage, and the number of calls in progress.
To save the current monitoring configurations:
Step 1 Choose System > Profile (or press Ctrl-Alt-P) to open the Preferences
window.
Step 2 In the Preferences Window, choose Save. The Save Current Configuration
window opens.
Step 3 Enter a name and a meaningful description for the configuration, as shown
in Figure 31-9, which will allow you to easily identify it later, and click OK
to save your configuration.
Figure 31-9 Configuring Real-Time Monitoring Tool Configuration Profiles
To make a copy of an existing configuration:
Step 1 Choose System > Profile (or press Ctrl-Alt-P) to open the Preferences
window.
Step 2 Select the configuration that should be duplicated.
Step 3 Click Restore to load the configuration.
Step 4 Click Save, enter a name and description for the configuration, and click
OK.
Using the Real-Time Monitoring Tool Effectively

The RTMT window monitors various aspects of Cisco CallManager performance by periodically
polling Windows 2000 performance counter values. RTMT discovers devices regardless of their
registration status, such as registered or failed, in the cluster. The tool searches by device name,
device description, IP address, IP subnet, or directory number (DN) and monitors the status of
discovered devices.
Some of the most useful monitoring objects on the RTMT window are these:
■ Summary—Gives administrators a quick overview of their Cisco voice over IP (VoIP)

system; this is the default view
■ Call Activity—Shows active calls as well as attempted and completed calls
■ Device Search—Allows administrators to define criteria for identifying which devices to
display for analysis
■ Perfmon—Allows system performance to be displayed within the RTMT window, which
eliminates the need for a Microsoft Performance Monitor window to be running at the same time
■ Alert Central—Displays current status and history of all the alerts in the Cisco CallManager
cluster
Viewing Call Activity

As shown in Figure 31-10, the Call Activity window allows you to monitor call activity for each
Cisco CallManager node in the cluster in real time. To display the Call Activity window, complete
these tasks:
Step 1 In the controlling center pane at the left, click the View tab.
Step 2 Click CallProcess.
Step 3 Click the Call Activity icon.
Figure 31-10 RTMT Call Activity

The Call Activity monitoring window displays the call activity for each Cisco CallManager node
in the cluster and draws a history graph for the past few minutes.
Using Device Search

You can select the devices to monitor based on shared characteristics, such as a status of registered
or rejected. RTMT device monitoring can be further refined to narrow the search characteristics to
DNs or subnet. In addition, you can select the attributes to monitor, such as Node or StatusReason,
and all others can be left out. Select only certain values to monitor because doing so saves time
and eliminates unnecessary data in the monitor window. To access these configuration parameters,
select the device for which you want to view monitoring information. The monitor then prompts
you to enter the characteristics to narrow the search, as shown in Figure 31-11.
Figure 31-11 RTMT Device Search Wizard
To create a search, complete these tasks:
Step 1 In the controlling center pane at the left, click the Device tab.
Step 2 Click Device Search.
Step 3 Double-click Phone in the white pane and follow the instructions.
Visiting Alert Central

In RTMT, you can configure alert notification for Perfmon counter value thresholds and schedule
alert checks and status changes of devices (for example, a port is out of service). The Tools tab in
the controlling center pane of the RTMT monitor includes the Alert Central category, as shown in
Figure 31-12. Alert Central provides both the current status and the history of all the alerts in the
Cisco CallManager cluster.
Figure 31-12 RTMT Alert Central
Figure 31-12 shows a sample alert for a system where the virtual memory of the system is
currently too low. To allow administrators to qualify the message (to determine whether this is a
one-time event because of a special situation or a continuing problem), RTMT also provides a time
stamp of the most recent memory leak alert and a history of the most recent messages. In addition,
you can configure the RTMT Alert to send e-mail alerts when the CallManager exceeds predefined
thresholds. This configuration requires you to enter Simple Mail Transfer Protocol (SMTP) server
information using the Alert > Config Email Server menu item.
Summary
This chapter discussed the concepts and configuration behind monitoring the Cisco CallManager
cluster. Both Microsoft and Cisco include tools to monitor various aspects of each Cisco
CallManager server or the cluster as a whole. The Microsoft Performance Monitor shows system
parameters, including Cisco CallManager counters. The Microsoft Event Viewer can help you
identify problems at the system level. The Event Viewer also records events generated from the
Cisco CallManager software.
Cisco Real-Time Monitoring Tool (RTMT) is a standalone Java plug-in that provides information
about Cisco CallManager from a remote workstation. You can save the RTMT configurations in
configuration profiles to allow quick reference of preset counters. The RTMT window gives you
the ability to monitor various aspects of Cisco CallManager performance and generate alerts,
which the RTMT can send to you via e-mail whenever the CallManager reaches a predefined alert
threshold.
Review Questions
1. Which two of the following statements about Microsoft Event Viewer are true? (Choose two.)
a. Microsoft Event Viewer needs to be installed separately and is included in the Microsoft
Windows 2000 support pack.
b. Microsoft Event Viewer assists administrators in troubleshooting Cisco CallManager
systems.
c. Microsoft Event Viewer is limited to the most recent 100 events per log type.
d. Microsoft Event Viewer stores system errors and warnings.
e. There are four log types in Microsoft Event Viewer.
2. How can Microsoft Performance Monitor be used to maintain Cisco CallManager systems?
(Choose three.)
a. To monitor Microsoft Windows 2000 counters
b. To monitor Cisco CallManager counters
c. To monitor events in real time
d. To analyze call flows
e. To analyze CDRs
f. To analyze trace files
g. To analyze Cisco CallManager configuration
3. What can RTMT be used for? (Choose two.)
a. Cisco CallManager performance monitoring
b. Cisco CallManager configuration
c. Cisco CallManager performance manipulation
d. Alert e-mail generation
e. Service activation
4. Which two of the following statements about the saving of configurations on RTMT are true?
a. If administrators exit RTMT, the active configuration is stored automatically.
b. Multiple configuration profiles can be stored.
c. RTMT needs to be restarted to load a saved configuration.
d. Configuration profiles can be identified by their number and date.
e. Configuration profiles can be identified by their name and description.
5. Which of the following statements about the RTMT window is true? (Choose three.)
a. The RTMT window supports tabs to allow many different elements to be viewed at
one time.
b. When switching between RTMT window tabs, real-time information graphs on inactive
(hidden) tabs are stopped.
c. The RTMT window provides performance monitoring similar to the Microsoft Perfor-
mance Monitor.
d. The RTMT window includes a link to start the Microsoft Performance Monitor MMC.
e. E)The RTMT window includes an alert central.
f. F)RTMT needs to be downloaded from Cisco.com.
6. Which area of the Microsoft Event Viewer would contain messages specific to Cisco
CallManager?
a. Error logs
b. Application logs
c. System logs
d. Warning logs
e. Security logs
7. What are the three views supported by Microsoft Performance Monitor? (Choose three.)
a. Histogram
b. Dynamic
c. Active
d. Alert
e. Graph
f. Report
g. Tabbed
8. Which of the following utilities will allow you to see processor and memory utilization levels
for a Cisco CallManager server? (Choose three.)
a. Microsoft Performance Monitor
b. CallManager Serviceability
c. Cisco RTMT
d. Windows Task Manager
e. Event Viewer
f. CallManager Control Center
following topics:
■ Identifying the functions of the Cisco

CallManager Serviceability Alarm interface
and its event levels and destinations
■ Configuring alarms for Cisco CallManager
services
■ Identifying and configuring the trace
functions of the Cisco CallManager
Serviceability Trace tool and services
■ Conducting a trace analysis and displaying
trace records in XML format
■ Collecting and compressing Cisco
CallManager service trace files
■ Explaining the features and functions of the
Bulk Trace Analysis tool
■ Identifying and explaining additional trace
tools and describing when to use each of
them
CHAPTER 32
Configuring Alarms and Traces
Administrators who install or maintain Cisco CallManager sometimes have to deal with
technical or design issues on their systems. To troubleshoot and monitor those issues, Cisco
CallManager provides alarms and traces similar to the functionality of the show and debug
commands of Cisco IOS software. This chapter describes how to configure traces on the Cisco
CallManager system and discusses tools that allow you to analyze those traces.
Alarm Overview
The Cisco CallManager Serviceability Alarm menu provides a web-based interface that has
two main functions:
■ To allow configuration of alarms and events
— Administrators can define what kind of information should be logged.

— Administrators can define where to store alarms and events.
■ To provide alarm message definitions
— Administrators can evaluate what kind of information (such as parameter and kind
of events) is included in which alarm.
Both functions assist you in troubleshooting and monitoring your Cisco CallManager system.
You can configure alarms for services (for example, Cisco CallManager, Cisco TFTP, or Cisco
CTIManager) for all Cisco CallManager servers in the cluster or for each server individually.
Alarms are used to provide the run-time status and state of the system and to take corrective
action for problem resolution; for example, to determine whether phones are registered and
working. Alarms contain information such as an explanation and recommended action. Alarm
information includes the application name, machine name, and cluster name to help you
troubleshoot problems that are not on the local Cisco CallManager.
You can configure the Alarm interface to send alarm information to multiple destinations, and
each destination can have its own alarm event level (from debug to emergency). CallManager
can direct alarms to the Microsoft Windows 2000 Event Log, a syslog server, system diagnostic
interface (SDI) trace log files, or signal distribution layer (SDL) trace log files.
714 Chapter 32: Configuring Alarms and Traces
When a service issues an alarm, the Alarm interface sends the alarm to the chosen monitors. Each
monitor forwards the alarm or writes it to its final destination (such as a log file). You can use this
information for troubleshooting or to pass over to another person for assistance (for example, the
Cisco Technical Assistance Center [TAC]).
You can turn on several alarm levels on Cisco CallManager. These alarm levels are equivalent to
the widely used syslog severity levels. Table 32-1 shows all available levels and describes the kind
of information that generates the alarm. As you can also see from the table, each level can be
identified by its name (debug to emergency) or by its number (0 to 7).
Table 32-1 Alarm Event Levels
Level Name Description

7 Emergency System unusable
6 Alert Immediate action needed
5 Critical Critical condition detected
4 Error Error condition
3 Warning Warning condition detected
2 Notice Normal but significant condition
1 Informational Information messages only
When the alarm event level is set to a certain value, it means that alarms that match the configured
level and alarms that match more severe levels are generated. In other words, an alarm level of 0
(debug) means all alarms of 0 or higher, and an alarm level of 4 means all alarms of level 4 or
higher. So if you configure an alarm level of 5, all critical, alert, and emergency alarms are logged.
Alarm Configuration
You can use the Alarm Configuration window in Cisco CallManager Serviceability to define
where CallManager will store the alarms and which level of alarms CallManager should store. You
can define the alarm level for each destination individually. Destinations for alarms are as follows:
■ Locally on the Cisco CallManager system in Microsoft Windows Event Log
■ Locally on the Cisco CallManager system in trace log files
■ Remotely on any syslog server; for example, Kiwi Syslog Daemon, a third-party application
that runs on Windows systems
Alarm Configuration 715
Configuring Alarms
To configure alarms in Cisco CallManager, follow this procedure:
Step 1 From Cisco CallManager Serviceability, choose Alarm > Configuration.

Step 2 Choose the appropriate server, which will then be displayed under Servers
on the left side of the window and at the top of the window. A box with
available services for alarms appears.
Step 3 From the Configured Services drop-down list, choose the service for
which you want to configure the alarm. The CallManager displays the
chosen service at the top of the window in the Current Service area, along
with the currently chosen server. A list of alarm monitors with event levels,
similar to the one shown in Figure 32-1 (which shows the alarm
configuration for the Cisco CallManager service), appears.
Figure 32-1 CallManager Alarm Configuration

Step 4 Check the check box for each desired alarm destination.
NOTE Only the Cisco CallManager and CTIManager services can use SDL Traces and only
these services will have the check box for Enable Alarm for SDL Trace available.
Step 5 From the Alarm Event Level drop-down menu, choose the desired alarm
event level for each of the available alarms.
Step 6 Save the configuration by clicking the Update button.
NOTE To apply the current settings for selected services to all nodes in a cluster, check the
Apply to All Nodes check box. To restore the default Cisco CallManager settings, click the
SetDefault button and then the Update button.
Configuring Alarms for Java Applications

You cannot configure alarms for Java-based applications, such as Java Telephony Application
Programming Interface (JTAPI) using the alarm configuration web pages. Use the Registry Editor
provided with the operating system to view the alarm configuration and to change Registry entries.
The Registry can be accessed by running the RegEdt32.exe or RegEdit.exe Registry Editors. Go
to HKEY_LOCAL_MACHINE\SOFTWARE\Cisco Systems, Inc.\Cisco Java
Applications\Monitors\Event Log in the Registry (as shown in Figure 32-2) and set these values:
■ Set the Enabled key to a value of 0 to turn off Event Log or to 1 to turn it on (by default, Event
Log is enabled for Java-based applications).
■ Set the Severity key to a value between 0 and 7 to set the alarm event level (the default severity
level for Java-based applications on Cisco CallManager systems is 6).
If you change the Registry entries, you must restart Java applications for the configuration changes
to take effect.
CAUTION It is recommended that you not change the Simple Network Management Protocol
(SNMP) Trap and Catalog configurations. Those settings influence SNMP traps generated by
the Cisco CallManager system. Those traps are used by network management applications, such
as CiscoWorks. Changing the settings could cause malfunctioning of the entire network
management system. Likewise, editing the Windows Registry is a dangerous process and can
cause the entire operating system to crash. Be sure you exercise caution when making changes
to the Windows Registry.
Alarm Configuration 717
Figure 32-2 CallManager Java Application Alarm Configuration
Alarm Details
Alarm definitions describe alarm messages. The definitions show what the alarms mean and how
to recover from them.
To reach the alarm message definition details for an alarm, complete these steps:
Step 1 From Cisco CallManager Serviceability, choose Alarm > Definition.

Step 2 From the Equals drop-down menu, choose the service for which to check
alarm definitions and click Find.
Step 3 From the window that opens (an example is shown in Figure 32-3), select
the alarm to see details. Figure 32-4 illustrates the alarm details for the
CallManagerFailure alarm.
Figure 32-3 Listing of Available Alarm Definitions
Cisco CallManager stores alarm definitions and recommended actions in a Structured Query
Language (SQL) server database. You can search the database for definitions of all the alarms. The
definition includes the alarm name, description, explanation, recommended action, severity,
parameters, and monitors. This information aids you in troubleshooting problems that Cisco
CallManager encounters.
NOTE Administrators can add their own text to the alarm definition by simply entering
information in the User Defined Text pane (not shown in Figure 32-4).
Trace Configuration 719
Figure 32-4 Alarm Details for the CallManagerFailure Alarm
Trace Configuration
Cisco CallManager Serviceability provides a web-based trace tool to assist the system
administrator and support personnel in troubleshooting Cisco CallManager problems. Cisco
CallManager Serviceability Trace provides three main functions:
■ Configuration—This function allows you to configure a variety of options when enabling

traces. Options include the level of trace details and the trace file format (.xml or .txt).
■ Analysis and Q.931 Translator—These functions allow you to analyze trace files.
NOTE The web-based Trace Analysis tool allows only analysis of Extensible Markup
Language (XML) files; the Q.931 Translator analyzes both text and XML files. The XML format
adds excessive overhead to the file, and causes the file to contain less information overall than
saving the file in another format. For in-depth troubleshooting, Cisco recommends using formats
other than XML.
■ Troubleshooting Trace Settings—This function allows you to enable troubleshooting traces.

With this feature, you can easily set up traces on multiple servers but there are fewer options
available than in the Trace Configuration function.
Types of Traces
Traces for Cisco CallManager services can be based on debug levels, specific trace fields, and
Cisco CallManager devices, such as phones or gateways. Two types of traces are available, SDI
trace and SDL trace.
SDI traces are also known as Cisco CallManager trace log files. Every Cisco CallManager service
includes a default trace log file. The system traces SDI information from the services and logs run-
time events and traces to a log file. Programmers typically use SDI traces for development
purposes.
SDL traces contain call-processing information from Cisco CallManager and Cisco CTIManager
services. The system traces the SDL of the call and logs state transitions in a log file. This log
information helps administrators to troubleshoot problems on the Cisco CallManager system.
TIP In most cases, extensive SDL traces will be gathered only when Cisco TAC requests it.
SDL traces track communication between the CallManagers, whereas SDI traces track internal
CallManager processing.
CAUTION Enabling traces decreases system performance; therefore, enable higher-level

trace only for troubleshooting.
Trace Configuration and Analysis Overview

Figure 32-5 provides an overview of trace configuration and analysis options:
■ Troubleshooting trace settings—Allows you to enable troubleshooting traces by server and

by service for all servers from a single page.
■ Trace configuration—Allows detailed trace configuration per server. The configuration

options include the trace file format (.xml or .txt).
■ Analysis—Allows you to analyze stored trace files (XML only).
■ XML or text editors—Allows you to examine the content of the stored trace files.
Figure 32-5 Trace Configuration and Analysis Overview
Troubleshooting
Settings for
Entire Cluster
Settings per
Server
Text or XML
Trace File Trace File
Web Access
(XML only)
XML or Text
Editor
Trace Configuration
Cisco CallManager provides many services for tracing. You can enable tracing for each service
individually for each server within the Cisco CallManager cluster. Perform these tasks in the Cisco
CallManager Serviceability Trace Configuration window to configure custom settings for a trace:
Step 1 In Cisco CallManager Serviceability, choose Trace > Configuration and

choose the server where you want to configure the trace settings.
Step 2 From the Configured Services list, choose the service to change the trace
settings, as shown in Figure 32-6. The Trace Configuration window opens.
Figure 32-7 displays a sample output of that window.
Figure 32-6 Selecting a Service to Trace
Step 3 To enable traces for the specified service, check the Trace On check box.
Step 4 Choose the desired debug level from the Debug Trace Level drop-down
menu.
Step 5 Choose the trace fields to include in the trace files. This additional
information is different for each service.
You can set the path and filename for the trace file. To allow creation of more than one file, Cisco
CallManager uses the entered filename and adds an eight-digit string, starting with 00000000, that
is incremented with each new file.
Figure 32-7 Trace Configuration
Enabling Clusterwide Trace Settings

The Troubleshooting Trace Settings menu item is misleading. This menu item allows you to
enable or disable troubleshooting traces on different servers throughout the cluster, as shown in
Figure 32-8. This window is usually used for troubleshooting when more than one server needs to
be monitored. Each service can be selected or deselected individually for each server.
Figure 32-8 Enabling Traces Clusterwide
NOTE Servers 172.30.2.67 and 172.30.2.68 shown in Figure 32-8 are currently offline, which
is why the Troubleshooting Trace Setting window shows their services as N/A.
Table 32-2 describes the logging of each service that can be turned on or off in the Troubleshooting
Trace Setting window.
Table 32-2 Trace Settings for Multiple Servers
Service Logging
Cisco CallManager Logs Cisco CallManager signaling information to trace files
Cisco Call Detail Records Logs information about writing of CDRs
(CDRs) Insert
Cisco Certificate Authority Logs information about issues of Cisco CAPF
Proxy Function (CAPF)
Cisco CTIManager Logs information about issues on CTIManager of the Cisco
CallManager
Trace Analysis 725
Table 32-2 Trace Settings for Multiple Servers (Continued)
Service Logging
Cisco CallManager Logs Cisco CallManager signaling information to trace files
Cisco Certificate Trust List Logs information about issues of Cisco CTL Provider
(CTL) Provider
Cisco Database Layer Logs information about database usage of Cisco CallManager
Monitor
Cisco Extended Functions Logs information about issues referring to any of the Cisco
CallManager extended functions, such as Quality Report Tool
(QRT)
Cisco CallManager Logs information about issues of the Cisco CallManager Extension
Extension Mobility Mobility service and Extension Mobility application
Cisco IP Manager Assistant Logs information about issues and usage of Cisco IPMA
(IPMA) functionality on Cisco CallManager
Cisco IP Voice Media Logs information about issues of the Cisco IP Voice Media
Streaming Application Streaming Application in conjunction with every involved device
Cisco Messaging Interface Logs information about issues of Cisco Messaging Interface
Cisco Music on Hold Logs information about issues with MoH on Cisco CallManager
(MoH) Audio Translator
Cisco Real-Time Logs information about issues involving collecting real-time
Information Server (RIS) information on Cisco CallManager
Data Collector
Cisco Telephony Call Logs information about issues with attendant consoles and pilot
Dispatcher (TCD) points on Cisco CallManager
Cisco TFTP Logs information about issues with the TFTP server service of
Cisco CallManager
Cisco WebDialer Logs information about issues with the click-to-dial functionality
on Cisco CallManager systems
Trace Analysis
The Trace Analysis tool is a postprocessing tool that allows analyzing of XML trace files via a web
GUI. The tool is available from the Cisco CallManager Serviceability Trace menu and provides
trace details to help narrow your investigation of system problems.
When you are using the Trace Analysis tool, it is possible to specify what kind of information you
need when analyzing the Cisco CallManager trace files. When troubleshooting, this makes it
easier to go through the trace files and get the necessary information.
In the Trace Analysis window, choose what kind of information to analyze and from which trace
file by following these steps:
Step 1 From the Cisco CallManager Serviceability window, select the Trace >
Analysis menu item.
Step 2 Choose a Cisco CallManager server for which trace files should be
analyzed.
Step 3 Choose the desired service, as shown in Figure 32-9.
Figure 32-9 Choosing XML Trace Files Using the Trace Analysis Tool
Step 4 Define whether to analyze SDI or SDL traces and click the List Files button
to get the list of all available SDI or SDL files that meet the criteria.
Step 5 Choose the file to analyze.
Step 6 Open the Cisco CallManager Serviceability web GUI of the Trace Analysis
tool. In the web GUI, define what information to list for the analysis.
Step 7 To process the records that meet your criteria, click Display Records.
Trace Analysis 727
NOTE The Trace Analysis tool supports only trace files containing less than 2 MB of data.
From the selections that you made for trace analysis, Cisco CallManager creates a web page giving
all the desired data. Figure 32-10 shows an example output.
Figure 32-10 Analyzing XML Trace Files Using the Trace Analysis Tool
You can directly open trace files from the folders where they are stored (subdirectories under
C:\Program Files\Cisco\Trace). You can use text editors (for XML and .txt files) or XML viewers
or editors to view XML files. Trace files in .txt format cannot be analyzed by using the Trace
Analysis tool. They can be analyzed by the Q.931 Translator tool; however, this tool interprets
only Q.931 messages. To examine other messages from stored .txt trace files, you can use a text
editor, as shown in Figure 32-11. Cisco TAC also provides other in-depth trace analysis tools such
as Translator X, Triple Combo, and the Voice Log Translator (VLT).
Figure 32-11 Analyzing TXT Trace Files
Trace Collection
To analyze trace files of a Cisco CallManager system locally on an administrator PC, it is
necessary to collect trace files from the Cisco CallManager file system and copy them to the PC
of the administrator. Instead of manually copying all trace files from all servers of the cluster, you
can use the Trace Collection tool, which automates this process.
If you want to analyze trace files from multiple servers on your local PC, use the Trace Collection
tool to transfer the trace files to your local PC, where they can then be analyzed by the Bulk Trace
Analysis tool or a text editor. There are three steps involved, as illustrated in Figure 32-12:
Step 1 When traces are enabled, Cisco CallManager writes trace information to
the local trace files.
Step 2 The Trace Collection tool can be used to download all trace files (including
XML trace files) from all Cisco CallManager systems in the cluster.
Step 3 The Bulk Trace Analysis tool allows analysis of XML trace files at the
administrator PC. You can view text files using a text editor.
Trace Collection 729
Figure 32-12 Remote Analysis of Trace Files
1 2 3
Collect XML or TXT traces with
Trace files are located on Examine collected traces files
Trace Collection tool in a .zip
Cisco CallManager servers. with Bulk Trace Analysis tool.
archive on the administrator PC.
The Trace Collection Tool

The Trace Collection tool allows you to collect trace information for any Cisco CallManager
service of any Cisco CallManager throughout the cluster, including the time and date of the trace
for that service. You can run this tool on any PC on the network because it is available from the
Cisco CallManager Install Plugins window.
After the Trace Collection tool is installed on the PC, enter the IP address, username, and password
for Cisco CallManager. As soon as the connection to the server is established, the window shown
in Figure 32-13 opens.
Figure 32-13 Trace Collection Tool

Three tabs are available in the Cisco CallManager Trace Collection Tool window. Each has its own
function, as described in Table 32-3.
Table 32-3 Tab Functions in the Cisco CallManager Trace Collection Tool Window
Tab Function
Select CallManager Provides a grid of services for the Cisco CallManager nodes in the cluster.
Services Choose all or some of the services for which traces should be collected by
checking the appropriate check boxes.
Select CallManager Provides the list of Cisco CallManager applications for which traces can be
Applications collected. Choose all or some of the applications in this tab.
Select System Provides a list of all system logs from which data can be selected.
Traces
NOTE The Back and Next buttons shown in Figure 32-13 move you between the three tabs.
After selecting the desired system logs, define the date range for the data to be selected to help
minimize the amount of data that needs to be collected. You can then click the Collect Traces
button for the application to start downloading and compressing all files locally to the PC. All files
are added to the system primary hard drive (C:\) to an archive named
CiscoCallManagerTraceCollection.zip.
CAUTION Selecting all kinds of data from all Cisco CallManager servers on the system leads
to extremely long download times, extensive file-compression processing, and large compressed
files.
Bulk Trace Analysis

The Bulk Trace Analysis tool is a postprocessing tool that allows analyzing XML trace files on
every PC locally. You must first download the trace files that you want to analyze using the Trace
Collection tool. You can download the Bulk Trace Analysis tool from the Cisco CallManager
Install Plugins window. This tool works similarly to the Trace Analysis web GUI on Cisco
CallManager, but it has some extra features. These extra features allow the user of the Bulk Trace
Analysis tool to analyze files more easily than with the Trace Analysis web GUI. This tool is ideal
to analyze trace files received from external sources because you only need the trace files for the
analysis and not access to the Cisco CallManager system where they were created.
Bulk Trace Analysis Features

The Bulk Trace Analysis tool supports the following features that are not provided within the
Trace Analysis web GUI on Cisco CallManager:
Bulk Trace Analysis 731
■ It creates reports of information that can be analyzed for troubleshooting, using multiple trace
files as the input source data.
■ It allows multiple views of a single report to compare and analyze multiple issues
simultaneously and concentrate on essential fields.
■ It allows users to customize report formats, sort trace information by type, and filter
information by special trace tags as well as date and time.
■ It allows saving and printing of reports for further analysis.
■ It works without using Cisco CallManager processing power because it runs locally on a PC.
■ You can use it to analyze large trace files (larger than 2 MB).
Using Bulk Trace Analysis

Figure 32-14 shows the analysis of an XML trace file using the Bulk Trace Analysis tool.
Figure 32-14 Using Bulk Trace Analysis
To generate this report, choose File > New Report. In the window that opens, choose whether to
create an SDI or an SDL report and where the file for analysis is located. To add new views for the
report, choose View > New View. In the window that opens, choose which kind of information
should be included in the new view. To change the order of the columns or to add or remove
columns, choose View > Customize Header and make the selections from the new window.
Additional Trace Tools

You can analyze the voice network using many additional trace tools provided by Cisco and other
vendors. Table 32-4 summarizes the situations that might require you to use these tools.
Table 32-4 Additional Trace Tools
Tool Use Available From

Q.931 Translator Analyzes Q.931 messages from Cisco Cisco CallManager
CallManager logs Serviceability
Voice Log Analyzes voice flows from Cisco CallManager Cisco TAC or Cisco.com
Translator logs Voice Downloads page
Dick Tracy Analyzes voice flow on T1/E1 and FXS ports Cisco.com
on Cisco Catalyst 6500 systems
Ethereal Captures network traffic, which also allows Ethereal.com
you to analyze voice traffic on the network
Examples of those additional tools are as follows:
■ Q.931 Translator—Allows analysis of Q.931 messages from text or XML log files created
on Cisco CallManager systems. The tool shows the ISDN messages and their Cisco IOS
translation. The Q.931 Translator can be used to save Cisco IOS translations to another log
file for further investigation. To access the Q.931 Translator, from Cisco CallManager
Serviceability, choose Trace > Q931 Translator.
■ Voice Log Translator—Formerly known as X-Log, which is a standalone application that

can be used to analyze the complete voice flow from Cisco CallManager log files. The tool is
based on Q.931 Translator but adds some additional functionality, such as Skinny Client
Control Protocol (SCCP) support for easier troubleshooting of voice flow issues. The Voice
Log Translator tool is available only from Cisco TAC to troubleshoot voice flows on
nonfunctioning Cisco CallManager systems.
■ Dick Tracy—Allows you to capture and analyze voice flows on T1/E1 or Foreign Exchange
Station (FXS) ports of Cisco Catalyst 6500 Series switch systems running the CatOS. To
download the Dick Tracy tool from Cisco.com, you must hold a valid Cisco.com account.
This tool is available to users with valid CCO account access at http://www.cisco.com/cgi-
bin/tablebuild.pl/dctool.
■ Ethereal—One of the most powerful, open-source network capture and analysis tools. It
supports analysis of nearly all types of packets captured from the network. For voice analysis,
it includes special features, such as painting graphs of voice over IP (VoIP) signaling flows or
saving G.711 Real-Time Transport Protocol (RTP) streams as .au files for playback with
media players. Ethereal is available for free at http://www.ethereal.com.
Summary
This chapter discussed the use of the alarm and trace features of Cisco CallManager. Eight alarm
levels are available on Cisco CallManager, which map to the syslog levels used by Cisco routers
and switches. The alarm configuration allows you to define where CallManager should save
alarms and what level of information the server should include.
Administrators can define the type and format of information that CallManager writes to trace
files. To analyze trace files, you can use the Serviceability Web interface, the Trace Analysis tool,
or a simple text or XML editor. The Trace Collection tool allows you to download and compress
CallManager trace files to a local PC. You can then analyze the trace files using the Bulk Trace
Analysis tool. Additional trace tools provided by Cisco and other vendors allow administrators to
further analyze their voice network.
Review Questions
1. What does Cisco CallManager Serviceability Alarm menu provide? (Choose two.)
a. Configuration of alarms
b. Alarm analysis
c. Alarm message definitions
d. Configuration of traces
e. Alarm modification
2. When you are configuring alarms on Cisco CallManager Serviceability, which of the
following statements is true?
a. You can define an e-mail address to which you can send alerts.
b. It is possible to define which fields should be included in written SDI and SDL trace
files.
c. More than one destination can be used to write alarm logs in parallel, and each of them
can use its own alarm level.
d. Alarm levels are predefined on Cisco CallManager and cannot be changed.
3. What is the difference between configuring alarms for Java applications and configuring
alarms for other services on Cisco CallManager?
a. Nothing; the alarms are both configured from CallManager Serviceability.
b. Java applications are not able to be monitored using Cisco CallManager.
c. Alarms for Java applications use system resources more heavily.
d. You must enable Java application alarms through the Windows Registry.
4. Which three of the following statements about SDI and SDL traces are true?
a. SDI traces log services and run-time events.
b. SDL traces log services and run-time events.
c. SDI traces log call-processing information.
d. SDL traces log call-processing information.
e. SDI and SDL traces can be written to plaintext and XML files.
f. SDI and SDL traces can be written to plaintext files only.
g. SDI and SDL traces can be written to XML files only.
5. What does the Trace Collection tool do?
a. Collects traces from Cisco CallManager systems and writes them to remote hosts
b. Allows you to configure the kind of information that should be collected and written to
trace files
c. Downloads and compresses trace files from Cisco CallManager systems to a computer
d. Saves disk space on Cisco CallManager by deleting trace files after downloading them
6. What can you do with the web-based Trace Analysis tool in Cisco CallManager
Serviceability? (Choose two.)
a. Analyze plaintext-formatted SDI trace files.
b. Analyze plaintext-formatted SDL trace files.
c. Analyze XML-formatted SDI trace files.
d. Analyze XML-formatted SDL trace files.
e. Analyze files larger than 2 MB.
7. What is the difference between the Q.931 Translator and the Voice Log Translator?
a. Q.931 Translator supports SCCP.
b. Q.931 Translator supports MGCP.
c. The Voice Log Translator can be used to analyze log files without access to the Cisco
CallManager system.
d. The Q.931 translator is provided for free, but to use Voice Log Translator, you must pay
a special license fee.
8. Which services support SDL trace logging? (Choose two.)
a. Cisco TFTP
b. Cisco CallManager
c. Database Layer Monitor
d. CTIManager
e. SDI Monitor
9. You receive an alarm level 5 message coming in from the Cisco CallManager service. What
well-known name is this alarm level assigned?
a. Emergency
b. Error
c. Critical
d. Alert
10. Which of the following represent a valid CallManager alarm destination? (Choose three.)
a. Microsoft Event Log
b. Syslog Server
c. SMTP Server
d. Trace Log File
following topics:
■ Describing the uses, features, and operation

of the CAR tool
■ Explaining the contents of CDR and CMR
■ Explaining the three levels of CAR users and
the reporting capabilities of each
■ Explaining the types of CAR reports and
which user levels can generate them
■ Configuring CAR system parameters
■ Configuring the system schedule to schedule
daily, weekly, and monthly reports
■ Configuring CAR alerts when the database
size exceeds a configured threshold
■ Generating, viewing, or mailing user reports
using CAR
CHAPTER 33
Configuring CAR
The Call Detail Record (CDR) Analysis and Reporting (CAR) tool is a powerful application that
gives an overview of the call volume of users or departments. CAR can also generate reports for
special route patterns and features such as Client Matter Codes (CMC) and Forced
Authorization Codes (FAC). This chapter discusses the concepts and configuration behind CAR
and the applications for a corporate IP telephony deployment.
CAR Overview
The Cisco CallManager Serviceability CAR tool, formerly known as Administrative Reporting
Tool (ART), generates reports of information for quality of service (QoS), traffic, user call
volume, billing, and gateways. The CAR tool generates reports in either Portable Document
Format (PDF) or Comma-Separated Values (CSV) format. The PDF format limits the number
of records in the CAR reports to 5000, and CSV format limits the number of records to 20,000.
If the number of records exceeds these limits, a message warns that the results are truncated. To
avoid truncating reports, reduce the date range and regenerate the reports.
NOTE CAR is not installed during the standard Cisco CallManager installation and must
be added separately.
If CAR is running on your system before you upgrade to a new version of Cisco CallManager,
the upgrade process automatically upgrades CAR. If Cisco CallManager is being installed for
the first time, CAR has to be installed manually from the Cisco CallManager Administration.
You can install CAR from the standard plug-ins web page (Application > Install Plugins). You
must install CAR on the Cisco CallManager publisher that hosts the CDR database. CAR uses
the Cisco Tomcat service.
CDRs and CMRs

Call detail records (CDRs) and Call Management Records (CMRs) are both stored in the CDR
database, accessible with the Microsoft SQL Server Enterprise Manager. Choose Start >
Programs > Microsoft SQL Server > Enterprise Manager to open the Microsoft SQL Server
Enterprise Manager on the Cisco CallManager publisher. Both types of records store
information about a call. The CDR table stores details about the call itself, whereas the CMR
738 Chapter 33: Configuring CAR
table stores information about QoS parameters for the same call. The SQL database refers to the
CDR and CMR tables as CallDetailRecord (for CDRs) and CallDetailRecordDiagnostic (for
CMRs), as shown in Figure 33-1. For every CDR entry, CallManager also generates a CMR entry.
Figure 33-1 CDR and CMR Tables in the SQL 2000 Database
NOTE When you purge a record in the CDR database, CallManager deletes the related entries
for a call in both the CallDetailRecord and CallDetailRecordDiagnostic tables.
The CallDetailRecord table stores extensive information about a call. Each record has a unique
identifier called the pkid that enables you to find and identify a single record. The
CallDetailRecord table also contains information about the calling party, called party, duration of
or information about call forwarding, CMC, and FAC, to name the most important fields of a
record. In total, the CallDetailRecord table contains 68 information fields.
The number of columns for the CallDetailRecordDiagnostic table, in contrast to the

CallDetailRecord table, is very small. It has only 18 information fields. A unique identifier (pkid)
for every call is included in the CallDetailRecordDiagnostic table as well. The pkid for a call in
the CallDetailRecordDiagnostic table is not the same as the pkid for the same call in the
CallDetailRecord table. To reference a call in the tables, use the oriLegCallIdentifier and
CAR Overview 739
destLegIdentifier field information. Among other things, information about the packets sent and
received, jitter, and latency are stored for QoS reports.
CAR Users
CAR has a distinct management interface from the Cisco CallManager Administration and
Serviceability web pages, as shown in Figure 33-2. The available menus in CAR depend on the
level of the user authenticating to the system. CAR provides reporting capabilities for three levels
of users:
■ Administrators can generate system reports for load balancing, system performance, and
troubleshooting. The administrator can also grant administrator access to other users,
configure the dial plan and gateway, and set the system preferences.
■ Managers can generate reports for users, departments, and QoS. The reports provide
information for budgeting or security purposes and for determining the voice quality of the
calls in their department.
■ Individual users can generate a billing report for their own calls.
Figure 33-2 CDR Analysis and Reporting Administrative Interface

CAR Report Types and User Levels

There are three report types available in CAR:
■ User Reports—Allows you to generate reports for bills, Top N users, Cisco IP Manager
Assistant (IPMA), computer telephony integration (CTI), and phone services. Bills can be
generated for individual users, groups of users, or all users. With the Top N report, it is easy
to find, for example, the top five users with the highest cost, highest call duration, or highest
number of calls.
■ System Reports—Allows you to generate reports for QoS, traffic, and malicious calls. With
these reports, it is possible to find part of the network where QoS does not work properly or
where there is more traffic than planned in the design phase. Reports for Multilevel
Precedence and Preemption (MLPP), CMC, and FAC are also available for information about
the cost of projects or controlled long-distance calls with authorization codes.
■ Device Reports—Generates information about the gateway, route plans, conferences, and
voice messaging. These kinds of reports are useful for the administrator in optimizing the
network.
The administrator can generate all types of reports: user reports, system reports, and device
reports. Managers can generate only parts of the user and system reports, such as billing, Top N,
and QoS reports, and only for employees who report directly or indirectly to the manager. Users
can check only their own bills in general or in detail, and send the report by e-mail. Table 33-1
summarizes these rights.
Table 33-1 Report and User Type Matrix
Report Administrator Manager User

User Reports Full Access Bills, Top N Only their own bills
System Reports Full Access QoS No Access
Device Reports Full Access No Access No Access
CAR Configuration
The configuration of CAR is done through a web-based interface that uses the same look and feel
as the Cisco CallManager Administration and Serviceability windows. After you have installed the
CDR application on the publisher server, a new CDR Analysis and Reporting menu item appears
under the Tools menu of the Cisco CallManager in a similar fashion to the Bulk Administration
Tool. The following sections walk you through the configuration and use of the CAR utility.
CAR Configuration 741
Initial Configuration
You must initially access the CAR web interface through the CallManager Serviceability window
by using the Tools > CDR Analysis and Reporting option. This menu option will only exist after
you have installed the CDR application on the publisher server through the Install Plug-ins
window. When initially accessing the CAR web interface, the authentication settings will be in a
strange state. Even if you have enabled Multilevel Administration (MLA) for Cisco CallManager,
you must use a default login to CAR of a username of admin and a password of admin.
NOTE After you have initially configured the CAR utility, users (and managers) will be able
to access the utility at https://<Publisher IP Address>/ART.
After you have authenticated, the only available option is the Admin Rights menu. Selecting this
option takes you to the window illustrated in Figure 33-3.
Figure 33-3 Initial CDR Admin Rights Window
From here, you must add the initial CDR user from the CallManager database (or Windows
database if you have not enabled MLA). CDR Analysis and Reporting immediately places this
initial user into the CDR Administrators group, giving this user full administrative privileges to
the web interface.
NOTE Following the addition of the first user, CallManager disables the default username and
password of admin.
CAR System Parameter Menu Configuration

After you have added the initial administrator to the CAR system, you must log in as the new user.
After a successful authentication, you will be able to see all CAR menu options. As part of the
initial setup, most administrators will configure the System > Service Parameters menu option,
Figure 33-4 Configuring CDR Service Parameters
You can configure additional CAR administrators by using the CAR tool itself. Choose System >
System Parameters > Admin Rights to assign administrative rights to any user in the Cisco
CallManager user database. The manager and user levels are not configured in the CAR tool.
These roles are based on the user configuration in the global directory. In Cisco CallManager
Administration, choose User > Global Directory to edit users in the global directory. For each
user, you can configure the user ID of the manager of the user in the Manager User ID field. CAR
assigns users who you have configured as managers to the manager level in CAR. CAR assigns
all other users to the user level in CAR.
You can configure the additional Service Parameters menu items as follows:
■ The Mail Parameters option allows you to configure an SMTP server allowing the CDR
Analysis and Reporting tool to send alarms and e-mails to administrators directly.
■ Cisco has configured CAR to categorize calls using the North American Numbering Plan
(NANP). If your organization uses additional dial plans (such as interoffice dialing), you can
add them to CAR using the Dial Plan Configuration menu item. This gives CAR the ability
to classify call types correctly on reports. CAR classifies external calls based on the outside
number.
■ In the Gateway Configuration menu item, you can provide the maximum number of ports
information for each gateway to enable CAR to generate accurate utilization reports.
■ Use the System Preferences menu item to specify generic parameters, such as the company
name, that CAR will insert in every report.
These items are discussed further in the following sections.
Administration Rights and Mail Parameters

Choose System Parameters > Admin Rights to grant or revoke administrative rights to users
from the Lightweight Directory Access Protocol (LDAP) directory. You can search for users in the
LDAP directory or directly enter the user ID. As stated earlier, access rights for managers and
users are not granted in the CAR tool but are derived from the roles configured in Cisco
CAR can send reports by e-mail. To enable this feature, choose System Parameters > Mail
Parameters and configure the e-mail account information. Enter the mail ID and the password
that CAR will use, as shown in Figure 33-5. This e-mail account must exist on the mail server. You
also need to specify the mail domain and the mail server name. The mail domain is added
automatically when a user enters only the name of the recipient without the mail domain. Every
time a report is sent with the CAR tool, this account will be used.
Figure 33-5 Configuring CDR Mail Parameters
Dial Plan Configuration

The default dial plan in CAR specifies the NANP, as shown in Figure 33-6. To configure the dial
plan, define the parameters for outgoing call classifications. Call classifications include
International, Local, Long Distance, On Net, and Others. For example, if local calls in the area are
six digits in length, specify a row in the dial plan as follows:
■ Condition— =
■ No of Digits— 6
■ Pattern— !
■ Call Type— Local

Figure 33-6 Configuring CDR Dial Plan Parameters
Table 33-2 describes the parameters in the Dial Plan Configuration window and lists their possible
values.
Table 33-2 Explanation of CAR Dial Plan Parameters
Parameter Possible Values Description

Condition > Condition of the rule—greater than (>), less than (<), or
equal to (=) the specified value in the No of Digits field.
<
=
No of Digits NA Number of digits in the directory number (DN) to which
this rule should be applied. If the number of digits does
A digit not affect the rule, specify NA.
continues
Table 33-2 Explanation of CAR Dial Plan Parameters (Continued)
Parameter Possible Values Description

Pattern G Used for the call classification:
T G means that the call is classified as specified in the rule

(that is, G is a wildcard for the gateway area codes
! specified in the “Gateway Configuration” section).
X T retrieves toll-free numbers.
! signifies multiple digits.
X signifies a single-digit number.

Call Type International Choose this call type if the condition is satisfied.
Local
Long Distance
On Net
Others
Toll-free Digits Numbers in the dial plan that can be placed without a
Numbers charge.
Gateway and System Preferences

Configure the gateways in CAR before using the CAR gateway reports, and update the gateway
configuration every time you add a new gateway to the Cisco CallManager Administration. When
you delete gateways in Cisco CallManager Administration, CAR automatically deletes them from
its tracking. CAR uses the area code information, shown in Figure 33-7, to determine whether calls
are local or long distance. Provide the maximum number of ports information for each gateway to
enable CAR to generate the utilization reports. Remember, G is a wildcard for the gateway area
codes used in dial plan configuration.
Figure 33-7 Configuring CDR Gateway Parameters
CAR uses the values you configured for the gateway when you added the gateway in Cisco
CallManager Administration. Therefore, some gateways might already have an area code setting
or have a maximum number of ports.
CAR provides default system preferences, shown in Figure 33-8.
Figure 33-8 Configuring CDR System Preferences Parameters

The following list provides a brief description of each of these fields:
■ COMPANY_NAME—The company name is used as header information in reports.
■ ERRORLOGFILESIZE—The maximum size of the error log file in kilobytes, with a range
from 1 to 9999. The default value is 100.
■ SESSIONTIMEOUT—The time in seconds that must pass without any activity before a user
is logged out of CAR, with a range from 60 to 86,400 (1 minute to 24 hours). The default
value is 1800 (30 minutes).
Report Scheduling
Every Cisco CallManager writes the CDR data first locally into flat files. A flat file is a temporary
file without a file extension stored in the C:\Program Files\Cisco\CallDetail\ folder on Cisco
CallManager. Flat files store call information to import the data on the publisher server into the
CDR database. At a specified time, all the CallManager servers transfer the flat files to the Cisco
CallManager publisher. Loading CDR data from flat files into the CDR database on the Cisco
CallManager publisher can cause performance degradation on the Cisco CallManager server.
Cisco recommends that you use the default loading time or schedule the loading to occur at a time
when Cisco CallManager performance will be least affected. By default, CDR data loads every
day from midnight to 5:00 a.m.
You can manage CDR loading through the System > Scheduler menu, as shown in Figure 33-9.
These menu options are discussed fully in the upcoming sections. Disable CDR loading when you
are installing or upgrading the system in the same off-hours during which CDR loading normally
occurs. Because loading CDRs drains Cisco CallManager resources, you can suspend CDR loads
until other operations complete. Of course, CallManager does not update the CDR data when CDR
loading is disabled. Be sure to enable CDR loading again as soon as possible. The CAR tool does
not affect CDR generation in Cisco CallManager.
In addition, the CDR Scheduler menu allows you to generate reports for these periods:
■ Days
■ Weeks
■ Months
Figure 33-9 Configuring CDR Scheduler Parameters
Managing the CDR Load

You can use the CDR Load menu to enable and disable the CDR load. Disable loading if you are
upgrading the Cisco CallManager in the time specified for CDRs to load. Otherwise, the CDR load
could consume critical system resources.
Configure the Load CDR & CMR area parameters as follows, shown in Figure 33-10:
■ The Time field specifies time when the CAR tool should start to load CDR data from the local
CDR files into the CAR database.
■ The Loading Interval specifies how often CAR will load the CDR entries. The minimum
interval is 15 minutes and the maximum is 24 hours.
■ The Uninhibited Loading of CDR section allows you to define the duration of the loading
interval defined to limit the loading time to a maximum length to minimize the impact on
Cisco CallManager resources during business hours.
Figure 33-10 Configuring CDR Load Parameters
NOTE Depending on your configured loading interval, CAR might not update the CDR
records for a full 24-hour period (which is the configured default). This minimizes the near-real-
time monitoring of placed calls in the IP telephony network. You will need to balance the loading
interval time cycle based on your available resources.
Configuring the Schedulers

You can configure schedulers to generate reports on a daily, weekly, and monthly basis. The
schedulers are preconfigured as follows:
■ CAR generates daily reports (shown in Figure 33-11) every day at 1:00 a.m.
Figure 33-11 Configuring CDR Daily Reports
■ CAR generates weekly reports (shown in Figure 33-12) every Sunday at 4:00 a.m.
Figure 33-12 Configuring CDR Weekly Reports
■ CAR generates monthly bills (shown in Figure 33-13) on the first day in the month at
3:00 a.m., and other reports at 2:00 a.m.
Figure 33-13 Configuring CDR Monthly Reports
The Life field in the schedule configuration specifies the number of days, weeks, or months
(depending on the type of the schedule) that the reports are stored on the system. Choose the
desired value; for example, for weeks, from 0 to 12. After the configured lifetime has elapsed,
CallManager deletes the reports.
System Database Configuration

In the System > Database menu, you can configure CAR to notify an administrator when the CAR
database size or CDR database size exceeds a percentage of the maximum number of records, as
shown in Figure 33-14. Set the message and the maximum number of records, and specify the alert
percentage. CAR restricts you from editing the maximum number of records in the CDR database.
Sometimes, it might be necessary to purge the database to reallocate the database space. You can
use the System > Database > Manual Purge or Configure Automatically to disable database
purging manually, or automatically purge the database.
System Database Configuration 753
Figure 33-14 Configuring CAR Database Alerts
CDR and CAR Database Alerts

With the database alerts, CAR informs the administrator about predefined alert thresholds:
■ For a CAR database alert (shown in Figure 33-14), the maximum number of rows in the
billing table is 2,000,000 rows. A notification is sent when 80 percent of the rows are used.
By default, the CAR administrator receives an e-mail, and the users specified in the CC field
are also notified. The predefined e-mail subject is “Alert for CAR database.” Also, the e-mail
message is predefined as “Number of rows in Billing table in the CAR database has crossed
the threshold limit.”
■ For a CDR database alert (shown in Figure 33-15), the alert mechanism works in the same
way, and the threshold is predefined at 80 percent also. The difference is that the maximum
number of rows is 1,500,000.
Figure 33-15 Configuring CDR Database Alerts
Database Purges
Storage in the database is limited, so you must purge the database from time to time. To do this
operation manually in the Manual Database Purge configuration area, follow these steps:
Step 1 In CAR, choose System > Database > Manual Purge. The screen that
appears is illustrated in Figure 33-16. Choose the database in the Select
Database drop-down list. The two possible options are CAR and CDR.
Step 2 Choose the table to purge using the Select Table drop-down list. Possible
values are the CallDetailRecord, Tbl_Billing_Data, Tbl_Error_Log,
and Tbl_Purge_History tables. The Table Information button displays an
overview of the number of records in each of the tables and shows when the
first and last records were stored in the database.
Step 3 To delete records, use the Delete Records option. Database entries older
than the date specified in the Older Than field or from the range of dates
specified in the Between fields will be deleted.
Step 4 Click the Purge button.
System Database Configuration 755
Figure 33-16 Manually Purging the CDR Database
In addition to the manual purge option, CallManager can purge the databases automatically by
configuring the parameters in the Configure Automatic Database Purge area. All data older than
the specified number of days will be deleted from the CDR or CAR database. In the example
illustrated in Figure 33-17, CallManager will automatically purge records in the database older
than 100 days.
Figure 33-17 Automatically Purging the CDR Database

In most cases, third-party products are used for billing. Make sure you do not purge the database
before the billing system has replicated the data. Some billing products are also able to delete the
data after a successful replication. Be aware that there is no way to restore the deleted database
entries after the purge. Automatic purging of both the CDR and CAR databases is disabled by
default.
User Report Configuration

Users can access the CAR tool by browsing to https://<Publisher IP Address>/ART and logging
in with their username and password. A single user has limited access to the CDR database and
can generate only a report for his or her own calls. This limitation helps ensure the privacy of the
call data for users.
A user can generate an individual bill in a summary or detailed form. To generate a report, the user
can use the Bills > Individual menu to access the screen shown in Figure 33-18. From here, they
can choose either the CSV or PDF report format, set the time range, and click the View Report
button. For a PDF report, Adobe Acrobat Reader must be installed on the PC from which the user
is browsing to CAR. To send the report by e-mail, the user can click the Send Report button.
Figure 33-18 Generating User Reports

User Report Configuration 757
Viewing the Report

As shown in Figure 33-19, the user clicked the View Report button after specifying PDF format,
and the individual bill was generated. As the figure shows, Sue Monk has no reported calls from
the date range of April 1, 2006 through April 11, 2006.
Figure 33-19 Viewing User Reports
Sending the Report

Alternatively, the user could click the Send Report button to display a window for sending an e-
mail after the user specifies the recipients. As shown in Figure 33-20, the user could also send
copies by specifying the relevant e-mail addresses in the CC field. The subject of the mail is
predefined. Users can add information in the text field if necessary and click Send to e-mail this
report to the specified recipients.
Figure 33-20 E-Mailing User Reports
Summary
This chapter discussed the use of the CDR Analysis and Reporting (CAR) utility. CAR was
formerly known as the Administrative Reporting Tool (ART) and comes as an optional utility. To
use it, you must install it on the publisher server from the plug-ins window. CallManager stores
the CDRs in the CallDetailRecord SQL table and the CMRs in the CallDetailRecordDiagnostic
SQL table.
Within the CAR utility, there are three levels of users: users, managers, and administrators.
Likewise, there are three different reports that are available: user, system, and device reports. You
should disable CDRs when you are upgrading Cisco CallManager because the resource utilization
can be high during CDR replication. In addition, the CDR database can grow quickly in a thriving
IP telephony network. You should purge the database regularly either manually or automatically.
Review Questions
1. Which three statements about the CAR tool are true?

a. PDF reports are limited to 5000 records.
b. PDF reports are limited to 10,000 records.
c. CSV reports are limited to 20,000 records.
d. CSV reports are limited to 25,000 records.
e. CAR has to be installed on the publisher only.
f. CAR does not have to be installed on the publisher.
2. Which statement about CMR and CDR is true?
a. Both are stored permanently in flat files.
b. CMR stores call details.
c. CDR stores QoS parameters.
d. They are related to each other.
3. Which username and password combination is used for the first login to CAR?
a. cisco and cisco
b. cisco and dipsy
c. admin and admin
d. admin and cisco
4. Which is a valid CAR report type?
a. QoS reports
b. CDR reports
c. Database reports
d. Device reports
5. What is the first thing to do after the first login to CAR?
a. Grant access rights.
b. Configure the mail server.
c. Configure the dial plan.
d. Adjust the system preferences.
6. When is the CDR data loaded by default?

a. From midnight to 1:00 a.m.
b. From midnight to 3:00 a.m.
c. From midnight to 5:00 a.m.
d. In real time
7. The CAR and CDR database alerts the CAR administrator by default when _____ percent of
the maximum number of rows is reached.
a. 70
b. 75
c. 80
d. 85
e. 90
8. Which call type is valid in individual bills?
a. On Net
b. Intersite
c. MLPP
d. Video
9. What two file formats are valid for report generation in CAR?
a. TXT
b. XLS
c. PDF
d. CSV
10. If the publisher server is unavailable in a CallManager cluster, what happens to CDRs?
a. They are lost.
b. CDRs are stored on the root of the subscriber hard drive.
c. CDRs are stored in the memory of the subscriber server.
d. They are stored on the subscriber in “flat files.”
following topics:
■ Explaining the use of SNMP, syslog, and

CiscoWorks ITEM in remotely managing and
maintaining a Cisco CallManager system
■ Explaining the use of dependency records,
enabling them, and accessing them
■ Using the Password Changer tool to change
passwords
■ Using the Cisco Dialed Number Analyzer to
analyze inbound and outbound calls in a
Cisco CallManager dial plan
■ Configuring the QRT feature to display IP
phone problem reports from users
CHAPTER 34
Using Additional Management
and Monitoring Tools
Management and monitoring tools are available to help system administrators monitor,
troubleshoot, and manage the health of an IP telephony network. This chapter explains Cisco
CallManager tools such as dependency records, Password Changer tool, and Cisco Dialed
Number Analyzer. The Quality Report Tool (QRT), a tool used by end users to report IP phone
issues to administrators, is also covered.
Remote Management Tools

Network management software, such as CiscoWorks IP Telephony Environment Monitor
(ITEM) or syslog servers, performs specific tasks to monitor and manage the health and
availability of devices in a network. These tools are typically used in large-scale data networks
(such as computer networks and telecommunication networks).
CiscoWorks ITEM Overview

CiscoWorks ITEM is a powerful suite of applications and tools that continuously evaluate and
report on the operational health of the Cisco IP telephony implementation. CiscoWorks ITEM
is used to manage Cisco IOS software-based IP telephony environments. CiscoWorks ITEM
provides the following:
■ Proactive health and fault monitoring of converged IP networks and IP telephony

implementations
■ Powerful tools to effectively manage the day-to-day customer care responsibilities of help-
desk personnel
■ The ability to capture performance and capacity management data
CiscoWorks ITEM consists of a product suite:
■ CiscoWorks IP Telephony Monitor (ITM) —Monitors Cisco voice elements in the

network to alert operations personnel to potential problems and to help minimize
downtime. It also includes CiscoWorks Common Services, a common foundation for data
storage, login, access privileges, and navigation and launch management for all
CiscoWorks applications.
764 Chapter 34: Using Additional Management and Monitoring Tools
■ CiscoWorks IP Phone Information Utility (IPIU) —Provides operational status and

implementation details about an individual IP phone. It also provides security reports that
document IP phone moves, adds, and changes, as well as information about the physical and
logical connections of every Cisco IP Phone installed in a given network.
■ CiscoWorks IP Phone Help Desk Utility (IPHDU) —Reports operational status and
implementation details about individual IP phones. It works in conjunction with IPIU to make
read-only access to Cisco IP Phone installation details available to help-desk personnel.
■ CiscoWorks ITEM Gateway Statistics Utility (GSU) —Collects performance and behavior
statistics about IP telephony gateways controlled by Cisco CallManager and Cisco IOS
software, which can be processed by third-party software to produce utilization and capacity
management reports.
■ CiscoWorks WAN Performance Utility (WPU) —Measures the performance, latency, and
availability of multiprotocol IP networks on an end-to-end and hop-by-hop (router-to-router)
basis.
CiscoWorks ITEM also provides real-time fault detection and determination about the underlying
Cisco IP fabric on which the IP telephony implementation executes. CiscoWorks ITEM reports
faults that occur on Cisco network devices, often identifying problems before users of network
services realize that a problem exists. CiscoWorks ITEM supports more than 200 types of the most
popular Cisco routers, switches, access servers, and hubs. For each of these supported devices,
CiscoWorks ITEM automatically looks for a broad spectrum of common problems at the device
and VLAN level, all without ever requiring operations managers to write rules or set polling or
threshold values. CiscoWorks ITEM can listen to traps or can send polls and pings to get
information for devices. CiscoWorks ITEM can send alerts and events automatically to an e-mail
client or a pager. To monitor the network, CiscoWorks ITEM can be accessed via a web-based
interface or a CiscoWorks desktop client.
Network management systems (NMSs) use Simple Network Management Protocol (SNMP), an
industry-standard interface, to exchange management information between network devices.
Simple Network Management Protocol

SNMP is an application-layer protocol, part of the TCP/IP protocol suite. SNMP enables
administrators to remotely manage network performance, find and solve network problems, and
plan for network growth. There are three versions of SNMP:
■ SNMP Version 1 (SNMPv1)

SNMPv1 and SNMPv2 have a number of common features, but SNMPv2 offers enhancements,
such as additional protocol operations. Standardization of SNMPv3 is pending.
SNMP Basics
An SNMP-managed network consists of three key components:
■ Managed devices—A managed device designates a network node that contains an SNMP
agent and resides on a managed network. Managed devices collect and store management
information and make it available by using SNMP.
■ Agents—An agent, as network management software, resides on a managed device. An agent

contains local knowledge of management information and translates it into a form that is
compatible with SNMP.
■ NMSs—An NMS comprises an SNMP management application together with the computer
on which it runs. An NMS executes applications that monitor and control managed devices.
An NMS provides the bulk of the processing and memory resources that are required for
network management. These NMSs share compatibility with Cisco CallManager:
— CiscoWorks2000
— HP OpenView
— Third-party applications that support SNMP and Cisco CallManager SNMP
interfaces
This list specifies Cisco CallManager SNMP trap messages that are sent to an NMS that is
specified as a trap receiver:
■ Cisco CallManager failed

■ Phone failed
■ Phones status update
■ Gateway failed
■ Media resource list exhausted
■ Route list exhausted
■ Gateway Layer 2 change
■ Quality report
■ Malicious call
SNMP itself is a simple request-and-response protocol. NMSs can send multiple requests without
receiving a response. Table 34-1 defines six SNMP operations.
Table 34-1 SNMP Operations
Operation Definition
Get Allows the NMS to retrieve an object instance from the agent.
GetNext Allows the NMS to retrieve the next object instance from a table or list within an
agent. In SNMPv1, when an NMS wants to retrieve all elements of a table from an
agent, it initiates a Get operation, followed by a series of GetNext operations.
GetBulk Makes it easier to acquire large amounts of related information without initiating
repeated get-next operations. GetBulk (new in SNMPv2) was designed to virtually
eliminate the need for GetNext operations.
Set Allows the NMS to set values for object instances within an agent.
Trap Allows the agent to asynchronously inform the NMS of some event.
Inform Allows one NMS to send Trap information to another (new in SNMPv2).
SNMP Configuration on Cisco CallManager

Administrators have to enable SNMP for each device in the network that they want to monitor. For
example, if you want to monitor Cisco CallManager with SNMP, on the Cisco CallManager
server, click Start > Programs > Administrative Tools > Services. From the Service menu,
search for the SNMP service and open the service to configure it. The SNMP Service Properties
window shown in Figure 34-1 opens.
Figure 34-1 CallManager SNMP Service Configuration

Cisco CallManager supports SNMPv1 and SNMPv2. SNMPv1 lacks authentication capabilities
(SNMPv2 increases the security capabilities of SNMP, and SNMPv3 supports authentication and
encryption), which results in vulnerability to a variety of security threats:
■ Masquerading consists of an unauthorized entity attempting to perform management

operations by assuming the identity of an authorized management entity.
■ Modification of information involves an unauthorized entity attempting to alter a message

generated by an authorized entity so that the message results in unauthorized accounting
management or configuration management operations.
■ Message sequence and timing modifications occur when an unauthorized entity reorders,
delays, or copies and later replays a message generated by an authorized entity.
■ Disclosure results when an unauthorized entity extracts values stored in managed objects, or
learns of notifiable events by monitoring exchanges between managers and agents.
Because SNMPv1 does not implement authentication, many vendors do not implement Set
operations, thus reducing SNMP to a monitoring facility.
The first thing you need to do to configure SNMP management is to enable SNMP access. Enable
access by configuring community strings, which act somewhat like passwords. The difference is
that there can be several community strings and that each of them can grant a different form of
access.
A community string can be the following:
■ Read-only—Gives read access to all objects in the MIB except the community strings but
does not allow write access
■ Read-write—Gives read and write access to all objects in the MIB but does not allow access
to the community strings
■ Read-write-all—Gives read and write access to all objects in the MIB, including the
community strings
To define the SNMP read community string, complete these steps:
Step 1 Choose Start > Programs > Administrative Tools > Services.
Step 2 Choose the SNMP service, double-click the service name, and choose
Security.
Step 3 In the Security window, define community strings and assign them read
permission.
This would now be the read community string a remote system could use to view various attributes
of the Cisco CallManager server. In the example shown in Figure 34-2, the community string is
“item.”
Figure 34-2 CallManager SNMP Community String Configuration
Cisco Systems supports numerous Management Information Bases (MIBs) that organize and
distribute information for a variety of network management devices.
You can use the MIB table that supports Cisco CallManager to provide all of the management
interfaces for monitoring and managing your Cisco CallManager network. This MIB table is
periodically updated, reflecting the current status of your Cisco CallManager network:
■ CISCO-CCM-MIB—To perform network management, you can use the CISCO-CCM-MIB

to get provisioning and statistical information about Cisco CallManager, its associated
devices (for example, IP phones and gateways), and its configuration.
■ CISCO-CDP-MIB—You can use the Cisco CallManager SNMP agent to implement the
Cisco Discovery Protocol MIB, CISCO-CDP-MIB. This MIB enables Cisco CallManager to
advertise itself to other Cisco devices on the network, allowing discovery of other Cisco
CallManager installations on the network.
NOTE To get more information about the MIB tables, go to Cisco.com and search for CISCO-
CCM-MIB.
Syslog Overview
Syslog allows logging of events across the network to various destinations. It provides an orderly
presentation of information that assists in the diagnosis and troubleshooting of system problems.
Theses messages can be saved in a file or sent to, for example, a CiscoWorks server, a third-party
syslog server (such as Kiwi Syslog Daemon), or the host itself.
TIP Kiwi Syslog Daemon is a freeware Windows syslog server. It receives, logs, displays, and
forwards syslog messages from hosts, such as routers, switches, UNIX hosts, and any other
syslog-enabled device. For more information, go to http://www.kiwisyslog.com.
When the local host is used, Remote Syslog Analyzer Collector (RSAC) software must be
installed. RSAC can be installed on a remote UNIX or Microsoft Windows 2000 or Windows NT
machine to process syslog messages.
Syslog Configuration in Cisco CallManager

Cisco CallManager syslog messages are configured in Cisco CallManager Serviceability. To
configure alarms, use the Cisco CallManager Serviceability web page and select Alarm >
Configuration. After selecting your server and the CallManager service, the Alarm Configuration
window shown in Figure 34-3 opens. To enable syslog messages, simply check the Enable Alarm
check box under the Syslog Trace section and enter the IP address of your syslog server.
Figure 34-3 CallManager Syslog Configuration

Table 34-2 lists alarm events that can be configured in the alarm trap.
Table 34-2 Configurable Alarm Events
Name Destination Description

Enable Alarm for Windows 2000 Event Viewer program. The program logs Cisco CallManager
Event Viewer errors in the application logs within Event Viewer, and provides a description
of the alarm and a recommended action.
Enable Alarm for Cisco CallManager Syslog capabilities. Check the Enable Alarm check box in
Syslog the Syslog Trace area of the Alarm Configuration window to enable the syslog
messages and configure the syslog server name. If this destination is enabled
and no server name is specified, Cisco CallManager sends syslog messages to
the local host. Cisco CallManager stores alarm definitions and recommended
actions in a Standard Query Language (SQL) server database. The system
administrator can search the database for definitions of all the alarms. The
definitions include the alarm name, description, explanation, recommended
action, severity, parameters, and monitors. This box is unchecked by default.
Enable Alarm for The SDI trace library. Ensure that this alarm destination is configured in Trace
System configuration of Cisco CallManager Serviceability.
Diagnostic
Interface (SDI)
Trace
Enable Alarm for The SDL trace library. This destination applies only to the Cisco CallManager
Signal and Cisco CTIManager services. Configure this alarm destination using Trace
Distribution SDL configuration.
Layer (SDL)
Table 34-3 lists alarm levels used by Cisco CallManager.
Alarm Event Levels
Level Name Description
7 Emergency This level designates the system as unusable.
6 Alert This level indicates that immediate action is needed.
5 Critical Cisco CallManager detects a critical condition.
4 Error This level signifies that an error condition exists.
3 Warning This level indicates that a warning condition is detected.
2 Notice This level designates a normal but significant condition.
1 Informational This level designates information messages only.

The Cisco CallManager Serviceability Alarms window provides a web-based interface that has
two main functions:
■ To configure alarms and events
■ To provide alarm message definitions
Both functions assist the system administrator and support personnel in troubleshooting Cisco
CallManager problems. Alarms can be configured for Cisco CallManager servers in a cluster and
for services for each server, such as Cisco CallManager, Cisco TFTP, and Cisco CTIManager.
Alarms can be forwarded to a Serviceability Trace file. The administrator configures alarms and
trace parameters and provides the information to a Cisco TAC engineer. Administrators can direct
alarms to the Windows 2000 Event Log, syslog, SDI trace log file, SDL trace log file (for Cisco
CallManager and CTIManager only), or to all these destinations.
Dependency Records
In Cisco CallManager Administration, numerous configuration elements are referenced by other
elements (for example, a route pattern refers to a route list, which refers to a route group, which
refers to a gateway). In many cases, you cannot delete such elements if they are currently
referenced elsewhere in the system. It can be difficult and time consuming to find out which
configuration element is referencing the element that you are trying to delete. The mechanism that
allows you to determine, delete, change, or modify a record in Cisco CallManager is called
dependency records. Dependency records help you to determine which records in the Cisco
CallManager database use other records. For example, which devices (such as computer telephony
integration [CTI] route points or IP phones) use a particular calling search space?
To delete a record from Cisco CallManager, you can use dependency records to show which
records are associated with the record that you want to delete. You can then reconfigure those
records so that they are associated with a different record.
For example, the administrator tries to delete a device pool. A message is displayed that some
devices still use this pool. The administrator clicks the dependency records link to find out which
devices use this device pool.
Enabling Dependency Records

Because dependency records are disabled by default, they must be activated in the Cisco
CallManager Administration Enterprise Parameters window if you want to use the feature. Set the
Enable Dependency Records parameter to True to activate dependency records and display them
as an option in Cisco CallManager Administration. To access this record, use the Cisco
CallManager Administration page and navigate to System > Enterprise Parameters. An

illustration of this is shown in Figure 34-4.
Figure 34-4 Enabling CallManager Dependency Records
CAUTION Displaying dependency records leads to high CPU usage and takes some time
because it executes in a low-priority thread. If you are monitoring CPU usage, you might see
high CPU usage alarms. To avoid possible performance issues, display dependency records only
during off-peak hours or during the next maintenance window. Close and reopen the web
browser for the parameter change to take effect.
Accessing Dependency Records

To access dependency records from a Cisco CallManager configuration window, click the
Dependency Records link, as shown in Figure 34-5. The Dependency Records Summary window
opens. This window displays the number and type of records that use the record that is shown in
the Cisco CallManager configuration window.
Figure 34-5 Accessing Dependency Records

Dependency Records link
NOTE If the dependency records are not enabled, the Dependency Records Summary window
displays a message stating that the Dependency Record feature is disabled (and also tells the
administrator how to enable this feature).
After you click the dependency records link in an administration window, you will see a list of all
records that refer to the item that you selected, as shown in Figure 34-6. This list is in summary
style and shows the depending records only by type and number. You can click an entry of the
summary list to view the detailed list of dependent records. You can click a single device in the
dependency records detail window to go to the configuration window of the device. To return to
the original configuration window, click the Back to <configuration window name> link.
Figure 34-6 Dependency Records Summary
Three buttons are available in the Dependency Records Summary window:
■ Refresh—Updates the window with the most up-to-date information.
■ Close—Closes the window but does not return to the Cisco CallManager configuration
window in which the user clicked the Dependency Records link.
■ Close and Go Back—Closes the window and returns to the Cisco CallManager configuration
window in which the user clicked the Dependency Records link.
Password Changer Tool

The Password Changer tool is used to change the CCMAdministrator, IPMASysUser,
CCMSysUser, or the Directory Manager accounts. The Password Changer tool is stored by default
on every Cisco CallManager server in the cluster. When you want to change the Directory
Manager password, you must change it on every Cisco CallManager server in the cluster
Password Changer Tool 775
(publisher and subscriber). For the other accounts, the password must be changed only on one
server in the cluster.
NOTE The CCMAdministrator account is used only when you are using MultiLevel
Administration (MLA). The Microsoft Windows administrator account is used when MLA is
not used. When you change the password for the Windows administrator account, you have to
use the new password when you log in to Cisco CallManager Administration. The Windows
administrator account cannot be changed with the Cisco Password Changer tool, whereas the
passwords for users in the DC Directory of Cisco CallManager can. When Cisco CallManager
is integrated in a Lightweight Directory Access Protocol (LDAP) directory, the users are
changed directly in the corresponding LDAP directory.
These users can be changed with the Password Changer tool:
■ CCMAdministrator—CCMAdministrator is used as the administrator account when you

are using MLA.
■ CCMSysUser—Cisco Extended Functions, Cisco Tomcat, and Cisco CallManager

Extension Mobility services use a special user, cn=CCMSysUser and mail=CCMSysUser
(Netscape) or SAMAccountName=CCMSysUser (Microsoft Active Directory), to
authenticate with Cisco CallManager.
■ IPMASysUser—IPMASysUser is used by Cisco IP Manager Assistant (IPMA) to

authenticate with Cisco CallManager.
■ Directory Manager—Directory Manager is the superuser account for the integrated LDAP
database in Cisco IP telephony systems. In Cisco CallManager, it is the DC Directory.
Using the Password Changer Tool

To start the Password Changer tool, complete these steps:
Step 1 Choose Start > Run > CCMPWDChanger in Cisco CallManager.

Step 2 After some time, you will be able to enter the DC Directory administrator
password (not the administrator password that is used to access the server).
Click Next.
Step 3 Choose the user whose password you want to change, as shown in
Figure 34-7.
Figure 34-7 Selecting the User
Step 4 Enter the new password twice.

Step 5 A message notifies you that the password for the user has been successfully
changed. Click OK to close the window.
Step 6 After finishing all password changes, click Exit to leave the Password
Changer tool. The next time that you log in to Cisco CallManager
Administration, the new passwords take effect.
Cisco Dialed Number Analyzer

The Dialed Number Analyzer is installed as a plug-in for Cisco CallManager. The tool allows you
to test a Cisco CallManager dial plan configuration prior to deploying it. The tool simulates
internal-to-internal calls and internal-to-external calls. You can also use the tool to analyze dial
plans after the dial plan is deployed.
A dial plan can be very complex, involving multiple devices, translation patterns, route patterns,
route lists, route groups, calling- and called-party transformations, and device-level
transformations. Because of this complexity, a dial plan can contain errors. You can use the Dialed
Number Analyzer to test a dial plan by providing dialed digits as input. The tool analyzes the
dialed digits and shows details of the calls. You can use these results to diagnose the dial plan,
identify problems, if any, and tune the dial plan before it is deployed.
The Cisco Dialed Number Analyzer runs as a service that can be accessed from the server on
which it is installed or from a remote PC. It runs as a low priority and does not affect Cisco
CallManager performance. With the Cisco Dialed Number Analyzer tool, you can analyze various
types of calls, such as IP phone-to-IP phone, gateway-to-IP phone, IP phone-to-gateway, and
gateway-to-gateway calls. Furthermore, you can analyze calls to feature-specific patterns, such as
CTI route points or pilot points.
Installing Cisco Dialed Number Analyzer

The Dialed Number Analyzer (DNA) includes a separate executable file that is available in the
Cisco CallManager plug-ins window. You can install the Dialed Number Analyzer if Cisco
CallManager Release 3.3(4) or later has been installed. You can install the Dialed Number
Analyzer on any Cisco CallManager node in a cluster, either publisher or subscriber. Install the
Dialed Number Analyzer preferably on the publisher to use the actual SQL database that is used
by all Cisco CallManager servers. During the installation, you have to enter the Cisco
CallManager cluster private password phrase you configured during the Cisco CallManager
installation. When the Dialed Number Analyzer is installed, it installs as a service called Cisco
Dialed Number Analyzer. You can start and log in to the Dialed Number Analyzer from the server
on which it is installed or from a remote PC by using a web browser (Microsoft Internet Explorer
6.0 or later versions).
The Dialed Number Analyzer tool can be installed from the Cisco CallManager Plugins web page
at Application > Install Plugins > Cisco Dialed Number Analyzer. After you have installed the
tool, you can access it at https://<cmaddress>/dna, where <cmaddress> specifies the node name
or IP address of the device on which the Dialed Number Analyzer is installed. In the User Name
field, enter a valid user ID and a password (for the administrator account to get access to Cisco
CallManager Administration). The Dialed Number Analyzer window shown in Figure 34-8 should
appear.
NOTE The Windows administrator account must be used to access the Cisco Dialed Number
Analyzer even if you have the Cisco MultiLevel Administrator (MLA) installed with Cisco
CallManager.
Figure 34-8 Cisco Dialed Number Analyzer
Using the Cisco Dialed Number Analyzer

The analysis involves entering calling-party and called-party digits in the Dialed Number
Analyzer tool and choosing a calling search space for the analysis. The Dialed Number Analyzer
uses this calling search space and analyzes the dialed digits. You need not choose specific devices
or provide any other input. The Dialed Number Analyzer allows you to analyze a route pattern,
translation pattern, directory number (DN), or CTI route point. Furthermore, beginning with Cisco
CallManager Release 4.1, the Dialed Number Analyzer tool supports Call Coverage, H.323
FastStart, Hospitality, trunk-to-trunk transfer, Forced Authorization Codes (FAC) and Client
Matter Codes (CMC), BRI, Multilevel Precedence and Preemption (MLPP), U.S. Department of
Defense (DoD) requirements, Q signaling (QSIG), and time-of-day features.
As shown in Figure 34-9, a call between 4015 and 4010 with the calling search space IntLocCSS
is analyzed. The Calling Party and the Dialed Digits fields are mandatory. When you want to test,
for example, time-of-day routing, you can specify a time zone as well. After you have chosen all
values, click the Do Analysis button to start the analysis.
The result of the analyzed call between 4015 and 4010 is displayed in a compact version, as shown
in Figure 34-10. For complete information, click the Expand All button. The match result
information is the first point to check. Is the call routed correctly or is it restricted?
RouteThisPattern means that the call is routed correctly.
Figure 34-9 Dialed Number Analyzer Example
Figure 34-10 Dialed Number Analyzer Results

Quality Report Tool

QRT is a voice-quality and general problem-reporting tool for Cisco CallManager IP Phones. The
Cisco Extended Functions service supports the QRT feature. The QRT Viewer, located in the Tools
menu of Cisco CallManager Serviceability, allows administrators to filter, format, and view
problem reports that are generated.
Administrators can configure Cisco IP Phones with QRT, which is installed as part of the Cisco
CallManager installation, so that users can report problems with IP phone calls. Users report
issues by using a Cisco IP Phone softkey that is labeled QRT. Any Cisco IP Phone that supports
an HTTP web server also includes support for QRT. The IP phone must be in the Connected,
Connected Conference, Connected Transfer, or On Hook state for the QRT softkey to be available.
QRT is a feature that extends to Cisco IP Phones as a Microsoft Windows NT service. Cisco
Extended Functions, which supports the QRT feature, must be enabled in the Cisco CallManager
service activation window. To enable Cisco Extended Functions, choose Cisco CallManager
Serviceability > Tools > Service Activation and activate the service called Cisco Extended
Functions.
When a user presses the QRT softkey, as shown in Figure 34-11, QRT displays a feedback
window. It is possible that while the user is interacting with the QRT screen, another application,
such as Cisco Call Back or Cisco IP Manager Assistant (IPMA), or function keys, such as settings,
directories, messages, and so on, could overwrite the QRT screen. In this situation, QRT cannot
send the feedback. To send feedback in such cases, the user has to press the QRT softkey again.
If a user presses the QRT softkey to generate a report and forgets to stop the logging process (the
user must manually stop the logging process by pressing the End softkey on the Cisco IP Phone),
the QRT tool periodically checks all IP phones that are generating reports and closes them. This
action prevents the device from consuming large amounts of resources that, over time, could
impact CTI performance. Currently, the default setting is to check every hour and to close devices
that have remained open for more than an hour.
QRT records are written only when the end user presses the QRT softkey on the IP phone, selects
a problem, and sends the report to the administrator. Otherwise, no records are written and the
administrator cannot troubleshoot the problem. The figure shows the logging process of an audio
call between two IP phones.
Figure 34-11 QRT Submission Process

End user recognizes Administrator uses QRT
quality issue during tool to verify issues in
the call. IPT network.
End user presses QRT

button on IP Phone.
QRT Result Page
Depending on how the system administrator configured QRT for the Cisco IP Phone, users can use
the QRT softkey in either of the two ways described in Table 34-4.
Table 34-4 QRT Interaction
Issue What To Do
To quickly report an audio Your IP phone system will collect and log call data for the
problem with a current call while current call and route this information to your system
on a call, press More > QRT. administrator.
To report a problem with your Select the problem that you want to report from the list of
phone calls, press More > QRT. problem categories. Some problem categories include a reason
code that you can select to provide more details about the
problem. Your IP phone system will store the information in a
database or file and the administrator can run reports to
diagnose the problem.
Activating the QRT Softkey

The system administrator can temporarily configure a Cisco IP Phone with QRT to troubleshoot
problems with calls. Reconfigure the softkey layout for the IP phone (choose Cisco CallManager
Administration > Device > Device Settings > Softkey Template > softkey_template_name >
Softkey Layout) to activate the QRT softkey, as shown in Figure 34-12. Users can now report
problems by using the QRT softkey during or after a call.
Figure 34-12 Adding the QRT Softkey
When the administrator modifies the softkey templates to activate the QRT softkey, the softkey
should be added to the following call states:
■ Connected
■ Connected Conference
■ Connected Transfer
■ On Hook
Viewing QRT Logs

Administrators can view the IP phone problem reports that are generated by QRT by using QRT
Viewer.
NOTE QRT collects streaming data only once per call. Therefore, if user A calls user B and
both submit reports for that call, only the first report includes streaming data (for example, delay,
jitter, and ports being used).
Use the following steps to view the QRT logs:
Step 1 To open the IP phone Problem Reporting window, choose Cisco

CallManager Serviceability > Tools > QRT Viewer.
Step 2 Choose the Cisco CallManager server for which you want to view a
problem report. Enter a start and end date in the Date fields. In the Time
fields, you can specify the time for those dates, if needed. An illustration of
this is shown in Figure 34-13.
Figure 34-13 Generating a QRT Query: Time and Date Fields
Step 3 Click the Get Logs button.

Step 4 From the Extension Number, Device, and Category drop-down menus,
choose the extension numbers, the devices, and the problem categories that
you want to include in the report, as shown in Figure 34-14.
Figure 34-14 Generating a QRT Query: IP Phone Filters
Step 5 From the List of Fields drop-down menu, select the fields that you want to
include in the report and click the down arrow to move the selected fields
to the Selected Fields pane.
The QRT output displays all IP phone problems for the specified time frame. For the filters set in
the query, all issues within the time frame are displayed. In Figure 34-15, two QRT reports
matched the submitted query. One of the QRT reports came from extension 4015 and the other
from extension 4010.
When you choose All from the Category field, all problems are displayed. The Category and
Reason Code columns are the areas that you should look at first. These two columns describe the
problem and the reason. The output can be saved in a Comma-Separated Values (CSV) file.
NOTE Because QRT reports are based only on entries in the QRT database and records are
added to that database only if a user presses the QRT softkey, QRT cannot detect any problems
on its own; it relies completely on user activity.
Summary 785
Figure 34-15 QRT Report Sample
Summary
This chapter discussed the use of additional CallManager management tools, such as CiscoWorks
ITEM, SNMP, and syslog. CiscoWorks ITEM is a suite of tools to monitor devices in an IP
telephony network. SNMP and syslog are standards-based monitoring and alerting protocols used
across nearly all Cisco platforms.
Cisco CallManager dependency records help in determining which devices are associated with
other devices. The Password Changer tool can change passwords of CCMSysUser,
CCMAdministrator, IPMASysUser, and Directory Manager with an easy-to-use interface. The
Dialed Number Analyzer tool helps diagnose, tune, and trace dial plans. QRT helps administrators
identify and resolve problems in an IP telephony network. End users can use QRT to send errors
and call information to the administrator.
Review Questions
1. What is the purpose of the dependency records?

a. Dial plans can be analyzed with the dependency records.
b. It helps to find links between devices that keeps you from deleting a record in Cisco
CallManager.
c. Users and profiles that are interacting with each other can be displayed with the depen-
dency records.
d. Collaborating system services in Cisco CallManager can be displayed with the depen-
dency records.
2. Which user password can be changed with the Password Changer tool?
a. IPMAAdminUser
b. CCMAdminUser
c. CCMAdministrator
d. CCMUser
3. The Cisco Dialed Number Analyzer can be used to analyze which of the following?
a. Dial plans in an IP telephony environment
b. Erlang b and Erlang c values
c. CARs
d. Dialed service numbers called by users
4. Complete the following sentence to make it true: When using QRT, reports are
created __________.
a. Automatically for all users
b. In a round-robin fashion for all users
c. When an administrator activates QRT Viewer
d. When an end user presses the QRT button on the IP phone
5. Which SNMP version supports encryption and authentication?
a. SNMP Version 1
b. SNMP Version 2
c. SNMP Version 3
d. SNMP Version 5
6. What is the drawback of using CallManager dependency records?

a. The dependency record function only shows devices that are directly related.
b. Dependency records can lead to high CPU utilization.
c. Dependency records require CDP to function.
d. Dependency records only work with Cisco equipment.
7. When using the Password Changer tool, the passwords of the specified user accounts are
changed _________.
a. In the LDAP Directory of a single server
b. In the LDAP Directory of all servers in the cluster
c. In the Windows user database of a single server
d. In the Active Directory for all servers in the cluster
8. How can you access the Password Changer tool?
a. Using the Serviceability Tools menu
b. Using the CallManager Administration Tools menu
c. By typing CCMPWDChanger from a Windows Run prompt
d. By selecting CCMPWDChanger from the Windows Start menu
9. You receive an alarm level 5 message coming in from the Cisco CallManager service. What
well-known name is this alarm level assigned?
a. Emergency
b. Error
c. Critical
d. Alert
10. 10)Which of the following is the valid way to access Cisco DNA?
a. Using a skin sample from John Chambers
b. By using the Application menu in CallManager Serviceability
c. By using the Application menu in CallManager Administration
d. By entering the URL https://<cm address>/DNA
Part VIII: Appendix
Appendix A Answers to Review Questions

APPENDIX A
Answers to Review Questions
Chapter 1
1. B
The Cisco Unified Communications strategy focuses around designing network equipment
equipped with the hardware resources and software features to handle a converged network
environment.
2. B
The Cisco CallManager is part of the call-processing layer. This layer is responsible for the
call routing intelligence behind the IP telephony network.
3. A, B, and D
The Cisco CallManager is responsible for call processing, signaling and device control, dial
plan administration, phone feature administration, directory services, and a programming
interface to external applications.
4. C
The Cisco CallManager uses the proprietary Skinny signaling to communicate with the
Cisco IP Phones.
5. A, B, and D
The Cisco CallManager is responsible for handling all call-processing functions between
Cisco IP Phones. The only communication element Cisco IP Phones are capable of
handling alone is the streaming of RTP audio (spoken voice) between devices.
6. A and C
Cisco CallManager runs using Windows 2000 and SQL 2000 as foundation operating
system components. Windows 2000 provides base functionality, whereas SQL 2000 acts as
the database store and replication agent for the information that drives the IP telephony
network.
7. A
Cisco includes a very simple user-management system called DC Directory to store user
information for the IP telephony network.
792 Appendix A: Answers to Review Questions
8. C
The Cisco MCS 7825 can support a maximum of 1000 IP Phones. Because of its lack of
redundant hardware, it is typically used in lab environments and small network deployments.
9. False
Cisco CallManager requires you to maintain a user directory primarily for advanced
configuration features such as Extension Mobility and the IP SoftPhone.
10. B and C
By default, Cisco IP Phones use Skinny signaling to communicate with the Cisco
CallManager server. Optionally, you can download a SIP image for the phone to allow it to
operate under the Cisco SIP Proxy Server or with a third-party management system.
Chapter 2
1. A, B, C and D
All four of these are valid Cisco CallManager designs. E is incorrect because you need at least
one call-processing agent in each cluster.
2. A
A 1:1 redundancy design offers maximum redundancy because each Cisco CallManager
involved in call processing has a dedicated backup server. This model is often very cost
prohibitive to most companies.
3. C and D
Although using a single cluster over WAN connections has very strict delay and bandwidth
requirements, there are plenty of benefits. A common dial plan across all sites allows the
network administrator to create a single dial plan, which replicates to all remote sites. In
addition, if a Cisco CallManager fails, the IP Phones can failover to a Cisco CallManager
server at another site, maintaining their full feature set.
4. B and C
Every Cisco CallManager cluster should have at least two servers, which allows call
processing to continue should a single server fail. The data of the voice network is made
redundant through the SQL database replication that occurs between servers in the cluster.
5. B
There can only be a single publisher server for the entire Cisco CallManager cluster. This is
a SQL 2000 database restriction as the SQL publisher maintains the only writable copy of the
database.
Chapter 3 793
6. C
A Cisco CallManager cluster supports a single SQL publisher and up to eight SQL subscriber
servers.
7. B
Survivable Remote Site Telephony (SRST) is absolutely necessary in a centralized Cisco
CallManager design. SRST supports the remote IP Phones, providing PSTN calling access if
the WAN connection fails.
8. True
If the Cisco CallManager server at a location fails, the IP Phones can failover to another Cisco
CallManager server at a remote site.
9. A and C
The H.323 gatekeeper and SIP proxy server have the ability to be the centralized point of dial
plan information. This keeps the network administrator from having to re-create the network
dial plan at each of the sites.
10. E
Cisco will support a maximum cluster size of 30,000 IP Phones in Cisco CallManager 4.0.
This large cluster size was introduced in Cisco CallManager version 3.3 and was a
tremendous increase from earlier Cisco CallManager versions.
Chapter 3
1. C
Cisco CallManager installs using Windows 2000 as a foundation operating system.
2. D and E
By default, all Subscriber servers install with the IIS web services enabled. This opens a point
of access for a potential intruder. In addition, it consumes resources on the server that could
be used for other mission-critical services that the Cisco CallManager provides.
3. B
The Hardware Detection disk allows the installation program to determine which operating
system software image is appropriate for your MCS platform. The Hardware Detection disk
only recognizes Cisco-approved hardware.
4. C
To install multiple servers into a cluster, you must first configure hostname resolution. The
most common method is through DNS; however, if DNS is unavailable, you can statically
type a LMHOSTS file on each of the servers, which is nothing more than a text file mapping
hostnames to IP addresses.
5. A, B, and E
The following services are not typically used on a production Cisco CallManager server
platform:
— DHCP client
— Fax service
— FTP Publishing Service
— Smart Card
— Smart Card Helper
— Computer browser
— Distributed File System
— License Logging Service
6. C
The Set Default button activates the minimum services necessary to run the Cisco
CallManager in a single-server configuration.
7. B
After the Cisco CallManager installation has completed, you should activate all services the
Cisco CallManager will use. If you apply any configuration before you activate the services,
unpredictable results can occur.
Chapter 4
1. A and D
Modern Cisco IP Phones support the G.711 (64 Kbps) and G.729 (8 Kbps) codecs. The old
Cisco IP Phones acquired directly from Selsius supported the G.711 and G.723 (6.3 Kbps)
codecs.
Chapter 4 795
2. A and D
Because Cisco was one of the first companies to market with IP Phone technology, the Power
over Ethernet (PoE) standard was not fully developed. Because of this, Cisco created its own,
prestandard inline power. In modern times, the IEEE has solidified the 802.3af inline power
standard. The Cisco IP Phone 7961G supports both of these inline power standards.
3. B, C, and E
The Cisco IP Phone 7905 and Cisco ATA 186 do not have built-in Ethernet switches; rather,
they terminate a 10BASE-T Ethernet connection. The Cisco IP Phone 7912 is exactly the
same as the 7905 but supplies the built-in 10/100 Ethernet switch. The same is true of the
Cisco ATA 188.
4. C
The Cisco 7980 is the only IP Phone that provides built-in video-processing capabilities.
5. A, C, D, and B
When the IP Phone goes through the boot process, it initially obtains inline power using the
Cisco prestandard or IEEE 802.3af. From there, it loads its built-in firmware image, which
directs it to receive VLAN information through CDP. After it has a VLAN assignment, it
broadcasts for a DHCP address, which also contains the TFTP server hostname or address. It
then contacts the TFTP server to download its configuration file.
6. C
DHCP Option 150 is used to deliver the IP address of a TFTP server to a Cisco IP Phone.
DHCP Option 66 is used to deliver the hostname of a TFTP server to a Cisco IP Phone.
7. A
Most Cisco IP Phones (all except the 7902G) support XML processing, which is an industry
standard for storing data with processing and formatting options.
8. A and D
The Cisco IP Phone 7961 supports both the Cisco prestandard and 802.3af inline power,
whereas the 7960 supports only the Cisco prestandard. In addition, the Cisco 7961 IP Phone
adds support for Gigabit Ethernet connections.
9. C
The Cisco ATA 188 provides two Foreign Exchange Station (FXS) ports that allow you to
convert up to two analog devices into VoIP devices.
10. A
The Cisco IP Phone 7902G is the most cost-effective, single-line IP Phone available from
Cisco. It is the only IP Phone not equipped with a display screen, so the features are limited
to that of a standard analog-style device.
Chapter 5
1. B, C, C, E, and F
Although a device pool can assign other settings, Cisco CallManager absolutely requires
these elements to create a device pool.
2. B
A Cisco IP Phone 7960 only has a total of six line/speed dial buttons available, which
immediately eliminates answers C and D. The IP Phone must have at least one line button
assigned to it, which eliminates answer A. Answer B represents the default phone button
template assigned to the Cisco IP Phone 7960.
3. B
To configure auto-registration, you must access the System > Cisco CallManager
configuration screen and access the server for which you want to perform the auto-
registration. Once there, you can define the range of extensions that the Cisco CallManager
should assign to the Cisco IP Phones.
4. D
The Cisco CallManager ties configuration information to IP Phones through the MAC
address. This is one of the most difficult aspects of adding Cisco IP Phones to the SQL
database.
5. A
The Region configuration allows you to define the codec used to communicate within the IP
Phone’s region (typically a LAN environment) and with other regions (typically
communicating over the WAN).
6. B
By changing the hostnames to IP addresses under the System > Server menu, the Cisco
CallManager will automatically update the TFTP server configuration files to use IP
addresses rather than hostnames. This eliminates an IP Phone’s reliance on DNS.
Chapter 6 797
7. D
The Cisco IP Phone uses a list of up to three redundant Cisco CallManagers for registration.
This is known as its CallManager Group.
8. C
You cannot modify or delete the default CMLocal date/time group. IP Phones using this date/
time group will use the same time and time zone the Cisco CallManager is using.
9. B
The softkey template governs the feature buttons available to the user on the bottom of their
IP Phone display.
10. A, B, and C
All of these three methods allow you to obtain the MAC address of the Cisco IP Phone.
Although the phone does also support a web interface where the MAC address can be found,
it does not support Telnet capabilities.
Chapter 6
1. B
To access the user web pages, you can use the URL http://<CCM IP Address>/CCMUser/
logon.asp. The most recent versions of CallManager also allow you to enable SSL support for
the user web pages.
2. C
DC Directory is compatible with the Lightweight Directory Access Protocol (LDAP). This is
a standard method for querying and modifying user database storage.
3. False
The IP telephony network will function correctly without the need for a user database.
However, to gain advanced application support and to allow users to modify their own phone
features through a web-based interface, you should add user database support to the network.
4. B, C, D, E, and H
These fields are all required when creating a user who is able to use a Cisco Softphone
application. The “SoftPhone Support Enabled” is not a valid option on the user creation web
page.
5. A and C
To configure a Cisco IP Phone through the CCMUser web pages, a user must have a valid user
account created for them in the Cisco CallManager database and that account must be
associated with at least one Cisco IP Phone.
6. D
To use both the personal address book and fast dial features, the user must first subscribe to
the appropriate XML services. These services will then be made available by pressing the
Services button on the Cisco IP Phone.
7. A
The default Message Waiting Lamp policy for the Cisco CallManager system configured on
each Cisco IP Phone is to use the system policy. The system policy default setting is to always
light the lamp and prompt the user.
8. False
Cisco CallManager only ships with one language: English. You can download additional
languages from Cisco as needed.
9. C
After subscribing to the fast dial service, a user can configure their IP Phone with up to 99 fast
dial codes associated to users in their personal address book.
10. A
The CCMUser pages only provide the Call Forward All option to the user. Call Forward Busy
and Call Forward No Answer are available from the CCMAdmin pages.
Chapter 7
1. C, D, and F
You can use Cisco’s BAT to insert IP Phones, users, device profiles, IPMA managers and
assistants, Catalyst 6000 FXS ports, VG200 analog ports, client matter codes, forced
authorization codes, Call Pickup groups, and phone certificates.
2. B
Auto-registration is used to allow newly installed Cisco IP Phones to gain a temporary
extension that the user or administrator can use to dial in to the TAPS extension to configure
their IP Phone.
3. B and C
To perform a successful device import using BAT, you must use a comma-separated value
(CSV) text file to define unique device values and a valid device template specifying common
device settings.
4. D
This Excel spreadsheet eases the CSV file creation process and is always stored in the
directory C:\CiscoWebs\BAT\ExcelTemplate.
Chapter 8 799
5. A
The Cisco BAT utility will only install on the Publisher server because it makes changes
directly to the SQL database.
6. A
The device template can define many line settings, but BAT will only configure the lines you
have specified in the CSV file. The trouble occurs when you have more lines defined in the
CSV file than exist in the device template. If this situation occurs, BAT will cut off the number
of lines on the device to equal that of the device template.
7. C
The Create Dummy MAC Address check box generates random MAC addresses for the IP
Phones and saves them in the CallManager database. The administrator can then manually
update the entries or use TAPS to automate the process.
8. B
You can accomplish bulk updates using a query in the BAT utility. A query can match devices
based on any number of criteria. You can then specify the setting you want to change on the
matched devices (music on hold, in this case).
9. B
TAPS relies on the Customer Response Application, which manages the automated prompt
and update process for the user.
10. A and D
CSV files are simply text files containing data separated by commas. Cisco has included the
Microsoft Excel template to make the generation of these files easier. However, an
administrator can define their own CSV format by manually creating a CSV file using a
simple text editor, such as Microsoft Notepad.
Chapter 8
1. E
The syntax to turn on PoE on a CatOS-based switch is set port inline power mod/port auto.
2. B, C, and D
Cisco Catalyst switches have the ability to provide PoE (both 802.3af and the Cisco
prestandard) and handle CoS tagging on incoming packets (both on the data and voice
VLANs).
3. A, B, and D
To support dual VLANs on a single port, you must configure the port with voice and access
VLANs and use the trunk tagging mechanism (802.1Q) to mark the packets. Of course, the
IP Phone must also have an internal switch to support an additional device (such as a PC)
connection.
4. A, C, and E
Cisco IP Phones can be powered through wall power (using an additional power brick sold
separately from the IP Phone), Cisco prestandard PoE, or the Cisco Power Patch Panel
(midspan power injection). All Cisco IP Phones do not yet support the IEEE 802.3af PoE
standard. This support will be added as newer models are released.
5. D
802.3af is the IEEE industry standard for PoE.
6. C
The Catalyst 3750 uses the NativeIOS operating system. The set-based syntax only applies to
the CatOS. In addition, there is no “on” mechanism for inline power, rather, only an auto-
detect mechanism exists.
7. B and C
Dual VLAN configurations can only be accomplished using the 802.1Q tagging mechanism.
When deploying this configuration on a 6500 series switch running the CatOS, you must use
the auxiliary VLAN configurations. Voice VLAN configurations apply to the NativeIOS.
8. D
When you enable the auxiliary VLAN on a CatOS switch, the tagging mechanism handles the
trunk configuration for you. No further configuration is needed.
9. C
A Cisco IP Phone tags all voice traffic using a CoS marking of 5. This is the highest user-
defined marking Cisco recommends.
10. B
Attached devices are considered untrusted (by default) by a Cisco IP Phone. This means that
the Cisco IP Phone will mark traffic with a CoS value of 0, unless otherwise specified by the
administrator.
Chapter 9 801
Chapter 9
1. A and D
Foreign Exchange Station (FXS) ports connect to analog station endpoints (such as fax
machines, modems, and telephones). Foreign Exchange Office (FXO) ports connect to the
PSTN or act as PBX trunk connections. The third type of analog interface (E&M) is not listed
in the question.
2. B and D
Cisco Catalyst switches have the ability to provide PoE (both 802.3af and the Cisco
prestandard) and handle CoS tagging on incoming packets (both on the data and voice
VLANs).
3. B and C
All Cisco IP Phones use Skinny signaling to communicate with the Cisco CallManager, but
only the router-like device that uses Skinny is the VG200 series. This is because the VG200
series acts as a number of end devices with numerous FXS ports.
4. D
The correct wildcard to use on voice gateway devices is the “.” which matches any dialed
digit. When configuring route patterns on the Cisco CallManager, you use the “X” wildcard
to accomplish this same thing.
5. C
The Call Classification feature gives you the ability to distinguish and monitor OnNet and
OffNet calls through the gateway. Using this information, you can prevent toll-fraud
techniques using conference calls and transfers.
6. B
When configuring an MGCP router, you can use the syntax mgcp call-agent ip_address to
designate the primary Cisco CallManager. You can use the ccm-manager redundant-host
syntax to designate the secondary and tertiary Cisco CallManager servers. The tertiary
CallManager in this syntax is 10.1.1.6.
7. A and B
You can configure a switch running the NativeIOS using the same configuration as you would
a standard voice gateway. Non-IOS MGCP is used if the 6500 Catalyst switch is running the
CatOS operating system.
8. B
Trunk configurations are logical links to other networks. They have the ability to determine
the location of an endpoint, but do not carry voice traffic. Gateways also act as logical links
to other networks, but the CallManager also uses them to route voice traffic.
9. B
Gatekeepers ease configuration of Intercluster Trunk connections by providing a central point
of trunking. Without a gatekeeper, every CallManager cluster must have an Intercluster Trunk
connection to every other cluster in a full-mesh relationship. Unfortunately, gatekeepers also
introduce a single point of failure in the network, which is why you should consider additional
failover mechanisms for this device.
10. C
CallManager uses the H.225 Registration, Admission, and Status (RAS) protocol to
communicate with the gatekeeper. You can think of this as a conversation protocol that allows
the CallManager to determine the location of an endpoint and the authority to send the call.
Chapter 10
1. D
CallManager route plans are always configured from the bottom-up. Devices/trunks come
first, then route groups, route lists, and, finally, route patterns.
2. False
You can only configure digit manipulation to occur at the route list and route pattern levels.
This is why you should group similar devices together in the same route group: All devices in
the route group will have the same digit manipulation settings applied.
3. A
After a device has been assigned to a route group, it is made unavailable to any other
configurations. If you want to use a device for multiple route patterns, associate its route
group with multiple route lists. You can only associate devices directly with route patterns if
you are not already using those devices in a route group.
4. C
Voice gateways are used to bridge VoIP networks to either non-VoIP networks (such as the
PSTN) or to other VoIP networks.
5. A
Route groups contain a list of gateways to use in precedence order.
6. B, C, and D
The ! wildcard represents a variable-length string, the . wildcard represents a single digit, and
the @ wildcard represents the entire NANP. The * is not a wildcard as it represents just the *
digit on a telephone keypad.
Chapter 11 803
7. C
Route lists contain an ordered list of route groups that CallManager should use when it
matches a route pattern.
8. E
When digits are contained within braces, they match only a single digit in the extension. In
this case, [45] could be read in plain English as, “Four or Five” (mentally insert a comma
between the digits).
9. B
To stop the secondary dial tone, you need to remove the check from the Provide Outside
Dial-Tone check box for the route pattern. It is checked by default when the route pattern is
created.
10. C and D
The X wildcard matches any dialed digit, whereas the ! wildcard matches any variable-length
dial string (zero digits or more). By placing the 7XX before the !, the CallManager expects to
see at least three dialed digits followed by any number of digits.
Chapter 11
1. D
Translation patterns are defined to provide a “translation of last resort.” CallManager
processes them just before it routes a call. After the translation pattern has modified the dialed
digits, CallManager sends the transformed digits through digit analysis again.
2. C
The DN 0111 passes through the right-justified X wildcards. The result is that the
CallManager prepends the 972555 to the 0111 and sends 9725550111 as the CLID
information.
3. B
The predot digit discard instructions remove all dialed digits before the period.
4. A
You must apply the route filter to either a route pattern or a translation pattern for it to take
effect. After you have applied it to a route pattern, you must select either Route This Pattern
or Block This Pattern to define the allow or deny action.
5. A
The 11D@10D digit discard instruction takes an 11-digit long-distance number and converts
it to a 10-digit number. This is typically used for toll-bypass purposes.
6. D
Applying a calling-party transformation mask of 8 causes all caller ID information to be
replaced and become just the digit 8. To prefix an 8 using a transformation mask, wildcard
digits (X) would be necessary.
7. B
Called-party transformation masks always transform the dialed-number information.
8. D
The order of application of digit translation applies the transformation mask before the prefix
digits, causing the resulting caller ID to display as 5555123.
9. D
In this case, you have created a routing loop within the CallManager. The route pattern
prefixes an 8 to the 4xxx dialed numbers and the translation pattern strips the 8 and returns
the digits to the CallManager for processing. The call will loop ten times through the
CallManager route plan and then return a fast busy signal.
10. B
Calling-party transformations change caller ID information, whereas called-party
transformations change dialed-number information.
Chapter 12
1. B
If you have checked the Use Personal Preferences check box in the Hunt Forward Settings
section of the Hunt Pilot Configuration window, CallManager will check the Call Forward No
Coverage settings for the original called number that forwarded the call to the hunt pilot. If
you have not defined a number for this setting, the caller will receive a reorder tone when
CallManager attempts to forward the call.
2. A
Using the top-down distribution algorithm causes the Cisco CallManager to distribute the
incoming call to the first member of the line group and work down if the parties are
unavailable. If you were using the circular distribution algorithm, the call would have been
distributed to party 4 because party 3 received the last call.
Chapter 12 805
3. B
The maximum hunt timer always overrides the RNA reversion timeout configured on the line
group. In this case, the call would only hunt through the first three members of the group
before the call reached the maximum timer of 15 seconds.
4. A, B, and C
To configure a hunt group, you must first configure a line group, which groups member DNs
together. Then, you must set up a hunt list, which orders one or more line groups together.
Finally, you must configure a hunt pilot pointing to the hunt list.
5. B
Available lines are serving an active call but can accept a new call. Idle lines are those
categorized as not serving any calls at this time.
6. B
The broadcast distribution algorithm rings all members of the line group at the same time. The
first one to retrieve the incoming call will receive it.
7. A
The Stop Hunting selection causes the CallManager to use just the first member DN in the
line group and stop if the member is unavailable.
8. E
By checking the Use Personal Preferences check box, CallManager ignores any call forward
settings defined under the hunt pilot. Instead, it looks to the Call Forward No Coverage
section of the original called party that triggered the call to the hunt group.
9. A
Hunt groups are configured from the bottom up, just like the CallManager route plan. You
must first add member DNs to the CallManager configuration, then group them into a line
group, followed by the hunt list and hunt pilot.
10. B
By default, CallManager matches the hunt pilot DN and hunts through the members listed in
all line groups defined in the hunt list. This could take a considerable amount of time if there
are many members or line groups configured.
Chapter 13
1. C and E
When configuring time-of-day routing, you must apply a time schedule to the partition
containing the DNs you want to restrict by time of day. You must also configure the partition
for the correct time zone (or choose to use the local time zone of the phone placing the call).
2. B
You can apply time-of-day calling restrictions only to the calling party through a calling
search space. Because of this, the time period should be configured to allow calls from 08:00
to 17:00 Monday to Friday and the time zone of user A.
3. B, D, and E
Any DN-assigned item can be assigned to a partition. Phones and gateways are not assigned
DNs; phone lines, route patterns, and translation patterns are assigned DNs (directory number
is used synonymously with phone line).
4. A
The correct order of Class of Service configuration is as follows: DNs are placed in partitions;
partitions are placed in calling search spaces; and calling search spaces are assigned to the
calling device.
5. A, B, and C
If you have assigned a calling search space to both the IP Phone and its directory number, the
calling search spaces are combined where the directory number takes precedence. Route
patterns are not calling devices and cannot be assigned a calling search space.
6. B
CallManager processes the call by looking through the partitions in the calling search space
from the top of the list to the bottom. To save on processing time, you should place the more
frequently used partitions at the top of the user list. You can also use the ordered processing
feature to implement access-list-like calling restrictions.
7. D
Time periods are created first and added to time schedules, which are then assigned to a
partition.
8. A
CallManager will allow you to create identical, duplicate route patterns provided you have
assigned them to different partitions. Likewise, you can only assign a route pattern to one
partition. In this example, you must create two sets of PSTN route patterns. One set should be
added to the partition restricted by the 8:00 a.m. to 5:00 p.m. time schedule. The other set
Chapter 14 807
should be added to a partition not restricted by a time schedule. A calling search space
containing the restricted partition is then assigned to the users limited to 8:00 a.m. to 5:00 p.m.
PSTN calling and a calling search space containing the unrestricted partition is assigned to
the users able to reach the PSTN anytime.
9. C
Time-of-day restrictions can only be applied to partitions. These partitions are then added to
a user’s calling search space, at which point the CallManager applies them.
10. D
Time periods allow you to configure recurring days by selecting either “Repeat Every Week
From…” or “Repeat Every Year On…”. In this case, we would choose to Repeat Every Year
On December 25th.
Chapter 14
1. B
Configuring an IOS-based gatekeeper eliminates the need to configure a full-mesh
Intercluster Trunk relationship between disparate Cisco CallManager clusters in your
organization. Instead, all the clusters can trunk directly to the H.323 gatekeeper, which
provides a centralized point of admission and bandwidth control.
2. B
Router-based QoS mechanisms are not designed to handle CAC. If the CallManager routes
too many calls over the IP WAN, the voice quality for all calls will degrade.
3. B and D
In a distributed call-control environment, an independent device is needed to manage
admission and bandwidth control. This responsibility falls on the H.323 gatekeeper device.
The Cisco CallManager only manages these functions in a centralized call deployment.
4. C
The IP Phones are assigned to a device pool, which points to an SRST Reference. This SRST
Reference contains the IP address of the gateway running SRST or points the IP Phone to its
default gateway. This configuration is sent to the IP Phone when it first registers with the
Cisco CallManager.
5. D
Cisco CallManager deducts 24 kbps for each call in a location-based CAC deployment. Using
this calculation, you can find that five calls consume 120 kbps, fitting nicely over a 128-kbps
connection.
6. A, B, and E
When AAR reroutes a call over the PSTN, CallManager takes the original DN of the IP Phone
and combines it with the external phone number mask. The resulting number is then
prepended with whatever additional digits are specified under the AAR group and forwarded
over the PSTN.
7. C
To configure SRST, you first enter the command call-manager-fallback from global
configuration mode. The router then places you in a subconfiguration mode where the
additional settings can be specified.
8. B
AAR will use the external phone number mask configured under Phone B’s DN to complete
the call. This mask prepends the necessary digits to the DN, at which point the AAR group
prepends any additional digits that might be required to exit the PSTN gateway and
CallManager routes the call across the PSTN.
9. D
Before AAR configurations can take effect, the feature must be enabled clusterwide. This is
accomplished from the Service > Service Parameters > Cisco CallManager Configuration
window.
10. D
AAR is meant to act as a call redirection mechanism for location-based bandwidth
restrictions. It does not reroute calls during a WAN failure. Although any of these answers
could cause AAR redirection to fail, answer D is the most likely explanation.
Chapter 15
1. C
MRGLs are an ordered list of MRGs. The MRGs contain the media resources in the Cisco
CallManager cluster. By assigning a MRGL to a device, you grant access to whatever
resources are contained in the MRGs the MRGL contains.
2. B and C
The Media Resource Manager (MRM) manages all resources in the CallManager cluster. The
software-based resources are all grouped under the Voice Media Streaming Application,
which is activated in a Cisco CallManager server. Transcoding is one example of a hardware-
based resource.
Chapter 16 809
3. A
Hardware-based conference bridge resources are identified to the CallManager by their MAC
address. Most of the configuration for these resources takes place on the resource itself.
4. B
The Voice Media Streaming Application is a service that runs on the Cisco CallManager
servers. It allows the CallManager server to participate in all media resource functionality,
including conferencing, media termination point, music on hold, and annunciator.
5. B
Network hold MoH is played anytime a person is placed on hold because of a network-related
function. These functions include conference calls, transfers, and Call Park. User hold MoH
sources are only played when the user literally presses the hold button on their IP Phone.
6. A
The Cisco CallManager server creates individual unicast streams for IP Phone users.
Multicast support must be configured on a per-server and per-audio source basis.
7. B
When a MRGL is assigned at both the device pool and IP Phone level, the MRGL on the IP
Phone takes precedence. The MGM will use the MTP resource listed first in the first MRG of
the IP Phone MRGL.
8. B
By default, the media resources are assigned to the default MRG, which cannot be modified
or deleted. The MRM will use a round-robin load-balancing system for these resources.
9. E
Because of the amount of processor resources consumed by session transcoding, this function
is limited to hardware-based digital signal processor (DSP) resources.
10. C
The fixed audio source is always assigned audio source ID 51. This audio source ID must be
linked to a fixed sound device in the CallManager server before it becomes active.
Chapter 16
1. A and D
After an administrator has configured IP phone Services to be made available to the users, the
users can subscribe to them through the User Options pages (http://<ccm_server_ip>/
ccmuser). After the user has subscribed to the services, they can access them by pressing the
Services button on the IP phone.
2. B, C, and F
You can assign softkey templates to an individual IP phone (at the device level), to a group of
IP phones (at the device pool level), or to all auto-registered IP phones (at the device defaults
level).
3. B
The transfer functionality works just fine without administrator configuration, provided you
have provisioned MTP resources for the cluster.
4. A
When a user performs a direct transfer (DirTrfr), the user making the transfer stays out of the
call. This uses minimal cluster resources to accomplish the task.
5. B
Barge only functions when multiple users are using the same line (shared line appearance).
6. D
Auto answer capabilities are configured on a per-DN basis. They allow a user to answer a
ringing phone automatically using either a headset or speakerphone.
7. C
Michael will hear a dial tone. The busy trigger configuration for a shared extension will
restrict any additional incoming calls after the line usage has reached the defined number.
Only the maximum calls parameter can restrict outbound calls.
8. B
Because you cannot modify the built-in softkey templates in Cisco CallManager, the only
solution is to create a new softkey template, add the Call Back softkey to the Ring Out and
On Hook call states, and assign the softkey template to all users in your organization.
9. B
Group Call Pickup allows users to pick up a ringing phone located within their own Call
Pickup Group or another Call Pickup Group. Standard Call Pickup allows users to pick up a
ringing phone located within their Call Pickup Group. Other Group Call Pickup allows users
to pick up a ringing phone in their pickup group or another pickup group without dialing a
group number. Call Park allows users to pick up a call on hold by dialing the appropriate
extension.
10. C
Conference Barge (cBarge) allows a user to barge in a call and use dedicated conference
resources to maintain the call. Furthermore, the on-screen display shows the user entering a
conference. The standard Barge feature uses the built-in conference capabilities of Cisco IP
Phones and does not show the user as being in a conference call.
Chapter 17 811
Chapter 17
1. D
When configuring the Cisco CallManager Extension Mobility feature, you must create a
device profile for all users who plan on using the Extension Mobility capabilities. This device
profile is then associated with the user account.
2. A and C
CallManager transmits the MCID messages over a PRI connection. These messages are sent
to a MCID-Terminating (T) device on the PSTN from an MCID-Originating (O) device.
3. F
If a user enters the incorrect CMC, the CallManager requires them to hang up and attempt the
call again.
4. B, D, E, and G
By permitting the Connected Name and Line information, User A will be able to see the
information of the room they are calling. By restricting the Calling Name and Line
information, the hotel guest will not be able to see the information for User A.
5. A
The default behavior of the Extension Mobility service parameters prevents a user from
multiple logins using Extension Mobility.
6. D
Forced Authorization Codes (FAC) allows users to enter a PIN to obtain a specific
authorization. You can then assign route patterns a minimum authorization level to keep
nonessential personnel from dialing high-cost PSTN numbers.
7. C
Malicious Call Identification (MCID) is triggered by pressing the MCID softkey assigned to
the Softkey Template Connected call state.
8. D
CallManager 4.1 added the Executive Override MLPP parameter. In earlier versions of
CallManager, Flash Override was the highest assignable MLPP level.
9. D
The Ignore Presentation Indicators (Internal Calls Only) configuration setting from the device
configuration allows you to overrule caller ID restrictions for internal calls on a device-by-
device basis.
10. C
You only need to create a single logged out device profile for the Cisco IP Phones in your
organization. They will adopt this profile when you select the Use Current Device Settings for
the Logged Out profile of the device.
Chapter 18
1. 1. D, 2. C, 3. E, 4. A, 5. B
2. B
By selecting one of the hunt group members as an “always-route member,” the Cisco
CallManager will route the call to the destination when the queue is full or the hold time has
been reached. This destination is typically a voice mail or auto-attendant system.
3. C and E
The Cisco Telephony Call Dispatcher (TCD) must be active to dispatch calls to the active
Attendant Console clients. The CTIManager must be active to integrate the CTI-based
Attendant Console software with the CallManager system.
4. C
Only the longest-idle distribution algorithm can be configured from the Pilot Point
Configuration window. All others must be configured either from the Hunt Group
Configuration window or from the Attendant Console Configuration Tool.
5. C
A CallManager cluster can have a maximum of 32 hunt groups.
6. B
When you have configured Broadcast Hunting (using the Attendant Console Configuration
Tool), the incoming calls to the pilot point will be immediately placed into a queue that the
attendants can answer, using the Attendant Console client, as they become available.
7. B
The ac user must be created and associated with all Cisco IP Phones used by the Attendant
Console clients. As a good security practice, this username and password should be changed
using the Attendant Console Configuration Tool.
8. D
Accessing the Attendant Console Configuration Tool is a strange task indeed. You must first
dig through the hard drive to find the batch file and execute it for the Java-based utility to
appear.
Chapter 19 813
9. A
An operator can revert a parked call in many ways. The easiest is to right-click on the parked
call in the Attendant Console software and choose Revert Park.
10. D
Attendant Console users are not added through the user menu; rather, they are added through
the Service > Cisco CM Attendant Console > Cisco CM Attendant Console User window.
Users added through the normal User menu will not be available to select as Attendant
Console users.
Chapter 19
1. B
IPMA managers only have the ability to change the divert target from the IPMA Manager
Configuration page. This allows them to direct incoming calls to another attendant.
2. C
It’s so obvious, the question is almost difficult. Configuring Cisco IPMA in shared-line mode
requires lines to be shared DNs between the manager and assistant.
3. A
Shared-line mode is supported by the CallManager 4.x versions and enabled features not
previously supported in the proxy-mode, such as Call Join, Privacy, and Direct Transfer.
4. B and C
The three flagship Cisco IP Phones support IPMA: The Cisco 7940, 7960, and 7970.
5. B
Shared-line mode supports a maximum of 10 assistants per manager.
6. B
Cisco used the Tomcat server (developed under Apache) applied as a Windows service to
manage many of the additional services that previously required the Customer Response
Solution (CRS) software add-on.
7. C
After changing the service parameters, the MA Servlet must be restarted through the Tomcat
web server.
8. B
From this list, the first thing you must configure is the IPMA manager. You can then add
assistants and install the Assistant Console.
9. D
The IPMA Assistant Console is installed from a custom IPMA page managed by the Tomcat
web server. By accessing the URL http://<IPMA server>/ma/Install/IPMAConsoleInstall.jsp,
an automated script runs and places shortcuts on the assistant’s desktop and Start menu to run
the application.
10. B
The IPMA service communicates to all servers in a CallManager cluster by using the
CTIManager. It only requires a single CTI connection for all users in a cluster.
Chapter 20
1. A
All critical hot fixes and patches to the Cisco IP Telephony Operating System are posted on
Cisco.com for download within 24 hours of the announcement from Microsoft. All noncritical
hot fixes and patches are bundled for a consolidated, monthly release from Cisco.com.
2. B
Heuristic scanning should not be allowed because this feature can prevent CallManager
Administration web pages from operating correctly.
3. C
A headless (standalone) version of the Cisco Security Agent (CSA) is provided by Cisco, free
of charge, to protect Cisco CallManager against malicious applications.
4. C
Password expiration policies should be disabled on all service accounts. If the service account
password were to expire, it could prevent a variety of CallManager services from operating
correctly.
5. C
The IIS service comprises 80 percent of all attacks on Windows servers. Because of this, it is
a best practice to stop the IIS service on all Cisco CallManager Subscribers.
6. C
The Cisco CallManager server is not designed to handle nonapproved third-party utilities.
Chapter 21 815
7. A
The most automated process for installing security updates and hot fixes to the CallManager
server is e-mail subscription to the Cisco CallManager Notification Tool. The updates must
then be manually downloaded and installed.
8. A and C
The first of these scripts focuses on securing the foundation Windows 2000 operating system
with security settings recommended for Cisco CallManager servers only. The second of these
scripts focuses on securing the SQL communication on the server.
9. D
There is not even a product called WebX SecureServer.
10. A and B
Only virus protection software can prevent a server against virus infection. In addition, only
the managed CSA version provides firewall functionality. The standalone CSA version allows
all inbound network connections.
Chapter 21
1. C
The default CallManager installation without MLA uses the Windows 2000 Administrator
account for both CallManager Administration and server administration. A sniffed username
and password could mean the demise of both the CallManager Administration interface and
the underlying Windows operating system.
2. B
The URL to access the Cisco CallManager Administration interface remains the same. The
only difference is using the HTTPS protocol rather than just HTTP.
3. C
There is no read-write access in Cisco MLA. This setting would be the same as full access.
4. D
After you enable MLA and restart the World Wide Web Service in Windows 2000, a new
account “CCMAdministrator” is stored in the Windows 2000 Registry for full CallManager
Administration access.
5. A
There is no Standard Device functional group. Rather, MLA splits the device functions into
the specific devices managed, such as Standard Phone or Standard Gateway.
6. B
There is no FullAccess group in the MLA user group defaults. The full access role is assigned
to the SuperUserGroup.
7. B
Self-signed certificates raise an automatic red flag in any client web browser. This is because
there is no trusted authority (a Certificate Authority) authenticating the web server.
8. A
The MLA user accounts are stored in an LDAP directory. By default, this is the DC Directory
installed with the Cisco CallManager; however, you can change this directory storage to
anther directory (such as Active Directory).
9. D
CallManager MLA stores the CCMAdminstrator account in the Windows Registry. This
allows a “back door” into the CallManager Administration interface if the LDAP directory is
unavailable.
10. A and C
By default, MLA is disabled for a clean CallManager install; however, it will remain enabled
(with a new, random CCMAdministrator password) if upgrading from previous MLA (and
Cisco CallManager) version that was enabled.
Chapter 22
1. B and C
Users can exploit the Call Forward All feature (which they can configure through a web
interface) and transfer from voice-mail features to make toll calls using the corporate voice
network.
2. A
To restrict external call transfers (which block OffNet-to-OffNet transfers), route patterns and
gateways should be classified as OnNet or OffNet. Route patterns are typically used to restrict
outbound calls, whereas gateways are used to restrict incoming calls.
3. A and D
By default, the ad hoc conference restrictions are set to Never, which opens the possibility of
toll fraud. You can choose to set the restrictions to When Conference Creator Drops Out or
When No OnNet Parties Remain in the Conference to restrict this toll fraud method.
Chapter 23 817
4. A
By default, the ad hoc conference restrictions are set to Never, which opens the possibility of
toll fraud. You can choose to set the restrictions to When Conference Creator Drops Out or
When No OnNet Parties Remain in the Conference to restrict this toll fraud method.
5. C
Forced Authorization Codes (FACs) allow you to restrict call routing to destinations based on
user codes. This is not only useful to prevent toll fraud, but also to determine which users or
departments are making toll calls for billing purposes.
6. B
Commonly exploited countries have special area codes that look like area codes of the United
States. Not all international numbers begin with the 011 identifier (in North America).
Because of this, you should be especially careful to restrict international area codes using
route patterns.
7. A
The most automated process for installing security updates and hot fixes to the CallManager
server is e-mail subscription to the Cisco CallManager Notification Tool. The updates must
then be manually downloaded and installed.
8. B
The correct configuration of time-of-day routing is as follows: create time periods, add the
time periods to a time schedule, assign the time schedule to a partition, and assign the partition
to a calling search space.
9. D
Client Matter Codes (CMCs) are typically used to track calls to a specific department or user.
However, they do not give you the varying authorization levels of the FAC feature.
10. A, D, and E
By classifying the route patterns, trunks, and gateways as OnNet or OffNet, you can
effectively curb OffNet-to-OffNet transfer toll fraud.
Chapter 23
1. A
The IP phone contains key information about the IP addresses of the Cisco CallManager,
network gateway, TFTP server, and DNS servers. Obtaining this information allows a hacker
to map out the location of key network resources.
2. A
Only Cisco’s flagship phones, the 7940, 7960, and 7970, support configuration file
authentication. The 7920 wireless handset does not support this feature.
3. B
You can find all the security settings for an IP phone under the Phone Configuration window
in the Cisco CallManager administration utility.
4. A
To access the built-in web server of an IP phone, simply access the URL http://IP-Phone’s-IP-
address. There are no security options; you will be directed to a page giving all the
information about the IP phone’s network settings.
5. B
Gratuitous ARP attackers always operate from the local network. This is necessary due to the
nature of ARP packets.
6. C
Even with authentication and encryption features enabled, SCCP maintains its signaling role.
There are no currently known attacks against Skinny signaling, so there was no reason to
secure it further.
7. D
An IP phone can provide all the preceding information with the exception of the intranet
server address.
8. D
Cisco CallManager does not allow you to disable the PC port of the 7912 IP Phone. Filling
the PC port of the phone with glue (or some similarly drastic step), or downgrading to the
7905 IP Phone is the best step for now.
9. B
Secure Sockets Layer (SSL) was the predecessor to Transport Layer Security (TLS). SSL was
primarily applied to HTTPS connections, whereas TLS became more universally used.
10. A
Since Cisco CallManager 3.3(3), the signed firmware validation feature is already enabled.
This prevents hackers from constructing their own, rogue firmware images for the Cisco IP
Phones.
Chapter 24 819
Chapter 24
1. E and F
Cryptographic services include functions for authenticity, confidentiality, integrity, and
nonrepudiation.
2. A and F
Because of its speed, symmetric encryption is a good choice for real-time encryption of bulk
data. This speed is achieved because the encryption key is the same as the decryption key.
3. D and F
Asymmetric encryption is very powerful because it provides a public key (capable of handling
encryption and decryption of data) and a private key (also capable of handling encryption and
decryption of data). Asymmetric encryption is often used to create signatures and for key
exchange only because of the high overhead associated with the algorithm.
4. B and F
If a hacker obtains a hash of some data, the only method they can use to reverse engineer the
data is to use a brute force attack, which is computationally difficult. Hash does work well for
ensuring data does not change accidentally; however, it does not protect against man-in-the-
middle attacks. Because of this, hashing is often combined with an encryption algorithm, such
as AES. AES can use MD5 or SHA-1 for hashing data.
5. F
Only F does not apply to digital signatures. Digital signatures are created by encrypting the
result of a hashing process using a private key.
6. D
SHA-1 is the modern standard for creating a 160-bit hash. MD5 is no longer recommended
because it can run into problems with duplicate hashes on large amounts of data.
7. A
Nonrepudiation methods prove to others that a certain source sent some data. This is very
similar to authentication; however, authentication can only prove to YOU that a certain device
sent the data rather than proving to OTHERS that a certain device sent the data.
8. B
Rivest, Shamir, and Adleman (RSA) is the only asymmetric algorithm in this list.
9. A
Asymmetric algorithms use two keys: one public and one private. The public key can be used
for encryption and decryption of data and is sent to any requesting host. The private key can
be used for encryption and decryption of data and is kept strictly for the sending host.
10. D
With current mathematical algorithms, it is feasibly impossible to generate a private key from
a public key.
Chapter 25
1. D
Symmetric keys provide the fast encryption standards needed by today’s applications;
however, these keys need a method for exchange over an unsecured network. PKI provides a
solution to this challenge.
2. A and F
Because of its speed, symmetric encryption is a good choice for real-time encryption of bulk
data. This speed is achieved because the encryption key is the same as the decryption key.
3. D and F
The trusted introducer and its clients must trust the root of a system. The root guarantees the
identity of the trusted introducer. Only the trusted introducer can guarantee the authenticity
of any member of the system.
4. D
A certificate includes the identity of the issuer of the certificate, the identity of the owner of
the certificate, and the public key of the owner.
5. C and F
Securing enrollment through a PKI can be a sticky situation. The best method is to perform
the enrollment over a trusted network (or significantly secured public network). Otherwise,
you must manually perform mutual out-of-band authentication between the PKI user and CA.
6. D
Certificate revocation is needed whenever the private key is not trustworthy anymore. This
can occur through a loss of the private key (from a system rebuild or replacement), or a
malicious compromise of the private key (from an intruder).
7. F
The web server certificate is used to authenticate the server to the client and to encrypt the
symmetric session keys used for the authentication and encryption of the data stream.
Chapter 26 821
8. B
The Diffie-Hellman algorithm is commonly used as an automated method to securely
exchange symmetric keys over a public network.
9. A
Asymmetric algorithms use two keys: one public and one private. The public key can be used
for encryption and decryption of data and is sent to any requesting host. The private key can
be used for encryption and decryption of data and is kept strictly for the sending host.
10. A
Certificates are not secret information and do not need to be encrypted in any way. The idea
is not to hide anything but to ensure the authenticity and integrity of the information contained
in the certificate.
Chapter 26
1. C and E
Integrity and loss of control are typically terms to describe one’s personal life rather than IP
telephony security.
2. B and F
Secure signaling is accomplished through Transport Layer Security (TLS). This security is
crucial because CallManager sends the keys for SRTP (which secures the media) through
signaling to the IP phone.
3. D and F
The trusted introducer and its clients must trust the root of a system. The root guarantees the
identity of the trusted introducer. Only the trusted introducer can guarantee the authenticity
of any member of the system.
4. C and D
In Cisco IP telephony PKI infrastructures, the CAPF has a self-signed certificate because the
IP phones refer to this as the CA of the PKI. Only the Cisco IP Phone 7940, 7960, and 7970
(and subsequent) models can have LSCs because these are the only models that support
device security at this point.
5. C and F
Securing enrollment through a PKI can be a sticky situation. The best method is to perform
the enrollment over a trusted network (or significantly secured public network). Otherwise,
you must manually perform mutual out-of-band authentication between the PKI user and CA.
6. C
CAPF enrollment supports the use of authentication strings. This is known as the manual
enrollment method, which requires the administrator to visit each IP phone he wants to enroll
and enter the correct string from the CAPF.
7. B
The CTL client uses a smart token for key storage. This smart token exists on a USB key
attached to the server running the CTL client. The smart token never leaves the key, but,
rather, acts as a separate authentication engine to validate the CTL.
8. D
TLS allows both the server and the IP phone to authenticate each other through a signed
certificate. This also allows them to authenticate the signaling message to ensure they came
from the correct source.
9. B and D
Certificates are only exchanged between the Cisco CallManager server and the IP phone. The
IP phones themselves do not exchange certificates directly. Likewise, the encrypted
transmission of SRTP session keys occurs between the IP phones and the Cisco CallManager
rather than between the IP phones.
10. E
The most accurate list of tasks is to enable services, set cluster to mixed mode, create a signed
CTL, deploy certificates to the IP phones, and set the device security mode.
Chapter 27
1. E
To configure a Cisco CallManager cluster for security, first enable services, then use the CTL
Client to set cluster to mixed mode and create a signed CTL. Finally, deploy the certificates
to the IP Phones and set the device security mode through the CallManager Administration.
2. A and C
You must enable both the Cisco CAPF and CTL Provider when configuring a Cisco
CallManager cluster for security.
3. A
At a minimum, the Cisco CTL Client requires the Windows 2000 operating system, and at
least one USB port, Smart Card service enabled. It does also function on the Windows XP
operating system.
Chapter 28 823
4. A
Upgrading an LSC on a Cisco IP Phone can be done through the CallManager Administration
utility without updating the CTL using the Cisco CTL Client.
5. A and F
These answers show the valid security configuration options of the Cisco IP Phone through
the CallManager Administration.
6. C and D
When a phone configured for authentication calls a nonsecure phone, the resulting call will
negotiate to the lowest-common-denominator, which is nonsecure. Likewise, when a phone
configured for encryption calls a phone configured for authentication, the resulting call will
negotiate to the lowest-common-denominator, which is authentication.
7. C and D
You can generate a query using the device name through the Find and List Phones window.
The certificate lifetime is not a searchable field.
8. A and E
You can query based on both the authentication mode and device security mode from the Find
and List Phones window.
9. A
The Cisco 7920 Wireless IP Phone does not support the phone security features.
10. B, C, and D
All of these communication types cannot be encrypted using phone security. In addition, calls
streaming music on hold (MOH) cannot be secured.
Chapter 28
1. B
The provisioned video bandwidth amounts always include room for the audio payload. The
payload for G.711 uses 64 kbps of bandwidth, leaving 320 kbps for the video payload.
2. A
Cisco recommends adding 20 percent to the requested video bandwidth amount to provide a
safe buffer for overhead. 256 kbps * 1.20 = 307.2 kbps reservation.
3. A
H.263 is the lowest-common-denominator video codec between an SCCP and H.323 device.
If two SCCP devices communicate, they will attempt to use the newer (and more efficient)
H.264 codec.
4. B
When using the nongatekeeper-controlled intercluster trunks, the only choice you will have
for call admission control is the CallManager Location feature. By placing the local IP Phones
and intercluster trunk in different locations, you can control how much bandwidth
CallManager allows over the trunk.
5. B
Within a cluster, CallManager locations control the WAN bandwidth used. This is typically
done in a centralized call-processing environment.
6. B
The H.320 standard is used to stream video over an ISDN network. To communicate with this
standard, you must employ a video gateway.
7. D
Videoconferencing requires you use a multipoint control unit (MCU) to mix the signals.
SCCP clients can handle a video call (two users), but cannot handle a conference (three users
or more).
8. C
Far-end camera control (FECC) is an H.323 capability allowing you to remotely control a
variety of settings on a video camera.
9. D
The Cisco proprietary wideband codec offers impeccable quality, but is considered a “LAN-
only” protocol due to the whopping 7 Mbps it consumes.
10. C
One of the major features supported by the Cisco CallManager 4.0 release was video (along
with the new IP telephony security structure).
Chapter 29
1. C
The icon of a video camera on the Cisco IP Phone status line means that the Cisco IP Phone
is configured to support video. This occurs when the administrator enables video capabilities
in the CallManager Administration utility.
Chapter 30 825
2. B
Cisco requires you to plug the PC with Cisco VT Advantage client software into the access
port of the IP phone. This allows the IP phone to recognize the client PC (and installed
software) and enable video capabilities for incoming and outgoing calls.
3. A and C
The Cisco Audio Session Tunnel is a Cisco proprietary protocol used to signal between the
VT Advantage software and IP phone. RTP is used to stream video.
4. E, F, and G
Only the Cisco 7940, 7960, and 7970 IP Phones (and later) can support video.
5. B and C
Both the PC port and Retry Video Call as Audio settings are enabled by default. You must
enable the video capabilities of the IP phone to support VT Advantage.
6. D
The gatekeeper always requests double the amount of bandwidth required by the codec to
allow sufficient room for overhead.
7. A
The only camera hardware supported by the VT Advantage software is the Cisco VT Camera.
8. D
The Cisco proprietary wideband codec offers impeccable quality, but is considered a “LAN-
only” protocol due to the whopping 7 Mbps it consumes.
9. D
The VT Advantage software can stream up to 30 fps. This level produces extremely smooth
(TV quality) video streams.
Chapter 30
1. A and E
Microsoft SQL Server 2000 Enterprise Manager allows you to manipulate the SQL database
in many ways. It can be a dangerous tool because it can corrupt the CallManager database
structure quite easily. Cisco recommends using the DBLHelper utility, which can quickly
verify the status of the database replication and make repairs, if necessary.
2. D
Cisco CallManager Serviceability provides services to monitor alarms, generate CARs, and
collect and analyze traces. All other statements are false.
3. A and C
The only services that can be managed through the Control Center are those that relate
directly to the Cisco CallManager. In this case, the Cisco CallManager and TFTP services are
valid choices. There is no Cisco DHCP service, and the Extension Mobility function is
controlled as a TomCat web service. Cisco WebDialer is a separate application install.
4. C and E
The Cisco CallManager services can be activated and deactivated through either the Windows
2000 Services MMC or in Cisco CallManager Service Activation. Cisco recommends the
latter of these.
5. A
The DBLHelper application should only be run directly on the publisher server because this
server holds the only writable copy of the SQL database.
6. D
Because the publisher holds the only writable copy of the SQL database, all configuration
changes are prevented until the publisher server is restored.
7. C
CDRs are the “exception to the rule” of modifying the database while the publisher server is
down. In actuality, the SQL database is not modified, rather, the subscribers will write CDRs
to their SQL transaction file while the publisher is down and replicate the updates back to the
publisher after connectivity is restored.
8. A and C
Only the publisher server will have a Replication Monitor (which has an image that looks like
a heartbeat monitor) utility and will have a Publications folder under the database you are
examining.
9. B
DBLHelper provides an extremely simple interface that holds little risk for damaging the SQL
Server 2000 database structure when repairing replication issues. SQL Server 2000 Enterprise
Manager should only be used if DBLHelper is unable to repair the replication.
10. C
A green smiley face means replication is working okay, whereas a red sad face indicates a
replication failure.
Chapter 31 827
Chapter 31
1. B and D
The Microsoft Event Viewer is a Windows 2000 utility that can assist administrators in
troubleshooting Cisco CallManager systems. The Microsoft Event Viewer stores system
errors and warnings as they occur on the CallManager system, which allows for a historical
viewing of any issues that might have occurred on the CallManager.
2. A, B, and C
Microsoft Performance Monitor does nothing but monitor, monitor, and more monitoring.
You can use it to monitor Windows 2000 counters and CallManager counters alike. All events
are monitored in real time through a graph, histogram, or report view.
3. A and D
The Real-Time Monitoring Tool (RTMT) allows you to view the performance of a specific
CallManager server, and is geared for CallManager-specific counters. It also offers the ability
to send e-mail alerts when the CallManager exceeds specific thresholds.
4. B and E
The RTMT allows you to save multiple configuration profiles, which allows you to open a
predefined set of counters to quickly monitor specific areas of the CallManager cluster. RTMT
identifies these profiles by their name and description.
5. A, C, and E
Cisco designed the RTMT window to allow easy navigation through the various performance
counters of a Cisco CallManager server. This includes the support of multiple tabs to allow
many different elements to be viewed at one time. The RTMT does provide performance
monitoring similar to the Microsoft Performance Monitor; however, it has many distinct
differences that tune it to be more effective in monitoring CallManager servers. The Alert
Central feature of RTMT offers the ability to send e-mail alerts when the CallManager
exceeds specific thresholds.
6. B
The Application logs of Event Viewer contain messages specific to Cisco CallManager. The
System logs contain messages specific to the underlying Windows 2000 operating system.
7. A, E, and F
The Histogram view provides a bar chart giving instantaneous performance levels for your
selected counters. The Graph view provides a line chart showing a history of your selected
counters over time. The Report view provides a numerical table with specific counter levels.
8. A, C, and D
The Microsoft Performance Monitor, Cisco RTMT, and Windows Task Manager all allow you
to see processor and memory utilization levels for a Cisco CallManager server.
Chapter 32
1. A and C
The CallManager Serviceability Alarm menu has only two options: Alarm and Alarm
Definitions. The first allows the configuration of alarms on individual servers and services.
The latter allows you to get a full definition of each alarm (and add your own custom notes to
the alarm, if necessary).
2. C
The only true statement is that more than one destination can be used to write alarm logs in
parallel, and each of them can use its own alarm level. Cisco CallManager relies on the
reporting destination to have the proper functionality if e-mail or any other notifications are
necessary.
3. D
You can configure alarms for every Cisco CallManager service through the Serviceability
pages; however, you must enable Java application alarms through the Windows Registry.
4. A, D, and E
SDI traces log services and run-time events, whereas SDL traces log call-processing
information. Both of these traces can be written to plaintext and XML files based on your
Cisco CallManager Serviceability configuration.
5. C
The Trace Collection tool alleviates much of the trace analysis function from the Cisco
CallManager server. This helps save resources and makes for easier, offline analysis of the
trace files. The Trace Collection tool downloads and compresses trace files from Cisco
CallManager systems to a computer. You can then use a text editor or the Bulk Trace Analysis
tool to analyze the files.
6. C and D
The Trace Analysis tool only has the ability to analyze trace files that are less than 2 MB in
size and are in XML format.
7. C
You must access the Q.931 Translator through the web interface of the CallManager
Serviceability pages. The Voice Log Translator can be used to analyze log files without access
to the Cisco CallManager system.
Chapter 33 829
8. B and D
The SDL trace logging focuses on call logs between IP telephony devices. Naturally, only the
Cisco CallManager and CTIManager services will support these types of log files.
9. C
The CallManager alarm levels use the same mappings as the syslog messages. An alarm level
5 is assigned a name of “Critical.” The only levels above this are Alert (level 6) and
Emergency (level 7).
10. A, B, and D
Cisco CallManager does not support sending alarms directly to SMTP servers (e-mail
addresses). It relies on the Real-Time Monitoring Tool (RTMT) to provide this functionality.
Cisco also assumes that you will configure the Microsoft Event Log or syslog server to use
an SMTP alerting function, if necessary.
Chapter 33
1. A, C, and E
When using CAR, PDF reports are limited to 5000 records and CSV reports are limited to
20,000 records. CAR does have to be installed on the publisher server to work correctly.
2. D
Based on these answers, the only thing CMRs and CDRs have in common is that they are
related to each other. CMRs store QoS parameters for the call, whereas CDRs store call
details. They are stored permanently in the SQL database and only temporarily in flat files.
3. C
By default, you must log in to CAR using a username and password of admin.
4. D
There are only three types of reports you can reference in CAR: User reports, System reports,
and Device reports.
5. A
Actually, the ONLY thing you are able to do after initially authenticating to CAR is to grant
administrative access rights to one of the users in the CallManager LDAP user database. All
other options are restricted until this step is completed.
6. C
The default restrictions load CDR data only from midnight to 5:00 a.m. This keeps the CDR
replication and processing from interfering with normal IP telephony network operations.
7. C
The CAR and CDR database alerts the CAR administrator by default when 80 percent of the
maximum number of rows is reached. At this point, the administrator should consider
manually purging the database.
8. A
The CAR tool allows you to log calls based on the NANP. This dial plan can be modified, and
includes an “On Net” classification by default.
9. C and D
CAR allows report generation in Adobe PDF or Comma Separated Values (CSV) files.
10. D
Even if the publisher is online, the CDRs are stored on the subscriber server in flat files. These
files are replicated to the publisher database during a scheduled interval.
Chapter 34
1. B
The dependency record capability was developed as a response to quite a bit of administrative
frustration in early CallManager versions. CallManager would restrict you from deleting an
item because something else was using it, but it would never tell you what that something else
was. Dependency records helps track which devices are associated with each other in
CallManager.
2. C
The CCMAdministrator, CCMSysUser, IPMASysUser, and Directory Manager passwords
can all be changed using the Password Changer tool.
3. A
Cisco DNA can be used to analyze and test dial plans in an IP telephony environment.
4. D
This question uses tricky wording. The QRT reports are actually created when the user presses
the QRT button. They are displayed when the administrator uses the QRT Viewer.
5. C
SNMP Version 3 (which is still awaiting standardization) adds support for both authentication
and encryption.
Chapter 34 831
6. B
Dependency records are disabled by default on the Cisco CallManager because they can lead
to high CPU utilization. To minimize the effect, you can enable them during the off-peak
hours.
7. B
The Password Changer tool only functions if you are using Cisco MLA, which prevents the
CallManager server from sharing the user database with Windows. The passwords of the
specified user accounts are changed in the MLA LDAP directory of all servers in the cluster.
8. C
The Password Changer tool can be accessed from the Run prompt within Windows.
9. C
The CallManager alarm levels use the same mappings as the syslog messages. An alarm level
5 is assigned a name of “Critical.” The only levels above this are Alert (level 6) and
Emergency (level 7).
10. D
You can access the Dialed Number Analyzer by entering the URL manually or accessing the
added icon in the Start menu or on the Windows desktop after you have installed it on a
CallManager server.
Index
Numerics administrators, Windows 2000 Server,

password policies, 472–475
7985 IP Phones, IP video telephony, 620 advanced route plans, CallManager, 209
DDIs (Discard Digit Instructions), 217–219
reports, 229
A route filters, 209–217
AAR (Automated Alternate Routing), 275 transformation masks, 219–225
configuring, 282–288 translation patterns, 226–228
locations-based call admission control, 280– AES algorithm, 539
288 Alarm service, Serviceability tool
abbreviated dialing, configuring, 348 (CallManager), 679–680
ac users, CallManager Attendant Console, alarms, CallManager, 713–714
adding, 417–419 configuration, 714–719
access analog station gateways, 162 definition details, 717–719
access analog trunk gateways, 162 event levels, 714
access control lists (ACLs), 461 Alert Central, RTMT (Real-Time Monitoring
access gateways, configuring, 164–174 Tool), 707– 708
H.323 gateways, 165–169 algorithms
MGCP gateways, 169–172 asymmetric cryptography, PKI (Public Key
Non-IOS MGCP gateways, 172–174 Infrastructure), 552– 553
accessing dependency records, CallManager, asymmetric encryption, 540–541
772–774 call-distribution algorithms, line groups,
aceess privileges, MLA (Multilevel 237–238
Administration), assigning, 491–493 symmetric cryptography, PKI (Public Key
ACLs (access control lists), 461 Infrastructure), 551– 552
Actions menu (Attendant Console), 427 symmetric encryption, 538–540
ad hoc conferencing, 311 Always Route Member field (hunt group
dropping, 516–517 configuration), 422
Adleman, Leonard M., 541 analog gateways, 161–162
administration rights, CAR (Call Detail Analog Telephone Adaptor (ATA), 60
Record {CDR} Analysis and Reporting) ANI (Automatic Number Identification), 219
tool, configuring, 743–744 announcements, annunciators, 319
Administrative Reporting Tool (ART). See annunciation, 309
CAR tool annunciators, 319–320
Administrator (CallManager) anouncements, 319
remote threats to, 483 configuring, 320–321
MLA (Multilevel Administration), 487–499 antivirus protection, Windows 2000 Server
settings, disabling, 526–530 Operating System, 467–468
API (application programming interface), CTIManager, activating, 423–424
CallManager, 7 directory, 412–413
Application menu, Serviceability tool, Directory pane, 430
CallManager, 683 hunt groups, 413–414
applications layer (Unified Communications), configuring, 420–422
7 interface, 427
Architecture for Voice, Video, and Integrated menus, 427
Data (AVVID). See AVVID (Architecture Parked Calls pane, 430–431
for Voice, Video, and Integrated Data) pilot points, 413–414
architecture, IPMA (IP Manager Assistant), configuring, 419–420
439–440 Speed Dials pane, 429–430
area codes, commonly exploited area codes, TCD (Telephony Call Dispatcher), 411–412
blocking, 507–509 activating, 423–424
ART (Administrative Reporting Tool). See terminology, 411
CAR tool users, adding, 416–417
assigning MRGLs (Media Resource Group audio, video call flow, 622
Lists), 339 audio codecs, IP Phones, support for, 67
Assistant Console, IPMA (IP Manager Audio Session Tunnel, 650
Assistant), installing, 450–452 audio source files, MoH (music on hold)
assistants, IPMA (IP Manager Assistant), configuring, 330–331
configuring, 444–448 creating, 326–327
asymmetric cryptography, PKI (Public Key audio source IDs, MoH (music on hold),
Infrastructure), 552–553 assigning, 334
asymmetric encryption, 540–541 audio translator, MoH (music on hold),
ATA (Analog Telephone Adaptor), 60 configuring, 327–328
attacks authentication, 535–538, 573, 597
DoS (denial of service) attacks, 573 CallManager, 10
identity spoofing, 573 certificates
man-in-the-middle enrollment attacks, PKI LSCs (locally significant certificates),
(Public Key Infrastructure), 560 602–607
Attendant Console (CallManager), 8, 409–410 storage, 585
ac users, adding, 417–419 configuring, 597–599
Broadcast Calls pane, 431 CTL client, 600–602
Call Control window, 428–429 digital signatures, 544–546
call queuing, 415 hash functions, 541–544
call routing, 414–415 integrity, 585–588
client, configuring and installing, 425–426 IP Phone, enabling, 530
configuring, 416–426 IP telephony threats, 573
Configuration Tool, 424–425 protecting against, 574–576
834 authentication
keys, storage, 585 devices

phone configuration files, CallManager, choosing, 123–124
574, 576 specifics, 124–125
phone images, CallManager, 574–576 devices for, 119–120
phone-to-server authentication, 587–588 installation, 121–123
PKI, enrollment, 583–585 IP Phones
PKI topologies, 576 adding, 123–135
CAPF (CallManager Certificate inserting, 132–134
Authority Proxy Funtion), 577 updating, 134–135
CTL client, 580–583 validating, 131–132
LSCs (locally significant certificates), records for, 119–120
577–579 TAPS (Tool for Auto-Registered Phones
MICs (manufacturing installed Support), 136–138
certificates), 577–578 templates, 438
multiple PKI topologies, 579 configuring, 126–127
self-signed certificates, 576–577 blocking commonly exploited area codes,
server-to-phone authentication, 586–587 507–509
SHA-1 keys, exchanges, 588 boot process, IP Phones, 64–67
TLS sessions, certificate exchanges, 586 braodcast call-distribution algorithm, 238
authentication strings, phone certificates, Broadcast Calls pane (Attendant Console),
issuing, 605–607 431
Authorized Self-Study Guide: Cisco Voice over Bulk Administration Tool (BAT). See BAT
IP (CVOICE), 161, 164 (Bulk Administration Tool)
Auto Answer, configuring, 349 Bulk Trace Analysis tool (CallManager), 730–
Auto Call Pickup, configuring, 362–363 732
Automated Alternate Routing (AAR). See button templates, IP Phones, 84–86
AAR (Automated Alternate Routing)
Automatic Number Identification (ANI), 219
automatic registrations, IP Phones, 90–92
Auxiliary VLANs, Catalyst switches, support
C
CA (Certificate Authority)
for, 145 CAPF, acting as, 583–584
availability, softkeys, modifying, 353–354 PKI (Public Key Infrastructure), 556–559
AVVID (Architecture for Voice, Video, and web servers, 564–567
Integrated Data), 5-6 CAC (call admission control)
gatekeeper-based call admission control,
288–296
B implementing, 275–276
bandwidth, IP video telephony locations-based call admission control,
calculations, 630–631 276–288
call admission control, 631–632 AAR (Automated Alternate Routing),
managing, 628–632 280–288
requirements, 629–630 configuring, 278–279
barge tones, 319 calculating IP video telephony badwidth, 630–
configuring, 365–366 631
BAT (Bulk Administration Tool), 119 call activity, RTMT (Real-Time Monitoring
components, 120 Tool), monitoring, 706–707
CSV files, creating, 128–131
CallManager 835
call admission control call transfers, external transfers, restricting,

between clusters, 637–643 511–516
gatekeepers, 641–643 call waiting, 347
location configuration, 635–637 called-party transformations, 221–222
region configuration, 634–635 configuring, 223–224
Retry Video Call as Audio setting, caller ID, 219, 396-397
638–640 Calling Line ID Presentation field
CallManager, 631–632 (CallManager), 396–397
within clusters, 632–640 calling privileges
call admission control (CAC). See CAC (call calling search spaces, 254–255
admission control) configuring, 258–259
Call Back, configuring, 364 gateways, 256
Call Control window (Attendant Console), intercluster trunks, 256
428–429 CoS (Class of Service), 253–254
Call Coverage, 235 enforcement of, 253
Call Detail Record (CDR) Analysis and partitions, 254–255
Reporting (CAR) tool. See CAR (Call configuring, 256–258
Detail Record {CDR} Analysis and time-of-day routing, 260
Reporting) tool configuring, 262–268
call detail records (CDRs), 673–674 time periods, 260–265
Call Dispatcher, 411–412 time schedules, 261, 265–268
activating, 423–424 usage scenario, 268–270
Call Display Restrictions, 395–396 user effects, 262
Calling Line ID Presentation, 396–397 Calling Search Space field (pilot point
Connected Line ID Presentation, 397 configuration), 420
Ignore Presentation Indicators, 397–398 calling search spaces, 253–255
call distribution, 235–236 Call Forward toll fraud, preventing, 505–
hunting and forwarding, 240–241 507
line groups, 236–237 configuring, 258–259
algorithms, 237–238 gateways, 256
configuring, 242–243 voice mail toll fraud, preventing, 505–507
hunt options, 238–240 calling-party transformations, 220–221
call divert targets, IPMA (IP Manager configuring, 223–224
Assistant), configuring, 449–450 CallManager, 5, 7–8, 35
call flow, video calls, 622–623 ad hoc conferencing, dropping, 516–517
call forwarding Administrator
CallManager, 107–108 disabling settings, 526–530
configuring, 349–351 remote threats to, 483
Call Join, configuring, 358 advanced route plans, 209
Call Park, configuring, 359–360 DDIs (Discard Digit Instructions),
Call Pickup, configuring, 360–363 217–219
call processing (CallManager), 8-9 reports, 229
call queuing, CallManager Attendant route filters, 209–217
Console, 415 transformation masks, 219–225
call routing translation patterns, 226–228
CallManager Attendant Console, 414–415 alarms, 713–714
external call routing, 187–188 configuring, 714–719
VT (Video Telephony) Advantage, 654 definition details, 717–719
event levels, 714
836 CallManager
API (application programming interface), 7 call-distribution components, 235–236

Attendant Console, 8, 409–410 hunting and forwarding, 240–241
adding ac users, 417–419 line groups, 236–243
adding users, 416–417 Calling Line ID Presentation field, 396–397
Broadcast Calls pane, 431 calling search spaces, 254–255
Call Control window, 428–429 configuring, 258–259
call queuing, 415 gateways, 256
call routing, 414–415 call-processing deployment models, 23
client, 425–426 clustering over the IP WAN, 30–31
Configuration Tool, 424–425 multisite deployment model, 25–27
configuring, 416–426 multisite deployment model with
CTIManager, 423–424 distributed call processing, 27–30
directory, 412–413 single-site deployment model, 23–25
Directory pane, 430 CAPF (CallManager Certificate Authority
hunt groups, 413–414, 420–422 Proxy Funtion), PKI topologies, 577
interface, 427 CAR (Call Detail Record {CDR} Analysis
menus, 427 and Reporting) tool, 690, 737
Parked Calls pane, 430–431 CDRs (call detail records), 737–739
pilot points, 413–414, 419–420 CMRs (Call Management Records),
Speed Dials pane, 429–430 737–739
TCD, 423–424 configuration, 740–748
TCD (Telephony Call Dispatcher), device reports, 740
411–412 report scheduling, 748–752
terminology, 411 system database configuration, 752–
authentication, 10 756
configuring, 597–599 system reports, 740
CTL client, 600–602 user levels, 740
Barge, configuring, 365–366 user report configuration, 756–758
BAT (Bulk Administration Tool) user reports, 740
components, 120 users, 739
installation, 121–123 Cisco Call Back, configuring, 364
IP Phones, 123–135 clean installation process, 35
TAPS (Tool for Auto-Registered installation configuration data, 36–38
Phones Support), 136–138 installation disks, 35–36
CAC (call admission control), 275–276 postinstallation procedures, 40–41
gatekeeper-based call admission service activation, 41–45
control, 288–296 clusters, 17
locations-based call admission intracluster run-time data, 18–19
control, 276–288 limits, 22
call admission control, 637–638 redundancy designs, 19–22
clusters, 632–643 SQL database cluster, 17–18
location configuration, 635–637 CMC (Client Matter Codes), 511
region configuration, 634–635 components, 10
Retry Video Call as Audio setting, Connected Line ID Presentation field, 397
638–640 CTL client, installing, 600–602
Call Join, configuring, 358 Customer Directory Plugin, 10
Call Park, configuring, 359–360 database replication, 38
Call Pickup, configuring, 360–363 DBLHelper, 689
call processing, 8–9 DC Directory, 10
CallManager 837
dependency records, 771 entry-level IP Phones, 54–56

accessing, 772–774 features, 61, 64
enabling, 771–772 interaction, 8–9
device control, 8 intracluster communication, 73–87
dial plan adminstration, 8 manual directory number
Dialed Number Analyzer, 690 configuration, 87–90
Direct Transfer, configuring, 357 midrange IP Phones, 56–59
directory services, 8 registering for, 67
DNA (Dial Number Analyzer), 776–779 startup process, 64–67
installing, 777–778 template buttons, 84–86
encryption upper-end IP Phones, 56–59
configuring, 597–599 IP telephony threats, protecting against,
CTL client, 600–602 574–576
Event Viewer, 689 IP telephony users, 101
Extension Mobility, 377–378 adding, 102–104
configuring, 378–389 call forwarding, 107–108
default device profiles, 382–383 configuration, 102–114
service parameters, 381–382 device pool configuration, 104–105
user device profiles, 383–386 Fast Dial service, 111–112
external call routing, 187–188 IP Phone services, 110–111
external-to-external call transfers, blocking, locale settings, 113–114
512–516 logon options, 105–106
FAC (Forced Authorization Codes), 5 Message Waiting Lamp page, 112–113
10– 511 Personal Address Book service,
functions, 8 111– 112
gateways speed dialing, 108–110
call survivability, 163 user database, 101–102
configuring, 164–174 User Options welcome screen, 106–
redundancy support, 162 107
Group Call Pickup, configuring, 361 IP video telephony, 619
group configuration, 79–81 bandwidth calculations, 630–631
H.323 gateways, configuring, 165–169 bandwidth management, 628–632
HTTPS (Secure HTTP), 483 bandwidth requirements, 629–630
securing through, 483–487 call admission control, 631–632
hunt lists, configuring, 243–245 codec support, 623–624
hunt pilots, configuring, 245–247 H.323 endpoints, 625–628
Ignore Presentation Indicators check box, intercluster trunks, 640
397–398 SCCP endpoints, 625–627
Immediate Divert, configuring, 358–359 video calls, 621–622
IP Phone Services video protocols, 624–628
configuring, 368–372 IP Voice Streaming application, activating,
displays, 372 310
IP Phones, 53 IPMA (IP Manager Assistant), activating,
automatic registration, 90–92 441
codec support, 67 LSCs (locally significant certificates),
configuring, 73–92 602– 607
device pool configuration, 75–87 management tools, 688–690
device pool design, 92–95 MCID (Malicious Call Identification),
DNS reliance removal, 73–75 configuring, 399–403
838 CallManager
MGCP gateways, configuring, 169–170, CRL (certificate revocation lists), 557

172 enrollment, 559–561
MLA (Multilevel Administration), 487–488 entities, 556–559
access privileges, 491–493 key storage, 562–563
enabling, 488–489 revocation methods, 561–562
functional groups, 489–499 Smart Cards, 563
user groups, 490–499 smart tokens, 563
MoH Audio Translator, activating, 310 SSL, 563–564
most recent version, 45 symmetric cryptograpy, 551–552
MRM (Media Resource Manager), 309– TLS (Transport Layer Security), 563–
310, 335 564
annunciators, 319–321 web server certificate exchange, 564–
conference bridges, 310–316 567
media resource design, 335–336 precedence, 403–404
MoH (music on hold), 324–334 preemption, 403–404
MRGLs (Media Resource Group Privacy, configuring, 367–368
Lists), 338–341 programming interface to external
MRGs (Media Resource Groups), applications, 8
336–338 QRT (Quality Report Tool), 780–784
MTPs (Media Termination Points), logs, 782–785
316–318 softkey activation, 781–782
transcoders, 321–324 QRT Viewer, 690
multiple call per line appearances, region configuration, 82
configuring, 355–357 reimaging, 46
Non-IOS MGCP gateways, configuring, route plans, 187, 204–205
172–174 configuration, 189–205
Notification Tool, 462 digit analysis, 200
Other Group Call Pickup, configuring, 361 digit collection, 201–203
partitions, 254–255 route groups, 189–192
configuring, 256–258 route lists, 192–194
Password Changer tool, 690, 774–776 route patterns, 194–199
Performance Monitor, 689 RTMT client, 690
performance monitoring, 695 SCCP (Skinny Client Control Protocol), 8–9
Event Viewer, 697–698 SDK (Software Development Kit), 7
performance counters, 695–697 secure media transfers, 574–575
Performance Monitor, 699–702 secure signaling, 574–575
RTMT (Real-Time Monitoring Tool), servers, 11
702–708 Serviceability tool, 678, 687–688
phone configuration files, authentication, Alarm service, 679–680
574-576 Application menu, 683
phone feature administration, 8 Control Center, 684–686
phone images, authentication, 574–576 Help menu, 684
phone loads, verifying, 657–658 interface, 679
PKI (Public Key Infrastructure), 551 menus, 678
as trusted third-party protocol, 553– Service Activation tool, 686
556 Services MMC, 687
asymmetric cryptograpy, 552–553 Tools service, 681–682
CA (Certificate Authority), 556–559 Trace service, 680–681
certificates, 557–559 services, 42–44
calls 839
Services MMC, 689 upgrading, 45–47

signaling, 9 VT (Video Telephony) Advantage
signaling control, 8 call routing, 654
SIP (Session Initiation Protocol), 179 calls, 651–652
components, 179–181 client installation, 661–667
configuration, 181–183 components, 647–648
integration, 181–183 configuring, 653–654
proxy servers, 179 deployment tools, 655–657
redirect servers, 180 IP phone configuration, 657–661
registrar servers, 180 protocols, 649–650
user agents, 180 supported standards, 648–649
softkey template configuration, 83–84 video modes, 652–653
SQL (Structured Query Language) Windows 2000 operating system, 10
databases, 673 Windows 2000 Server Operating System
DBLHelper, 677–678 administrator password policies, 472–
management tools, 673–674 475
SQL Server 2000 Enterprise Manager, antivirus protection, 467–468
674–677 common exploits, 475–476
SQL (Structured Query Language) Server CSA (Cisco Security Agent), 468–472
2000, reliance upon, 10 hardening, 463–467
SQL Server 2000 Enterprise Manager, 689 hot fix policy, 461–462
static IP information, choosing, 37 security, 459
stopping manually, 19 security patches, 461–462
system administrators, password protection, security settings, 476–478
38 threats to, 459–461
time-of-day routing, 260, 509–510 CallManager Certificate Authority Proxy
configuring, 262–268 Function (CAPF). See CAPF (CallManager
time periods, 260–265 Certificate Authority Proxy Function)
time schedules, 261, 265–268 CallManager certificates, CTL client, 580
usage scenario, 268–270 CallProcess category, RTMT (Real-Time
user effects, 262 Monitoring Tool), 703
traces, 720 call-processing deployment models,
Bulk Trace Analysis tool, 730–732 CallManager servers, 23
configuring, 719–725 clustering over the IP WAN, 30–31
Dick Tracy tool, 732 multisite deployment model, 25–27
Ethereal tool, 732–733 multisite deployment model with distributed
Q.931 Translator, 732 call processing, 27–30
Trace Analysis tool, 725–728 single-site deployment model, 23–25
Trace Collection tool, 728–730 call-processing layer (Unified
Voice Log Translator, 732 Communications), 7
trunks, 174–175 calls
H.225 gatekeeper controlled trunks, video calls, 621
175 call flow, 622–623
intercluster gatekeeper-controlled CallManager, 621–622
trunks, 175 codec support, 623–624
intercluster nongatekeeper-controlled VT (Video Telephony) Advantage
trunks, 175 protocols, 651–652
intercluster trunks, 176–179 video modes, 652–653
SIP trunks, 176
840 CAPF (CallManager Certificate Authority Proxy Function)
CAPF (CallManager Certificate Authority voice VLANs, 152–154

Proxy Function), 43 wall power, 146
CA (Certificate Authority), acting as, 583– CatOS, dual VLANs, configuring, 153
584 CCMAdministrator account, password,
certificates, CTL client, 580 changing, 775
external CA (Certificate Authority), acting CCMSysUser account, password, changing,
as a proxy to, 584–585 775
phone certificates, issuing, 607 CDR inserts, CallManager, 43
PKI topologies, 577 CDRs (call detail records), 673–674
reports, generating, 609–611 CAR (Call Detail Record {CDR} Analysis
CAR (Call Detail Record {CDR} Analysis and and Reporting) tool, 737–739
Reporting) tool, 690, 737 Certificate Authority (CA). See CA
CDRs (call detail records), 737–739 (Certificate Authority)
CMRs (Call Management Records), certificates
737– 739 CAPF (CallManager Certificate Authority
configuration, 740 Proxy Funtion), PKI topologies, 577
administration rights, 743–744 CTL clint, 580-583
dial plan parameters, 744–746 CallManager certificates, 580
gateways, 746–747 CAPF certificates, 580
initial configuration, 741–742 MICs (manufacturing installed
mail parameters, 743–744 certificates), 581
system parameters, 742–743 TFTP server certificates, 580
system preferences, 747–748 HTTPS (Secure HTTP), 484–485
device reports, 740 self-signed certificates, 485–487
report scheduling, 748–752 LSCs (locally significant certificates),
system database, configuring, 752–756 602– 607
system reports, 740 PKI topologies, 577–579
user levels, 740 MICs (manufacturing installed certificates),
user reports, 740 PKI topologies, 577–578
configuring, 756–758 PKI (Public Key Infrastructure), 557
users, 739 self-signed certificates, 558–559
Catalyst switches, 145 web servers, 564–567
Auxiliary VLAN support, 145 X.509v3 certificates, 557–558
CoS (class of service), configuring, self-signed certificates, PKI topologies,
155–156 576–577
CoS marking, 146 storage, IP telephony, 585
data VLANs, 152–154 TLS sessions, exchanges, 586
dual VLANs, 153–154 CFA (Call Forward All) toll fraud, 504
inline power, 145 preventing, calling search spaces, 505–507
IP Phones, powering, 146–151 CHAP (Challenge Handshake Authentication
IP telephony, role in, 145–146 Protocol), 544
midspan power injection, 146 circular call-distribution algorithm, 238
PoE (Power over Ethernet), 146–148 Cisco upgrades, release of, 46
Catalyst EtherSwitch modules, 149 Cisco 7900 series IP Phones, introduction of, 5
Catalyst modular switching, 148 Cisco Call Back, configuring, 364
Catalyst stackable switching, 149 Cisco CallManager 4.1 Software Disk, 36
configuring, 150 Cisco IP Telephony Server Operating System
device detection, 147–148 Hardware Detection Disk, 35
verifying, 151
configuration 841
Cisco IP Telephony Server Operating System CMRs (Call Management Records), CAR
Installation and Recovery Disk, 36 (Call Detail Record {CDR} Analysis and
Cisco IP Telephony Solution Network Reporting) tool, 737–739
Reference Design, 25 Code Matter Codes (CMCs). See CMCs
Cisco IP Telephony Solution Reference (Code Matter Codes)
Network Design (SRND), 25, 314 codecs
Cisco IP Voice Media Streaming application, IP Phones, support for, 67
312 video codecs, CallManager, 623–624
Cisco QOS Exam Certification Guide, Second commands
Edition, 155 IOS gatekeepers, 293
Cisco Voice Gateways and Gatekeepers, 161, SRST, 299
164, 289 Comma-Separated Values (CSV), 737
CiscoWorks ITEM (IP Telephony commonly exploited area codes, blocking,
Environment Monitor), 763–764 507–509
Class of Service (CoS). See CoS (Class of components
Service) BAT (Bulk Administration Tool), 120
clean installation, CallManager, 35 SIP (Session Initiation Protocol), 179–181
installation configuration data, 36–38 conference bridges, 310–311
installation disks, 35–36 ad hoc conferencing, 311
postinstallation procedures, 40–41 Cisco IP Voice Media Streaming
service activation, 41–45 application, 312
client applications, 411 hardware, 311–315
client layer (Unified Communications), 7 configuring, 315–316
clients, VT (Video Telephony) Advantage, IP/VC-3500 Series Video Multipoint
installing on, 661–667 Conference Unit, 313
closest match routing, CallManager, 201–202 Meet-Me conferencing, 311
clusters NM-HD network modules, 313
call admission control, 632–643 NM-HD-2VE network modules, 313
gatekeepers, 641–643 NM-HDV network modules, 313
location configuration, 635–637 NM-HDV2 network modules, 313
region configuration, 634–635 WS-SVC-CMM conference bridge type,
Retry Video Call as Audio setting, 312
638–640 WS-X6608-E1 conference bridge type, 312
CallManager servers, 17 WS-X6608-T1 conference bridge type, 312
clustering over the IP WAN, 30–31 conferencing, 309
intracluster run-time data, 18–19 ad hoc conferencing, 311
limits, 22 dropping, 516–517
redundancy designs, 19–22 Meet-Me conferencing, 311
SQL database cluster, 17–18 configuration
intercluster trunks, CallManager, 640 AAR (Automated Alternate Routing),
intracluster communication, IP Phones, 282– 288
73–87 abbreviating dialing, 348
CMCs (Code Matter Codes), 389, 391–392 access gateways, 164–170, 172–174
configuring, 392–394 H.323 gateways, 165–169
creating, 392 MGCP gateways, 169–170, 172
route patterns, enabling, 394–395 Non-IOS MGCP gateways, 172–174
toll fraud, preventing, 511 annunciators, 320–321
authentication, 597–599
CTL client, 600–602
842 configuration
Auto Answer, 349 dual VLANs

Barge, 365–366 CatOS, 153
call admission control NativeOS, 153–154
location configuration, 635–637 encryption, 597–599
region configuration, 634–635 CTL client, 600–602
call forwarding, 349–351 FACs (Forced Authentication Codes),
Call Join, 358 392– 394
Call Park, 359–360 Group Call Pickup, 361
Call Pickup, 360–363 hunt groups, call-distribution components,
called-party transformations, 223–224 235–241
calling search spaces, 258–259 hunt lists, 243–245
calling-party transformations, 223–224 hunt pilots, 245–247
CallManager, 416-426 Immediate Divert, 358–359
adding ac users, 417–419 intercluster trunks, 176–179
adding users, 416–417 IP Phones, 64–67, 368-372
alarms, 714–719 device security mode, 607–608
call queuing, 415 for CallManager, 73–87
call routing, 414–415 intracluster communication, 73–87
client, 425–426 line groups, 242–243
Configuration Tool, 424–425 locations-based call admission control,
configuring, 416–426 278–279
CTIManager, 423–424 MCID (Malicious Call Identification),
Extension Mobility, 378–389 399– 403
hunt groups, 420–422 MoH (music on hold), 327–334
IP Phones, 87–92 audio source files, 330–331
IP telephony users, 102–114 audio source ID assignements, 334
locale settings, 113–114 audio translator, 327–328
pilot points, 419–420 fixed audio source, 332–333
route plans, 189–205 servicewide settings, 331–332
TCD (Telephony Call Dispatcher), MRGLs (Media Resource Group Lists),
423–424 338–339
traces, 719–725 MRGs (Media Resource Groups), 337–338
CAR (Call Detail Record {CDR} Analysis MTPs (Media Termination Points), 318
and Reporting) tool, 740 multiple calls per line appearances, 355–357
administration rights, 743–744 Other Group Call Pickup, 361
dial plan parameters, 744–746 partitions, 256–258
gateways, 746–747 PoE (Power over Ethernet), Catalyst
initial configuration, 741–742 switches, 150
mail parameters, 743–744 Privacy, 367–368
system database, 752–756 route filters, 212–214
system parameters, 742–743 shared-line support, IPMA (IP Manager
system preferences, 747–748 Assistant), 440– 452
user reports, 756–758 SIP (Session Initiation Protocol),
Cisco Call Back, 364 CallManager, 181–183
CMCs (Code Matter Codes), 392–394 SRST (Survivability Remote Site
conference bridge hardware, 315–316 Telephony)
CoS (class of service), Catalyst switches, references, 300–301
155–156 routers, 298–299
Direct Transfer, 357 time-of-day routing, 262–268
cyclic redundancy check (CRC) checksums 843
time periods, 263–265 asymmetric cryptography, 552–553

time schedules, 265–268 CA (Certificate Authority), 556–559
transcoders, 323–324 certificates, 557–559
translation patterns, 226–227 CRL (certificate revocation lists), 557
VT (Video Telephony) Advantage, 647 enrollment, 559–561
CallManager, 653–657 entities, 556–559
IP phones, 657–661 key storage, 562–563
configuration data, CallManager, clean revocation methods, 561–562
installations, 36–38 Smart Cards, 563
configuration files, phone configuration files, smart tokens, 563
authentication, 574-576 SSL, 563–564
configuration profiles, RTMT (Real-Time symmetric cryptography, 551–552
Monitoring Tool), saving, 704–705 TLS (Transport Layer Security),
Configuration Tool (Attendant Manager), 563– 564
424–425 web server certificate exchange,
Connected Line ID Presentation field 564– 567
(CallManager), 397 services, 535
Control Center, Serviceability tool symmetric cryptography, PKI (Public Key
(CallManager), 684–686 Infrastructure), 551– 552
controlling center pane, RTMT (Real-Time CSA (Cisco Security Agent), 468–472
Monitoring Tool), 703 headless agent, 469
coresident deployment, MoH (music on hold), managed agent, 469–470
325 protection rules, 471
CoS (class of service), 253-254 supported applications, 470–471
Catalyst switches, configuring, 155–156 CSV (Comma-Separated Values), 737
CoS (class of service) marking, 146 CSV files, creating, BAT (Bulk
counters (performance), 695–696 Administration Tool), 128– 131
RTMT (Real-Time Monitoring Tool), CTI category, RTMT (Real-Time Monitoring
696– 697 Tool), 704
CRC (cyclic redundancy check) checksums, CTIManager, 42
542 activating, 423–424
CRLs (certificate revocation lists), 557 CTL (Certificate Trust List) Providor,
PKI (Public Key Infrastructure), 562 CallManager, 43
cryptography, 535–537 CTL (Certificate Trust List) client
asymmetric cryptography, PKI (Public Key authentication, 600–602
Infrastructure), 552– 553 certificates, 580–583
authentication, 537–538 CallManager certificates, 580
digital signatures, 544–546 CAPF certificates, 580
hash functions, 541–544 MICs (manufacturing installed
data authenticity, 535 certificates), 581
data confidentiality, 535 TFTP server certificates, 580
data integrity, 535 encryption, 600–602
data nonrepudiation, 535 installing, 600–602
encryption, 537–538 Customer Directory Plugin, CallManager, 10
asymmetric encryption, 540–541 cyclic redundancy check (CRC) checksums,
symmetric encryption, 538–540 542
PKI (Public Key Infrastructure), 551
as trusted third-party protocol,
553– 556
844 Daemen, Joan
D device reports, CAR (Call Detail Record

{CDR} Analysis and Reporting) tool, 740
Daemen, Joan, 539 Device Search wizard, RTMT (Real-Time
data authenticity, 535 Monitoring Tool), 707
data confidentiality, 535 device security mode, IP Phone, 607–609
data integrity, 535 devices
data nonrepudiation, 535 BAT (Bulk Administration Tool),
data VLANs, Catalyst switches, 152–154 119, 123– 124
Database Layer Monitor, CallManager, 43 softkeys, assigning to, 354
database replication, CallManager, 38 specifics, BAT (Bulk Administration Tool),
databases, 673 124– 125
management tools, 673–674 DHCP (Dynamic Host Configuration
DBLHelper, 674, 677–678 Protocol), 37
SQL Server 2000 Enterprise Manager, IP Phones, TFTP server addresses, 66
674–677 dial plan administration (CallManager), 8
date/time group configuration, CallManager, dial plan parameters, CAR (Call Detail
80–81 Record {CDR} Analysis and Reporting)
DBHelper, 674, 677–678 tool, configuring, 744–746
DBLHelper, 689 Dialed Number Analyzer, 690
DC Directory, CallManager, 10 Dialed Number Analyzer (DNA). See DNA
DDIs (Discard Digit Instructions), 217–219 (Dialed Number Analyzer)
default device profiles, CallManager, Dick Tracy tool (CallManager), 732
Extension Mobility, 382–383 Differentiated Services Code Point (DSCP)
definition details, CallManager alarms, settings, 653
configuring, 717–719 digit analysis, CallManager, configuring, 200
deleting softkey templates, 355 digit collection, CallManager, configuring,
denial of service (DoS) attacks. See DoS 201–203
(denial of service) attacks digital gateways, 161–162
dependency records, 771 digital signatures, 544–546
accessing, 772–774 RSA algorithm, 545–546
enabling, 771–772 Direct Transfer, configuring, 357
deployment tools, VT (Video Telephony) directories, CallManager Attendant Console,
Advantage, 655–657 412–413
design models, IP telephony networks, 23 directory display, IP Phone Services, 372
clustering over the IP WAN, 30–31 Directory Manager, passwords, changing, 775
multisite deployment model, 25–27 Directory Number field (hunt group
multisite deployment model with distributed configuration), 422
call processing, 27–30 Directory pane (Attendant Console), 430
single-site deployment model, 23–25 directory services (CallManager), 8
destination IP, 280 Disable Speakerphone and headset setting (IP
Device category, RTMT (Real-Time Phone), 526
Monitoring Tool), 703 disabling settings, CallManager
device control (CallManager), 8 Administration, 526– 530
Device Pool field (pilot point configuration), Discard Digit Instructions (DDIs). See DDIs
420 (Discard Digit Instructions)
device pools Discovery Protocol, 650–651
CallManager, configuring, 104–105 displays
IP Phones call forwarding, configuring, 349–351
configuring, 75–87 IP Phone Services, 372
designs, 92–95
features 845
DNA (Dialed Number Analyzer), 776–779 event levels, CallManager, 714

installing, 777–778 Event Viewer, 689, 697–698
DNs (directory numbers), partitions, 254–255 exploits
configuring, 256–258 toll frauds, 503–504
DNS reliance, IP Phones, removing, 73–75 ad hoc conferencing, 516–517
DNS servers, identifying, 37 CFA (Call Forward All), 504–507
DoS (denial of service) attacks, 573 CMC (Client Matter Codes), 511
DSCP (Differentiated Services Code Point) commonly exploited area codes,
settings, 653 507– 509
DSPs (digital signal processors), 310 external transfer restrictions, 511–516
DTMF (dual tone multifrequency), gateways, FAC (Forced Authorization Codes),
162 510–511
dual VLANs, configuring, 153–154 inside facilitators, 505
Dynamic Host Configuration Protocol social engineering, 505
(DHCP), 37 time-of-day routing, 509–510
dynamic link libraries (DLLs), 461 transfers from voice mail, 505–507
Windows 2000 Server, 475–476
extended functions, CallManager, 43
E extension mobility, CallManager, 43
Extension Mobility (CallManager), 377–378
Edit menu (Attendant Console), 427
Emergency Responder (ER), 461 configuring, 378–389
enabling dependency records, CallManager, default device profiles, 382–383
771–772 service parameters, 381–382
encryption, 535–538, 573, 588–589, 597 user device profiles, 383–386
asymmetric encryption, 540–541 external call routing, CallManager, 187–188
configuring, 597–599 external transfers, restricting, 511–516
CTL client, 600–602
IP Phone, enabling, 530
IP telephony threats, 573 F
secure call flow, 591 FACs (Forced Authentication Codes), 389–
SRTP media encryption, 589–590 390
packet format, 590 configuring, 392–394
symmetric encryption, 538–540 creating, 393–394
TLS AES encryption, 589 route patterns, enabling, 394–395
endpoints toll fraud, preventing, 510–511
AAR (Automated Alternate Routing), failover process, SRST (Survivability Remote
configuring, 285–288 Site Telephony), 298
H.323 endpoints, 625–628 Fast Dial service, CallManager, 111–112
IP phones, threats to, 523–530 features, 347, 377
SCCP endpoints, 625–627 abbreviated dialing, 348
enrollment, PKI (Public Key Infrastructure), Auto Answer, 349
559-561 Barge, 365–366
IP telephony, 583–585 Call Display Restrictions, 395–396
man-in-the-middle enrollment attacks, Calling Line ID Presentation,
559-561 396– 397
entry-level IP Phones, 54–56 Connected Line ID Presentation, 397
ER (Emergency Responder), 461 Ignore Presentation Indicators,
Ethereal tool (CallManager), 732–733 397– 398
EtherSwitch modules, PoE (Power over call forwarding, 349–351
Ethernet), Catalyst switches, 149 Call Join, 358
846 features
Call Park, 359–360 frauds, toll frauds, 503–504

Call Pickup, 360–363 ad hoc conferencing, 516–517
call waiting, 347 CFA (Call Forward All), 504–507
CallManager Extension Mobility, 377–378 CMC (Client Matter Codes), 511
configuring, 378–389 commonly exploited area codes, 507–509
default device profiles, 382–383 external transfer restrictions, 511–516
service parameters, 381–382 FAC (Forced Authorization Codes),
user device profiles, 383–386 510–511
Cisco Call Back, 364 inside facilitators, 505
CMCs (Code Matter Codes), 389–392 social engineering, 505
configuring, 392–394 time-of-day routing, 509–510
creating, 392 transfers from voice mail, 505–507
route patterns, 394–395 front-end camera control (FECC), video call
Direct Transfer, 357 flow, 622
FACs (Forced Authentication Codes), fully qualified domain name (FQDN), 37
389– 390 functional groups, MLA (Multilevel
configuring, 392–394 Administration), 489–490
creating, 393–394 creating, 492–499
route patterns, 394–395 functions, CallManager, 8
Group Call Pickup, 361
hold, 347
Immediate Divert, 358–359
IP Phone Services, 368–372
G
gatekeeper-based call admission control,
MCID (Malicious Call Identification), 399 288– 296
configuring, 399–403 gatekeeper-controlled intercluster trunks, 640
multiple calls per line appearances, gatekeepers
configuring, 355–357 call admission control, options, 641–643
Other Group Call Pickup, 361 H.323 gatekeepers, communication,
precedence, 403–404 289–296
preemption, 403–404 Gateway Statistics Utility (GSU), 764
Privacy, 367–368 gateways, 161
redial, 347 analog gateways, 161–162
softkey templates, 351–352 calling search spaces, 256
adding, 352 CallManager
assigning, 354 call survivability, 163
deleting, 355 redundancy support, 162
modifying, 353–354 CAR (Call Detail Record {CDR} Analysis
speed dialing, 348 and Reporting) tool, configuring,
transfers, 347 746–747
FECC (front-end camera control), video call communications, 163–164
flow, 622 configuring, 164–174
File menu (Attendant Console), 427 controlling, SIP, 164
file sharing, Windows 2000 Server, security, core requirements, 162–163
467 digital gateways, 161–162
fixed audio sources, MoH (music on hold), DTMF (dual tone multifrequency) relay
configuring, 332–333 capabilities, 162
Forced Authentication Codes (FACs). See H.323 gateways, 163
FACs (Forced Authentication Codes) configuring, 165–169
FQDN (fully qualified domain name), 37
impersonation attacks 847
MGCP gateways, 163 hardware requirements, VT (Video

configuring, 169–172 Telephony) Advantage, 661– 662
Non-IOS MGCP gateways, configuring, hash functions, 541–544
172–174 headless agent, CSA (Cisco Security Agent),
SCCP gateways, 164 469
supplementary services support, 162 Help menu (Attendant Console), 427
generating CAPF reports, 609–611 Help menu, Serviceability tool
Generic Route Encapsulation (GRE), 525 (CallManager), 684
graph view (Performance Monitor), 700–701 HIPS (host-based intrusion prevention
graphical display (IP Phone Services), 372 system), 468
Gratuitous ARP setting (IP Phone), 526 histogram view (Performance Monitor),
disabling, 528–529 700– 701
GRE (Generic Route Encapsulation), 525 hold feature, 347
Group Call Pickup, configuring, 361 host-based intrusion prevention system
group configuration, CallManager, 79–81 (HIPS), 468
groups, MLA (Multilevel Administration) hot fix policy, Windows 2000 Server
functional groups, 489–499 Operating System, 461–462
user groups, 490–499 HTTPS (Secure HTTP)
GSU (Gateway Statistics Utility), 764 CallManager, 483
securing, 483–487
certificates, 484–485
H self-signed certificates, 485–487
Hunt Configuration menu, 241
H.225 gatekeeper controlled trunks, 175
H.225 gatekeeper-controlled intercluster hunt groups, 235, 411
trunks, 640 call-distribution components, 235–236
H.261 video codec, 624 hunting and forwarding, 240–241
H.263 video codec, 623–624 line groups, 236–243
H.264 video codec, 624 CallManager Attendant Console, 413–414
H.320 gateways, IP video telephony, 619 configuring, 420–422
H.323 endpoints, IP video telephony, 625–628 Hunt List Configuration window, 244
H.323 gatekeepers hunt lists, 241–242
communication, 289–296 configuring, 243–245
IP video telephony, 620 hunt options, line groups, 238–240
H.323 gateways, 163 Hunt Pilot Configuration window, 246
configuring, 165–169 hunt pilots, 241–242
hardening configuring, 245–247
IP phones, 523 hunting and forwarding, 240–241
disabling settings, 526–530
endpoint threats, 523–530
rogue image blocking, 525–526 I
Windows 2000 Server Operating System, identity spoofing, 573
463–467 Ignore Presentation Indicators
file sharing, 467 (CallManager), 397–398
IP Telephony Operating System image authentication, IP Phone, 525–526
scripts, 464–466 image display (IP Phone Services), 372
hardware, conference bridges, 311–315 images, rogue images, blocking, 525–526
configuring, 315–316 Immediate Divert, configuring, 358–359
impersonation attacks, 573
848 inbound telephone calls
inbound telephone calls, routing, methods, Serviceability tool (CallManager), 679

409 International Telecommunication Union
incoming telephone calls, CallManager Telecommunication (ITU-T), audio codec
Attendant Console standards, 67
queuing, 415 intracluster communication, IP Phones,
routing, 414– 415 configuring for, 73–87
independent and separated PKI topology, 579 intracluster run-time data, CallManager
informational tones, 309 server, 18–19
infrastructure layer (Unified intrgrity, loss of, 573
Communications), 7 IOS gatekeepers, commands, 293
initial configuration, CAR (Call Detail Record IP Communicator, 60
{CDR} Analysis and Reporting) tool, 741– IP Conference Station 7936, 60
742 IP Manager Assistant, CallManager, 43
inline power, Catalyst switches, 145 IP Manager Assistant (IPMA). See IPMA
input display (IP Phone Services), 372 (IP Manager Assistant)
inserting, IP Phones, BAT (Bulk IP Phone 7902G, 54–55
Administration Tool), 132– 134 IP Phone 7905G, 55
inside facilitators, toll fraud, 505 IP Phone 7910G+SW, 55
installation IP Phone 7911G, 55
Assistant Console, IPMA (IP Manager IP Phone 7912G, 55
Assistant), 450– 452 IP Phone 7914 Expansion Module, 60
Attendant Console client, 425–426 IP Phone 7940G, 59
BAT (Bulk Administration Tool), 121–123 IP Phone 7941G, 57, 59
CallManager, clean installation process, IP Phone 7941G-GE, 57
35– 45 IP Phone 7960G, 59
CTL client, 600–602 IP Phone 7961G, 57, 59
DNA (Dial Number Analyzer), 777–778 IP Phone 7961G-GE, 57
subscription servers, 38 IP Phone 7970G, 58
VT (Video Telephony) Advantage, IP Phone 7971G-GE, 58
663– 665 IP Phone 7980G, 58
clients, 661–667 IP Phone 79XXG-GE, 57
installation configuration data, CallManager, IP Phone Help Desk Utility (IPHDU), 764
clean installations, 36–38 IP Phone Information Utility (IPIU), 764
installation disks, CallManager, clean IP Phones, 53
installations, 35–36 authentication, enabling, 530
integration, SIP (Session Initiation Protocol), BAT (Bulk Administration Tool), 123–135
CallManager, 181–183 inserting, 132–134
integrity, authentication, 585–588 TAPS (Tool for Auto-Registered
Interactive Voice Response (IVR), 8 Phones Support), 136–138
intercluster gatekeeper-controlled trunks, 175 updating, 134–135
intercluster nongatekeeper-controlled trunks, validating, 131–132
175 CallManager
intercluster trunks automatic registration, 90–92
calling search spaces, 256 configuration for, 87–92
CallManager, 640 configuring for, 73–87
configuring, 176–179 device pool design, 92–95
interdigit timeouts, CallManager, 202–203 group configuration, 79–81
interfaces interaction, 8–9
Attendant Console, 427
IP video telephony 849
manual directory number IP telephony, 5

configuration, 87–90 IP telephony endpoints, 59–61, 64
region configuration, 82 IP Telephony Monitor (ITM), 763
registering with, 67 IP telephony networks, design models, 23
services, 110–111 clustering over the IP WAN, 30–31
softkey template configuration, 83–84 multisite CallManager deployment model,
Catalyst Switches, obtaining power 25–27
from, 65 multisite CallManager deployment model
codec support, 67 with distributed call processing, 27–30
configuration, 368-372 single-site CallManager deployment model,
intracluster communication, 73–87 23–25
device pools, configuring, 75–87 IP Telephony Operating System scripts,
device security mode Windows 2000 Server, hardening, 464–466
configuring, 607–608 IP telephony systems, threats to, 573
negotiating, 608–609 protecting against, 574–576
displays, 372 IP telephony users, CallManager
DNS reliance, removing, 73–75 adding, 102–104
encryption, enabling, 530 call forwarding, 107–108
endpoints, threats to, 523–530 configuration, 102–114
entry-level IP Phones, 54–56 device pool configuration, 104–105
features, 61, 64 Fast Dial service, 111–112
G.711 coder-decoder (codec), 24 IP Phone services, 110–111
Gratuitous ARP setting, disabling, 528–529 locale settings, 113–114
hardening, 523 logon options, 105–106
image authentication, 525–526 Message Waiting Lamp page, 112–113
IP addresses, obtaining, 66 Personal Address Book service, 111– 112
IP video telephony, 620 speed dialing, 108–110
maximum cluster size, 18 user database, 101–102
midrange IP Phones, 56–59 User Options welcome screen, 106– 107
PC Port setting, disabling, 526–527 IP video telephony, 619
PC Voice VLAN Access setting, disabling, 7985 IP Phone, 620
529–530 bandwidth
rogue images, blocking, 525–526 calculating, 630–631
RTP (Real-Time Transport Protocol), IP call admission control, 631–632
Phones, 8 management, 628–632
settings, disabling, 526–530 requirements, 629–630
Settings Access setting, disabling, 526–527 call admission control, 637–638
SIP (Session Initiation Protocol), 8 clusters, 632–643
startup process, 64–67 location configuration, 635–637
stored phone images, loading, 65 region configuration, 634–635
template buttons, 84–86 Retry Video Call as Audio setting,
TFTP servers 638–640
addresses, obtaining, 66 CallManager, 619
contacting, 66 components, 619–621
upper-end IP Phones, 56–59 H.320 gateways, 619
VLAN, configuring, 65 H.323 endpoints, 625–628
VT (Video Telephony) Advantage, H.323 gatekeepers, 620
configuring for, 657–661 MCUs (Multipoint Control Units), 619
Web Access setting, disabling, 527–528 protocols, 624–628
850 IP video telephony
SCCP endpoints, 620, 625–627 ITU-T (International Telecommunication

video calls, 621 Union Telecommunication), audio codec
call flow, 622–623 standards, 67
CallManager, 621–622 IVR (Interactive Voice Response), 8
codec support, 623–624
VT (Video Telephony) Advantage, 620, 647
call-routing, 654
calls, 651–652
J-K
Java applications, CallManager alarms,
client installation, 661–667 configuring, 716–717
components, 647–648 JTAPI (Java Telephony Application
configuring, 653–654 Programming Interface), 716
configuring IP phones for, 657–661
deployment tools, 655–657 key exchange, PKI (Public Key
hardware requirements, 661–662 Infrastructure), 551
installing, 663–665 as trusted third-party protocol, 553– 556
protocols, 649–650 asymmetric cryptography, 552–553
software requirements, 661–662 CA (Certificate Authority), 556–559
supported standards, 648–649 certificates, 557–559
video modes, 652–653 CRL (certificate revocation lists), 557
IP Voice Media Streaming application, 312 enrollment, 559–561
CallManager, 42 entities, 556–559
IP Voice Streaming application, key storage, 562–563
activating, 310 revocation methods, 561–562
IP WANs, CallManager clustering, 30–31 Smart Cards, 563
IP/VC 3500 H.320 gateways, 619 smart tokens, 563
IP/VC 3500 Series MCUs (Multipoint Control SSL, 563–564
Units), 619 symmetric cryptography, 551–552
IP/VC-3500 Series Video Multipoint TLS (Transport Layer Security), 563– 564
Conference Unit, 313 web server certificate exchange, 564– 567
IPHDU (IP Phone Help Desk Utility), 764 keys, storage
IPIU (IP Phone Information Utility), 764 IP telephony, 585
IPMA (IP Manager Assistant), 437 PKI (Public Key Infrastructure), 562– 563
activating, 441
architecture, 439–440
assistants, configuring, 444–448
Assitant Console, intalling, 450–452
L
call divert targets, configuring, 449–450 L2TP (Layer 2 Protocol Tunneling), 525
managers, configuring, 444–448 Layer 2 Protocol Tunneling (L2TP), 525
proxy-line support, 438 Layers, Unified Communications, 7
restarting, 444 LDAP (Lightweight Directory Access
service parameters, configuring, 442–443 Protocol) directory, CallManager, 101–102
shared-line support, 438–439 Line Group Configuration window, 242
configuring for, 440–452 line groups, 236–237, 241–242
IPMASysUser account, password, call-distribution algorithms, 237–238
changing, 775 configuring, 242–243
IPsec, 525 hunt options, 238–240
ITM (IP Telephony Monitor), 763 Line Number field (hunt group
configuration), 422
LMHOST files, creating, 37
media resources 851
loading stored phone images, IP Phones, 65 shared-line support, configuring for,

local name resolution, 37 440–452
locale settings, CallManager, 113–114 shared-line supprt, 438–439
locally significant certificates (LSCs). See managers, IPMA (IP Manager Assistant),
LSCs (locally significant certificates) configuring, 444–448
location configuration, call admission control, man-in-the-middle enrollment attacks, PKI
635–637 (Public Key Infrastructure), 560
Location field (pilot point configuration), 420 manual directory number configuration, IP
locations-based call admission control, Phones, 87–90
276– 288 MCID (Malicious Call Identification), 399
AAR (Automated Alternate Routing), configuring, 399–403
280– 288 MCSs (Media Convergence Servers),
configuring, 278–279 CallManager, 11
logon options, users, CallManager, 105–106 MCUs (Multipoint Control Units), 653
logs, QRT (Quality Report Tool), viewing, IP video telephony, 619
782–785 Media Convergence Servers (MCSs).
longest idle time call-distribution See MCSs (Media Convergence Servers)
algorithm, 238 Media Gateway Control Protocol (MGCP)
loss of intgrity threats, 573 gateways, 25
loss of privacy threats, 573 Media Resource Group Lists (MRGLs), 653
LSCs (locally significant certificates), 577-579, Media Resource Managers (MRMs). See
602–607 MRMs (Media Resource Managers)
media resources, 309–310
annunciators, 319–320
M configuring, 320–321
conference bridges, 310–311
mail parameters, CAR (Call Detail Record
{CDR} Analysis and Reporting) tool, ad hoc conferencing, 311
configuring, 743–744 Cisco IP Voice Media Streaming
Malicious Call Identification (MCID). See application, 312
MCID (Malicious Call Identification) hardware, 311–313, 315
managed agent, CSA (Cisco Security Agent), hardware configuration, 315–316
469–470 IP/VC-3500 Series Video Multipoint
management tools Conference Unit, 313
CallManager, 688–690 Meet-Me conferencing, 311
databases, 673–674 NM-HD network modules, 313
DBLHelper, 674, 677–678 NM-HD-2VE network modules, 313
SQL Server 2000 Enterprise Manager, NM-HDV network modules, 313
674–677 NM-HDV2 network modules, 313
Manager Assistant, 437 WS-SVC-CMM conference bridge
activating, 441 type, 312
architecture, 439–440 WS-X6608-E1 conference bridge
Assistant Console, installing, 450–452 type, 312
assistants, configuring, 444–448 WS-X6608-T1 conference bridge
call divert targets, configuring, 449–450 type, 312
managers, configuring, 444–448 conferencing, 309
proxy-line supprt, 438 informational tones, 309
restarting, 444 MoH (music on hold), 309, 324–325
service parameters, configuring, 442–443 audio source file creation, 326–327
configuring, 327–334
852 media resources
MRM (Media Resource Manager), 335 access privileges, 491–493

media resource design, 335–336 enabling, 488–489
MRGLs (Media Resource Group functional groups, 489–499
Lists), 338–341 user groups, 490–499
MRGs (Media Resource Groups), modular switching, PoE (Power over
336–338 Ethernet), Catalyst switches, 148
MTPs (Media Termination Points), MoH (music on hold), 309, 324–325
316– 317 audio source files, creating, 326–327
configuring, 318 configuring, 327–334
spoken word progress tones), 309 audio source files, 330–331
transcoders, 321–323 audio source ID assignments, 334
configuring, 323–324 audio translator, 327–328
Media Termination Points, 309 fixed audio source, 332–333
Media Termination Points (MTPs), 316–318 servicewide settings, 331–332
media transfers, secure media transfers, coresident deployment, 325
CallManager, 574–575 standalone deployment, 325
Meet-Me conferencing, 311 MoH Audio Translator, CallManager, 42
menu (IP Phone Services), 372 activating, 310
menu bar, RTMT (Real-Time Monitoring monitoring performance, 695
Tool), 703 Event Viewer, 697–698
menus Performance Monitor, 699
Attendant Console, 427 graph view, 700–701
Serviceability tool (CallManager), 678 histogram view, 700–701
Message Waiting Lamp page, CallManager, report view, 699–700
112–113 Windows Task Manager, 702
messaging interface, CallManager, 42 RTMT (Real-Time Monitoring Tool),
MGCP (Media Gateway Control Protocol) 702– 706
gateways, 25, 163 Alert Central, 707–708
configuring, 169–172 call activity monitoring, 706–707
Microsoft Event Viewer. See Event Viewer configuration profiles, 704–705
Microsoft Performance Monitor Device Search wizard, 707
graph view, 700–701 performance counters, 695-697
histogram view, 700–701 MRGLs (Media Resource Group Lists), 653
report view, 699–700 assigning, 339
Windows Task Manager, 702 configuring, 338–339
Microsoft Performance Monitor. See location design, 340–341
Performance Monitor type design, 340
Microsoft SQL Server 2000 Enterprise MRGs (Media Resource Groups), 336–337
Manager, 674–677, 689 configuring, 337–338
Microsoft Windows Task Manager, MRMs (Media Resource Managers), 309, 335
Perfomance Monitor, 702 annunciators, 319–320
MICs (manufacturing installed certificates) configuring, 320–321
CTL client, 581 conference bridges, 310–311
PKI topologies, 577–578 ad hoc conferencing, 311
midrange IP Phones, 56–59 Cisco IP Voice Media Streaming
midspan power injection, Catalyst switches, application, 312
146 hardware, 311–313, 315
MLA (Multilevel Administration), hardware configuration, 315–316
CallManager, 487–488
Other Group Call Pickup 853
IP/VC-3500 Series Video Multipoint

Conference Unit, 313
N
Meet-Me conferencing, 311 NANP (North American Numbering Plan),
NM-HD network modules, 313 route definitions, 209
NM-HD-2VE network modules, 313 NAT (Network Address Translation), 608
NM-HDV network modules, 313 NativeOS, dual VLANs, configuring, 153–154
NM-HDV2 network modules, 313 negotiation, IP Phone, device security mode,
WS-SVC-CMM conference bridge 608–609
type, 312 Network Address Translation (NAT), 608
WS-X6608-E1 conference bridge NM-HD network modules, 313
type, 312 NM-HD-2VE network modules, 313
WS-X6608-T1 conference bridge NM-HDV network modules, 313
type, 312 NM-HDV2 network modules, 313
media resource design, 335–336 nongatekeeper-controlled intercluster
MoH (music on hold), 324–325 trunks, 640
audio source file creation, 326–327 Non-IOS MGCP gateways, configuring,
configuring, 327–334 172–174
MRGLs (Media Resource Group Lists) North American Numbering Plan (NANP),
assigning, 339 route definitions, 209
configuring, 338–339 Notification Tool, 462
location design, 340–341 NXX, 210
type design, 340
MRGs (Media Resource Groups), 336–337
configuring, 337–338 O
MTPs (Media Termination Points), OCSP (Online Certificate Status Protocol),
316– 317 PKI (Public Key Infrastructure), 562
configuring, 318 OffNet calls, defining, 512–514
transcoders, 321–323 OnNet calls, defining, 512–514
configuring, 323–324 operating systems
MTPs (Media Termination Points), 316–318 Red Hat Linux, 459
Multilevel Administration (MLA). See MLA upgrading, 461
(Multilevel Administration) Windows 2000 Server Operating System
multiple call per line appearances, administrator password policies,
configuring, 355–357 472– 475
multiple-site IP telephony deployments antivirus protection, 467–468
implementing, 275 common exploits, 475–476
CAC (call admission control), CSA (Cisco Security Agent), 468–472
275– 288 hardening, 463–467
SRST (Survivability Remote Site hot fix policy, 461–462
Telephony), 296–298 security, 459
failover process, 298 security patches, 461–462
reference configuration, 300–301 security settings, 476–478
router configuration, 298–299 threats to, 459–461
multipoint control units (MCUs), 653 Other Group Call Pickup, configuring, 361
multisite CallBack design, 25–27
multisite CallBack design with distributed
call processing, 27–30
music on hold (MoH). See MoH
(music on hold)
854 packets
P call activity monitoring, 706–707

configuration profiles, 704–705
packets, SRTP packets, 590 Device Search wizard, 707
parameters, video codecs, 624 periods, time-of-day routing, 260–261
Parked Calls pane (Attendant Console), configuring, 263–265
430– 431 permissions (calling)
Partition field (hunt group configuration), 422 calling search spaces, 254–255
Partition field (pilot point configuration), 420 configuring, 258–259
partitions, 253 gateways, 256
DNs (directory numbers), 254–255 intercluster trunks, 256
configuring, 256–258 CoS (Class of Service), 253–254
Password Changer tool, 774–776 enforcement of, 253
Password Changer tool (CallManager), 690 partitions, 254–255
password policies, administrators, Windows configuring, 256–258
2000 Server, 472–475 time-of-day routing, 260
password protection, CallManager, 38 configuring, 262–268
PAT (Port Address Translation), 608 time periods, 260–261, 263–265
patches, Windows 2000 Server Operating time schedules, 261, 265–268
System, security patches, 461–462 usage scenario, 268–270
PBX systems, reliability, 17 user effects, 262
PC Port setting (IP Phone), disabling, Personal Address Book service, CallManager,
526–527 111–112
PC Voice VLAN Access setting Personal Assistant, 8
(IP Phone), 526 phone certificates, issuing
disabling, 529–530 authentication strings, 605–607
PDF (Portable Document Format), 737 CAPF, 607
Performance category, RTMT (Real-Time phone configuration files, authentication,
Monitoring Tool), 704 CallManager, 574, 576
performance counters, 695–696 phone feature administration
RTMT (Real-Time Monitoring Tool), (CallManager), 8
696– 697 phone images, authentication, CallManager,
Performance Monitor, 689, 699 574–576
graph view, 700–701 phone loads, verifying, 657–658
histogram view, 700–701 phones. See IP Phones
report view, 699–700 phone-to-server authentication, 587–588
Windows Task Manager, 702 Pilot Name field (pilot point
performance monitoring, 695 configuration), 420
Event Viewer, 697–698 Pilot Number field (pilot point
performance counters, 695–696 configuration), 420
RTMT (Real-Time Monitoring Tool), pilot points, 411
696–697 CallManager Attendant Console, 413–414
Performance Monitor, 699 configuring, 419–420
graph view, 700–701 PKI (Public Key Infrastructure), 551
histogram view, 700–701 as trusted third-party protocol, 553–556
report view, 699–700 asymmetric cryptography, 552–553
Windows Task Manager, 702 CA (Certificate Authority), 556–559
RTMT (Real-Time Monitoring Tool), certificates, 544, 557
702– 706 self-signed certificates, 558–559
Alert Central, 707–708 X.509v3 certificates, 557–558
public-key encryption 855
CRLs (certificate revocation lists), 557, 562 privileges (calling)

enrollment, 559–561 calling search spaces, 254–255
man-in-the-middle enrollment configuring, 258–259
attacks, 560 gateways, 256
entities, 556–559 intercluster trunks, 256
IP telephony, enrollment, 583–585 CoS (Class of Service), 253–254
key storage, 562–563 enforcement of, 253
OCSP (Online Certificate Status partitions, 254–255
Protocol), 562 configuring, 256–258
revocation methods, 561–562 time-of-day routing, 260
Smart Cards, 563 configuring, 262–268
smart tokens, 563 time periods, 260–261, 263–265
SSL, 563–564 time schedules, 261, 265–268
symmetric cryptography, 551–552 usage scenario, 268–270
TLS (Transport Layer Security), 563–564 user effects, 262
web server certificate exchange, 564–567 Product Security Incident Response Team
PKI topologies, IP telephony, 576 (PSIRT) Advisory Notification Tool, 462
CAPF (CallManager Certificate Authority programming interface to external
Proxy Funtion), 577 applications (CallManager), 8
CTL clint, 580–583 protection rules, CSA (Cisco Security
LSCs (locally significant certificates), Agent), 471
577–579 protocols
MICs (manufacturing installed certificates), video protocols, 624–628
577–578 VT (Video Telephony) Advantage,
multiple PKI topologies, 579 protocols, support, 649–650
self-signed certificates, 576–577 proxy servers, SIP (Session Initiation
PoE (Power over Ethernet), Catalyst switches, Protocol), 179
146–148 proxy-line support, IPMA (IP Manager
Catalyst EtherSwitch modules, 149 Assistant), 438
Catalyst modular switching, 148 PSTN (preferred public switched telephone
Catalyst stackable switching, 149 network), 209
configuring, 150 PSTN (public switched telephone
device detection, 147–148 network), 275
verifying, 151 PTSN (public switched telephone
Port Address Translation (PAT), 608 network), 161
Portable Document Format (PDF), 737 public key cryptography, PKI (Public Key
positioning softkeys, modifying, 353–354 Infrastructure), 552–553
postinstallation procedures, CallManager, Public Key Infrastructure (PKI). See PKI
clean installations, 40–41 (Public Key Infrastructure)
powering IP Phones, Catalyst switches, public keys (PKI)
146–151 exchanging, 553– 567
precedence (CallManager), 403–404 storage, 562– 563
preemption (CallManager), 403–404 public switched telephone network (PSTN),
preferred public switched telephone network 161, 275
(PSTN), 209 public-key encryption, 540–541
Privacy, configuring, 367–368
privacy, loss of, 573
privileges, access privieleges MLA (Multilevel
Administration), 491–493
856 Q.931 Translator (CallManager)
Q restrictions (calling)
calling search spaces, 254–255
Q.931 Translator (CallManager), 732 configuring, 258–259
QRT (Qulaity Report Tool), 780–784 gateways, 256
logs, viewing, 782–785 intercluster trunks, 256
softkey, activating, 781–782 CoS (Class of Service), 253–254
QRT Viewer, 690 enforcement of, 253
queuing incoming telephone calls, external call transfers, 511-516
CallManager Attendant Console, 415 partitions, 254–255
time-of-day routing, 260
R configuring, 262–268
Real-Time Monitoring Tool (RTMT). See time periods, 260–261, 263–265
RTMT (Real-Time Monitoring Tool) time schedules, 261, 265–268
Real-Time Transport Protocol (RTP). See usage scenario, 268–270
RTP (Real-Time Transport Protocol) user effects, 262
records, BAT (Bulk Administration Tool), 119 Retry Video Call as Audio setting, call
Red Hat Linux, 459 admission control, 638–640
redial feature, 347 revocation methods, PKI (Public Key
redirect servers, SIP (Session Initiation Infrastructure), 561–562
Protocol), 180 Rijmen, Vincent, 539
redundancy, CallManager, gateways, 162 Rijndael cipher, 539
redundancy designs, CallManager server, Ring No Answer Reversion (RNAR), 236
19–22 ringback tones, 319
references, SRST (Survivability Remote Site RIS (Real-Time Information Server) Data
Telephony), configuring, 300–301 Collector, CallManager, 42
region configuration, CallManager, 82 Rivest, Ronald L., 541
call admission control, 634–635 RNAR (Ring No Answer Reversion), 236
registrar servers, SIP (Session Initiation rogue images
Protocol), 180 blocking, 525–526
registrations, IP Phones, automatic IP phones, blocking, 525–526
registrations, 90–92 route filters, route patterns, 209–210
reimaging, CallManager, 46 applying, 214–215
remote management tools configuring, 212–214
ITEM (IP Telephony Environment example, 215–217
Monitor), 763–764 route filter tags, 211–212
SNMP (Simple Network Management route groups, CallManager, configuring,
Protocol), 764–768 189–192
Syslog, 769–771 route lists, CallManager, configuring, 192–194
report scheduling, CAR (Call Detail Record route patterns
{CDR} Analysis and Reporting) tool, CallManager, configuring, 194–199
748–752 CMCs (Code Matter Codes), enabling,
report view (Performance Monitor), 699–700 394–395
reports FACs (Forced Authentication Codes),
CAPF reports, generating, 609–611 enabling, 394–395
route plans, generating, 229 route plans, CallManager, 187, 204–205, 209
restarting IPMA (IP Manager Assistant), 444 configuration, 189–205
DDIs (Discard Digit Instructions), 217–219
digit analysis, 200
security 857
digit collection, 201–203 integrity, 585–588

reports, 229 key storage, 585
route filters, 209–217 phone-to-server authentication,
route groups, 189–192 587– 588
route lists, 192–194 server-to-phone authentication,
route patterns, 194–199 586– 587
transformation masks, 219–225 CallManager, HTTPS (Secure HTTP),
translation patterns, 226–228 483–487
routers, SRST (Survivability Remote Site cryptography, 535–537
Telephony), configuring, 298–299 authentication, 537–538
routing inbound telephone calls data authenticity, 535
methods, 409 data confidentiality, 535
CallManager Attendant Console, 414– 415 data integrity, 535
time-of-day routing, 509–510 data nonrepudiation, 535
RSA algorithm, 541 encryption, 537–538
digital signatures, 545–546 services, 535
RTMT (Real-Time Monitoring Tool), 690, encryption, 588–589
696– 697, 702–706 asymmetric encryption, 540–541
Alert Central, 707–708 call flow, 591
call activity, monitoring, 706–707 configuring, 597–599
configuration profiles, saving, 704–705 CTL client, 600–602
Device Search wizard, 707 SRTP media encryption, 589–590
RTP (Real-Time Transport Protocol), 8, 650 symmetric encryption, 538–540
TLS AES encryption, 589
IP telephony threats, protecting against,
S 574–576
PKI (Public Key Infrastructure), 551, 576
SCCP (Skinny Client Control Protocol), 650
CallManager, 8–9 as trusted third-party protocol,
endpoints, IP video telephony, 553– 556
620, 625–627 asymmetric cryptography, 552–553
gateways, 164 CA (Certificate Authority), 556–559
schedules, time-of-day routing, 261 CAPF (CallManager Certificate
applying, 266–268 Authority Proxy Funtion), 577
configuring, 265–266 certificates, 557–559
SDK (Software Development Kit), CRL (certificate revocation lists), 557
CallManager, 7 CTL client, 580–583
Secure HTTP (HTTPS). See HTTPS (Secure enrollment, 559–561, 583-585
HTTPS) entities, 556–559
secure media transfers, CallManager, key storage, 562–563
574–575 LSCs (locally significant certificates),
secure signaling, CallManager, 574–575 577–579
security MICs (manufacturing installed
authentication certificates), 577–578
certificate storage, 585 multiple PKI topologies, 579
configuring, 597–599 revocation methods, 561–562
CTL client, 600–602 self-signed certificates, 576–577
digital signatures, 544–546 Smart Cards, 563
hash functions, 541–544 smart tokens, 563
SSL, 563–564
858 security
symmetric cryptography, 551–552 interface, 679

TLS (Transport Layer Security), menus, 678
563– 564 Service Activation tool, 686
web server certificate exchange, Services MMC, 687
564– 567 Tools service, 681–682
Windows 2000 Server Operating Trace service, 680–681
System, 459 services
administrator password policies, CallManager, activating, 41–45
472– 475 IP Phones, 110–111
antivirus protection, 467–468 Services MMC, Serviceability tool
common exploits, 475–476 (CallManager), 687-689
CSA (Cisco Security Agent), 468–472 servicewide settings, MoH (music on hold),
hardening, 463–467 configuring, 331–332
hot fix policy, 461–462 Session Initiation Protocol (SIP). See SIP
patches, 461-462 (Session Initiation Protocol)
security patches, 461–462 settings, CallManager Administration,
security settings, 476–478 disabling, 526–530
threats to, 459–461 Settings Access setting (IP Phone), 526–527
security settings, Windows 2000 Server, SHA-1 keys, exchanges, 588
476–478 Shamir, Adi, 541
self-signed certificates shared-line support, IPMA (IP Manager
CallManager, accessing through, 485–487 Assistant), 438–439
PKI (Public Key Infrastructure), 558–559, activating, 441
576–577 assistant configuration, 444–448
Server category, RTMT (Real-Time Assistant Console installation, 450– 452
Monitoring Tool), 703 call-divert target configuration, 449– 450
servers configuring for, 440–452
CallManager, 11 manager configuration, 444–448
call-processing deployment models, restarting, 444
23–31 service parameters, 442–443
clusters, 17–22 signaling, CallManager, 8-9
configuring, 416–426 secure signaling, 574–575
subscription servers, installing, 38 signatures, digital signatures, 544–546
server-to-phone authentication, 586–587 Simple Network Management Protocol
Service Activation tool, Serviceability tool (SNMP), 764-768
(CallManager), 686 single-site CallBack design, 23–25
Service category, RTMT (Real-Time SIP (Session Inititation Protocol), 179
Monitoring Tool), 704 CallManager, 179
service parameters, CallManager, Extension components, 179–181
Mobility, 381–382 configuration, 181–183
Serviceability Alarm menu (CallManager), integration, 181–183
713–714 proxy servers, 179
Serviceability Reporter, CallManager, 43 redirect servers, 180
Serviceability tool (CallManager), registrar servers, 180
678, 687– 688 user agents, 180
Alarm service, 679–680 gateways, controlling, 164
Application menu, 683 IP Phones, 8
Control Center, 684–686 trunks, 176, 640
Help menu, 684
system preferences 859
Skinny Client Control Protocol (SCCP). SSL (Secure Socket Layer), PKI
See SCCP (Skinny Client Control Protocol) (Public Key Infrastructure), 563–564
Smart Cards, PKI (Public Key stackable switching, PoE (Power over
Infrastructure), 563 Ethernet), Catalyst switches, 149
smart tokens, PKI (Public Key standalone deployment, MoH (music on
Infrastructure), 563 hold), 325
SNMP (Simple Network Management Standard IPMA Shared Mode Manager
Protocol), 764–768 softkey template, 438
social engineering toll fraud, 505 startup process, IP Phones, 64–67
softkey template configuration, CallManager, static IP information, CallManager, 37
83–84 stopping CallManager server manually, 19
softkeys storage
QRT (Quality Report Tool), activating, certificates, 585
781–782 keys, PKI (Public Key Infrastructure),
Standard IPMA Shared Mode Manager 562– 563, 585
template, 438 stored phone images, IP Phones, loading, 65
templates, 351–352 subscriber servers, installing, 38
adding, 352 Summary category, RTMT (Real-Time
assigning, 354 Monitoring Tool), 703
deleting, 355 Survivability Remote Site Telephony (SRST).
modifying, 353–354 See SRST (Survivability Remote Site
software requirements, VT (Video Telephony) Telephony)
Advantage, 661– 662 Switched Port Analyzer (SPAN), 525
source IP, 280 switches, Catalyst switches, 145
SPAN (Switched Port Analyzer), 525 Auxiliary VLAN support, 145
speed dialing, 108-110, 348, 429-430 CoS (class of service), 155–156
Speed Dials pane (Attendant Console), CoS marking, 146
429–430 data VLANs, 152–154
spoken work progress tones, 309 dual VLANs, 153–154
SQL (Structured Query Language) inline power, 145
databases, 673 IP Phones, 146–151
clusters, 17-18 IP telephony, 145–146
management tools, 673–674 midspan power injection, 146
DBLHelper, 674, 677–678 PoE (Power over Ethernet), 146–151
SQL Server 2000 Enterprise Manager, voice VLANs, 152–154
674–677 wall power, 146
SQL (Structured Query Language) Server symmetric cryptography, PKI (Public Key
2000, 10 Infrastructure), 551–552
SQL Server 2000 Enterprise Manager, symmetric encryption, 538–540
674– 677, 689 Syslog, 769–771
SRST (Survivability Remote Site Telephony), system database, CAR (Call Detail Record
275, 296–298 {CDR} Analysis and Reporting) tool,
commands, 299 configuring, 752–756
configuring, 300–301 system parameters, CAR (Call Detail Record
failover process, 298 {CDR} Analysis and Reporting) tool,
references, configuring, 300–301 configuring, 742–743
routers, configuring, 298–299 system preferences, CAR (Call Detail Record
SRTP media encryption, 589–590 {CDR} Analysis and Reporting) tool,
860 system reports
system reports, CAR (Call Detail Record usage scenario, 268–270

{CDR} Analysis and Reporting) tool, 740 user effects, 262
TLS (Transport Layer Security)
PKI (Public Key Infrastructure), 563–564
T sessions, certificate exchanges, 586
TLS AES encryption, 589
tab bar, RTMT (Real-Time Monitoring
Tool), 704 TLS SHA-1 sessions, key exchanges, 588
tags, route filters, 211–212 toll frauds, 503–504
TANDBERG SCCP endpoints, IP video ad hoc conferencing, dropping, 516–517
telephony, 620 CFA (Call Forward All), preventing,
TAPS (Tool for Auto-Registered Phones 504–507
Support), 119 CMC (Client Matter Codes), using, 511
BAT (Bulk Administration Tool), 136–138 commonly exploited area codes, blocking,
TCD (Telephony Call Dispatcher), 411-412 507–509
activating, 423–424 external transfers, restricting, 511–516
telephones. See IP Phones FAC (Forced Authorization Codes), using,
Telephony Call Dispatcher, CallManager, 42 510–511
telephony endpoints (IP), 59–61, 64 inside facilitators, 505
templates social engineering, 505
BAT (Bulk Administration Tool) time-of-day routing, using, 509–510
templates, 438 transfers from voice mail, preventing,
configuring, 126–127 505–507
softkey templates, 351–352 Tomcat Web Application Manager,
adding, 352 CallManager Extension Mobility,
assigning, 354 configuring, 379–381
deleting, 355 Tool for Auto-Registered Phones Support
modifying, 353–354 (TAPS). See TAPS (Tool for Auto-
Standard IPMA Shared Mode Registered Phones Support)
Manager, 438 Tools service, Serviceability tool
text display (IP Phone Services), 372 (CallManager), 681–682
TFTP server certificates, CTL client, 580 top down call-distribution algorithm, 237, 239
TFTP servers, 42 Trace Analysis tool (CallManager), 725–728
addresses, obtaining, 66 Trace Collection tool (CallManager), 728–730
contacting, IP Phones, 66 Trace service, Serviceability tool
threats (CallManager), 680–681
CallManager, Administrator, 483 traces, 720
IP phones Bulk Trace Analysis tool, 730–732
endpoints, 523–530 configuring, 719–725
rogue images, 525–526 Dick Tracy tool, 732
IP telephony systems, 573 Ethereal tool, 732–733
protecting against, 574–576 Q.931 Translator, 732
Windows 2000 Server Operating System, Trace Analysis tool, 725–728
459–461 Trace Collection tool, 728–730
time-of-day routing, 253, 260 Voice Log Translator, 732
configuring, 262–268 transaction files, 673
time periods, 260–261, 263–265 transcoders, 321–323
time schedules, 261, 265–268 configuring, 323–324
toll fraud, preventing, 509–510 transcoding, 309
transfers, 347
user features 861
external transfers, restricting, 511–516 Auto Answer, 349

voice mail, toll fraud, 505–507 Barge, 365–366
transformation masks, 219–220 Call Display Restrictions, 395–396
called-party transformations, configuring, Calling Line ID Presentation,
220–224 396– 397
example, 224–225 Connected Line ID Presentation, 397
translation patterns, 226 Ignore Presentation Indicators,
configuring, 226–227 397– 398
practical use of, 227–228 call forwarding, 349–351
Transport Layer Security (TLS). See TLS Call Join, 358
(Transport Layer Security) Call Park, 359–360
trunks, 161, 174–175 Call Pickup, 360–363
H.225 gatekeeper controlled trunks, 175 call waiting, 347
intercluster gatekeeper-controlled CallManager Extension Mobility, 377–378
trunks, 175 configuring, 378–389
intercluster nongatekeeper-controlled default device profiles, 382–383
trunks, 175 service parameters, 381–382
intercluster trunks user device profiles, 383–386
calling search spaces, 256 Cisco Call Back, 364
CallManager, 640 CMCs (Code Matter Codes), 389, 391–392
configuring, 176–179 configuring, 392–394
SIP trunks, 176 creating, 392
route patterns, 394–395
Direct Transfer, 357
U FACs (Forced Authentication Codes),
389– 390
ultilevel precedence, 403–404
ultilevel preemption, 403–404 configuring, 392–394
Unified CallManager. See CallManager creating, 393–394
Unified Communications, 5–7 route patterns, 394–395
CallManager, 7–8 Group Call Pickup, 361
components, 10 hold, 347
IP Phone interaction, 8–9 Immediate Divert, 358–359
servers, 11 IP Phone Services, 368–372
layers, 7 MCID (Malicious Call Identification), 399
updating IP Phones, BAT (Bulk configuring, 399–403
Administration Tool), 134– 135 multiple calls per line appearances,
upgrades, Cisco upgrades, release of, 46 configuring, 355–357
upgrading Other Group Call Pickup, 361
CallManager, 45–47 precedence, 403–404
operating systems, 461 preemption, 403–404
upper-end IP Phones, 56–59 Privacy, 367–368
user agents, SIP (Session Initiation redial, 347
Protocol), 180 softkey templates, 351–352
user database, CallManager, 101–102 adding, 352
user device profiles, CallManager, Extension assigning, 354
Mobility, 383–386 deleting, 355
user features, 347, 377 modifying, 353–354
abbreviating dialing, 348 speed dialing, 348
transfers, 347
862 user groups
user groups, MLA (Multilevel video codecs, CallManager, 623–624

Administration), 490–491 video modes. VT (Video Telephony)
creating, 492–499 Advantage, 652– 653
user levels, CAR (Call Detail Record {CDR} video protocols, 624–628
Analysis and Reporting) tool, 740 video telephony (IP), 619
User Name field (hunt group 7985 IP Phone, 620
configuration), 422 bandwidth
User Options welcome screen, CallManager, calculating, 630–631
106–107 call admission control, 631–632
user reports, CAR (Call Detail Record {CDR} management, 628–632
Analysis and Reporting) tool, 740 requirements, 629–630
configuring, 756–758 call admission control, 637–638
users clusters, 632–643
CallManager location configuration, 635–637
adding, 102–104 region configuration, 634–635
call forwarding, 107–108 Retry Video Call as Audio setting,
configuration, 102–114 638–640
Fast Dial service, 111–112 CallManager, 619
IP Phone services, 110–111 components, 619–621
locale settings, 113–114 H.320 gateways, 619
logon options, 105–106 H.323 gatekeepers, 620
Message Waiting Lamp page, 112–113 MCUs (Multipoint Control Units), 619
Personal Address Book service, protocols, 624–628
111– 112 SCCP endpoints, 620
speed dialing, 108–110 video calls, 621
User Options welcome screen, call flow, 622–623
106– 107 CallManager, 621–622
CallManager Attendant Console, adding, codec support, 623–624
416–417 VT (Video Telephony) Advantage, 620, 647
CAR (Call Detail Record {CDR} Analysis call routing, 654
and Reporting) tool, 739 calls, 651–652
client installation, 661–667
components, 647–648
V configuring, 653–654
configuring IP phones for, 657–661
validation, IP Phones, BAT (Bulk
Administration Tool), 131– 132 deployment tools, 655–657
verification protocols, 649–650
dual VLANs supported standards, 648–649
CatOS, 153 video modes, 652–653
NativeOS, 153–154 View menu (Attendant Console), 427
PoE (Power over Ethernet), Catalyst viewing pane, RTMT (Real-Time Monitoring
switches, 151 Tool), 704
video calls, 621 VLANs (virtual LANs), IP Phones,
CallManager, 621–623 configuring for, 65
configuring for, 653–654 voice gateways, 161
codec support, 623–624 analog gateways, 161–162
video call flow, 622 CallManager
call survivability, 163
redundancy support, 162
Wireless IP Phone 7920 863
communications, 163–164 hardware requirements, 661–662

configuring, 164–170, 172–174 installing, 663–665
core requirements, 162–163 IP phones, configuring for, 657–661
digital gateways, 161–162 IP video telephony, 620
DTMF (dual tone multifrequency) relay protocols, 649–650
capabilities, 162 software requirements, 661–662
H.323 gateways, 163 supported standards, 648–649
configuring, 165–169 video modes, 652–653
MGCP gateways, 163
configuring, 169–170, 172
Non-IOS MGCP gateways, configuring,
172–174
W
wall power, Catalyst switches, 146
SCCP gateways, 164 WAN Performance Utility (WPU), 764
supplementary services support, 162 WANs (wide area networks), CallManager
Voice Log Translator (CallManager), 732 clustering, 30–31
voice mail Web Access setting (IP Phone), 526
Immediate Divert, configuring, 358–359 disabling, 527–528
toll fraud, preventing, 505–507 web servers, PKI (Public Key Infrastructure),
voice VLANs, Catalyst switches, 152–154 certificate exchange, 564–567
voice-capable Catalyst switches, 145 WebDialer, CallManager, 43
Auxiliary VLAN support, 145 wildcards, route patterns, 197–199
CoS (class of service), configuring, Windows 2000 Server Operating System, 10
155–156 administrator password policies, 472–475
CoS marking, 146 antivirus protection, 467–468
dual VLANs, configuring, 153–154 common exploits, 475–476
inline power, 145 CSA (Cisco Security Agent), 468–472
IP Phones headless agent, 469
data VLANs, 152–154 managed agent, 469–470
midspan power injection, 146 protection rules, 471
PoE (Power over Ethernet), 146–147 supported applications, 470–471
powering, 146–151 hardening, 463–467
voice VLANs, 152–154 file sharing, 467
wall power, 146 IP Telphony Operating System scripts,
IP telephony, role in, 145–146 464–466
PoE (Power over Ethernet), 146–148 hot fix policy, 461–462
Catalyst EtherSwitch modules, 149 security, 459
Catalyst modular switching, 148 security patches, 461–462
Catalyst stackable switching, 149 security settings, 476–478
configuring, 150 threats, 459–461
device detection, 147–148 Windows Internet Naming Service
verifying, 151 (WINS), 37
VT (Video Telephony) Advantage, 59, 647 Windows Task Manager, Perfomance
CallManager Monitor, 702
call routing, 654 WINS (Windows Internet Naming
configuring, 653–654 Service), 37
deployment tools, 655–657 Wireless IP Phone 7920, 61
calls, 651–652
clients, installing on, 661–667
components, 647–648
864 wizards
wizards, BAT (Bulk Administration Tool)

device selection, 123–124
device specifics, 124–125
worksheets, configuration data,
CallManager, 38
WPU (WAN Performance Utility), 764
WS-SVC-CMM conference bridge type, 312
WS-X6608-E1 conference bridge type, 312
WS-X6608-T1 conference bridge type, 312
X-Z
X.509v3 certificates, PKI (Public Key
Infrastructure), 557–558
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Cisco IP Telephony CIPT

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Authorized Self-Study Guide

Cisco IP Telephony (CIPT)

Cisco IP Telephony (CIPT) (Authorized Self-Study Guide), Second Edition

Warning and Disclaimer

Corporate and Government Sales

About the Author

About the Technical Reviewers

Chapter 1 Introduction to Cisco Unified Communications and Cisco Unified

Chapter 2 Cisco Unified CallManager Clustering and Deployment Options 17

Chapter 3 Cisco Unified CallManager Installation and Upgrades 35

Chapter 4 Cisco IP Phones and Other User Devices 53

Chapter 7 Cisco Bulk Administration Tool 119

Chapter 8 Cisco Catalyst Switches 145

Chapter 9 Configuring Cisco Gateways and Trunks 161

Chapter 10 Cisco Unified CallManager Route Plan Basics 187

Chapter 11 Cisco Unified CallManager Advanced Route Plans 209

Chapter 12 Configuring Hunt Groups and Call Coverage 235

Chapter 13 Implementing Telephony Call Restrictions and Control 253

Chapter 14 Implementing Multiple-Site Deployments 275

Chapter 15 Media Resources 309

Chapter 16 Configuring User Features, Part 1 347

Chapter 17 Configuring User Features, Part 2 377

Chapter 18 Configuring Cisco Unified CallManager Attendant Console 409

Chapter 19 Configuring Cisco IP Manager Assistant 437

Part V: IPT Security 457

Chapter 20 Securing the Windows Operating System 459

Chapter 21 Securing Cisco Unified CallManager Administration 483

Chapter 22 Preventing Toll Fraud 503

Chapter 23 Hardening the IP Phone 523

Chapter 24 Understanding Cryptographic Fundamentals 535

Chapter 25 Understanding the Public Key Infrastructure 551

Chapter 26 Understanding Cisco IP Telephony Authentication and Encryption

Chapter 27 Configuring Cisco IP Telephony Authentication and Encryption 597

Chapter 28 Introducing IP Video Telephony 619

Chapter 29 Configuring Cisco VT Advantage 647

Chapter 30 Introducing Database Tools and Cisco Unified CallManager

Chapter 31 Monitoring Performance 695

Appendix A Answers to Review Questions 791

Part II: IPT Devices and Users 51

Chapter 7 Cisco Bulk Administration Tool 119

Cisco Trunk Concepts 174

Chapter 12 Configuring Hunt Groups and Call Coverage 235

Part IV: VoIP Features 307

Call Join 358

Multilevel Precedence and Preemption 403

Part V: IPT Security 457

Restricting External Transfers 511

PKI Revocation and Key Storage 561

Chapter 27 Configuring Cisco IP Telephony Authentication and Encryption 597

Chapter 29 Configuring Cisco VT Advantage 647

Chapter 31 Monitoring Performance 695

Initial Configuration 741

Icons Used in This Book

Directory Terminal File Web Ciscoworks ATM Modem

Camera Laptop Cisco IP IBM Front End Cluster Multilayer

Headquarters Cisco Cisco IP Network Cloud Ethernet Serial Switched

Command Syntax Conventions

Goals and Methods

Who Should Read This Book?