SCCP and SIP SRST Admin Guide PDF

Cisco Unified SCCP and SIP SRST
System Administrator Guide

(All Versions)
November 30, 2018
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Cisco Unified SCCP and SIP SRST System Administrator Guide (All Versions)
© 2018 Cisco Systems, Inc. All rights reserved.
.
CONTENTS
Preface i
Audience ii-i
Conventions ii-i
Obtain Documentation and Submit a Service Request ii-ii
Cisco Unified Survivable Remote Site Telephony Feature Roadmap iii-iii
Contents iii-iii
Documentation Organization iii-iv
Feature Roadmap iii-v
Information About New Features in Cisco Unified SRST iii-xi
New Features for Unified SRST Version 12.3 iii-xi
New Features for Unified SRST Version 12.2 iii-xi
New Features for Unified SRST Version 12.1 iii-xii
New Feature for Unified SRST Version 12.0 iii-xii
New Features for Cisco Unified SRST Version 11.0 iii-xii
New Features for Cisco Unified SRST Version 10.5 iii-xii

Support for Cisco Unified DX650 SIP IP Phones iii-xii
Support for Cisco Unified 78xx SIP IP Phones iii-xiii
New Features in Cisco Unified SRST Version 10.0 iii-xiii
Cisco Jabber for Windows iii-xiii
Version Negotiation for Cisco Unified SIP IP Phones iii-xiv
New Features in Cisco Unified SRST Version 9.5 iii-xiv
After-hour Pattern Blocking Support for Regular Expressions iii-xiv
Call Park Recall Enhancement iii-xv
Park Monitor iii-xv
Display Support for Name of Called Voice Hunt Groups iii-xvi
Preventing Local-Call Forwarding to Final Agent in Voice Hunt Groups iii-xvii
Trunk-to-Trunk Transfer Blocking for Toll Fraud Prevention on Cisco Unified SIP IP
Phones iii-xvii
transfer-pattern iii-xviii
Backward Compatibility iii-xviii
Dial Plans iii-xix
transfer max-length iii-xix
transfer-pattern blocked iii-xix
Cisco Unified SIP SRST System Administrator Guide

i
Contents
conference-pattern blocked iii-xx

Configuring the Maximum Number of Digits for a Conference Call iii-xxi
Prerequisites iii-xxi
DETAILED STEPS iii-xxii
Configuring Conference Blocking Options for Phones iii-xxii
Prerequisites iii-xxiii
transfer-pattern blocked iii-xxiv
conference transfer-pattern iii-xxv
New Features in Cisco Unified SRST Version 9.1 iii-xxv
KEM Support for Cisco Unified SIP IP Phones iii-xxvi
Enhancement in Speed-Dial Support iii-xxvi
Voice Hunt Group Support iii-xxvi
New Features in Cisco Unified SRST Version 9.0 iii-xxvii
Support for Cisco Unified 6901 and 6911 SIP IP Phones iii-xxvii
Support for Cisco Unified 6921, 6941, 6945, and 6961 SIP IP Phones iii-xxviii
Support for Cisco Unified 8941 and 8945 SIP IP Phones iii-xxviii
Multiple Calls Per Line iii-xxviii
Cisco Unified 8941 and 8945 SCCP IP Phones iii-xxviii
Cisco Unified 6921, 6941, 6945, 6961, 8941, and 8945 SIP IP Phones iii-xxviii
Voice and Fax Support on Cisco ATA-187 iii-xxix
New Features in Cisco Unified SRST Version 8.8 iii-xxix
Support for Cisco Unified 6945, 8941, and 8945 SCCP IP Phones iii-xxix
New Features in Cisco Unified SRST Version 8.0 iii-xxix
New Features in Cisco Unified SRST Version 7.0/4.3 iii-xxix

New Features in Cisco Unified SRST Version 4.2(1) iii-xxx
New Features in Cisco Unified SRST Version 4.1 iii-xxx

New Features in Cisco Unified SRST Version 4.0 iii-xxx
Additional Cisco Unified IP Phone Support iii-xxx
Cisco IP Communicator Support iii-xxxi
Fax Pass-through using SCCP and ATAs Support iii-xxxi
H.323 VoIP Call Preservation Enhancements for WAN Link Failures for SCCP
Phones iii-xxxi
Video Support iii-xxxi
New Features in Cisco Unified SRST Version 3.4 iii-xxxii
Cisco SIP SRST 3.4 iii-xxxii
New Features in Cisco SRST Version 3.3 iii-xxxii
Secure SRST iii-xxxii
Cisco Unified IP Phone 7970G and Cisco Unified 7971G-GE Support iii-xxxiii
Enhancement to the show ephone Command iii-xxxiii

ii
Contents
New Features in Cisco SRST Version 3.2 iii-xxxiii

Enhancement to the alias Command iii-xxxiii
Enhancement to the cor Command iii-xxxiv
Enhancement to the pickup Command iii-xxxiv
Enhancement to the user-locale Command iii-xxxiv
Increased the Number of Cisco Unified IP Phones Supported on the Cisco 3845 iii-xxxiv
MOH Live-Feed Support iii-xxxiv
No Timeout for Call Preservation iii-xxxv
RFC 2833 DTMF Relay Support iii-xxxv
Translation Profile Support iii-xxxv
New Features in Cisco SRST Version 3.1 iii-xxxv
Cisco Unified IP Phone 7920 Support iii-xxxvi
Cisco Unified IP Phone 7936 Support iii-xxxvi
New Features in Cisco SRST Version 3.0 iii-xxxvi

Additional Language Options for IP Phone Display iii-xxxvi
Consultative Call Transfer and Forward Using H.450.2 and H.450.3 for SCCP
Phones iii-xxxvii
Customized System Message for Cisco Unified IP Phones iii-xxxvii
Dual-Line Mode iii-xxxvii
E1 R2 Signaling Support iii-xxxviii
European Date Formats iii-xxxix
Huntstop for Dual-Line Mode iii-xxxix
Music-on-Hold for Multicast from Flash Files iii-xxxix
Ringing Timeout Default iii-xxxix
Secondary Dial Tone iii-xxxix
Enhancement to the show ephone Command iii-xl
System Log Messages for Phone Registrations iii-xl
Three-Party G.711 Ad Hoc Conferencing iii-xl
Support for Cisco VG248 Analog Phone Gateway 1.2(1) and Higher Versions iii-xl
New Features in Cisco SRST Version 2.1 iii-xl

Additional Language Options for IP Phone Display iii-xlii
Cisco SRST Aggregation iii-xlii
Cisco ATA 186 and ATA 188 Support iii-xlii
Cisco Unified IP Phone 7902G Support iii-xlii
Cisco Unified IP Phone 7905G Support iii-xliii
Cisco Unified IP Phone 7912G Support iii-xliii
Cisco Unified IP Phone Expansion Module 7914 Support iii-xliii
Enhancement to the dialplan-pattern Command iii-xliii
New Features in Cisco SRST Version 2.02 iii-xliv

iii
Contents
Cisco Unified IP Phone Conference Station 7935 Support iii-xliv

Increase in Directory Numbers iii-xliv
Cisco Unity Voicemail Integration Using In-Band DTMF Signaling Across the PSTN and
BRI/PRI iii-xlv
Where to Go Next iii-xlv
Cisco Unified SRST Feature Overview 1-1
Contents 1-1
Cisco Unified SCCP SRST 1-1

Information About SCCP SRST 1-2
Prerequisites for Configuring Cisco Unified SCCP SRST 1-4
Installing Cisco Unified Communications Manager 1-5
Installing Cisco Unified SCCP SRST 1-5
Integrating Cisco Unified SCCP SRST with Cisco Unified
Communications Manager 1-5
Restrictions for Configuring Cisco Unified SCCP SRST 1-7
Cisco Unified SIP SRST 1-9
Information About SIP SRST 1-9
Prerequisites for Configuring Cisco Unified SIP SRST 1-9
Restrictions for Configuring Cisco Unified SIP SRST 1-9
Cisco Unified SRST Licenses 1-12
Cisco Unified SRST Permanent License 1-13
Collaboration Professional Suite License 1-13
Cisco Smart License 1-14
Configure Call Home 1-16
Licensing Modes 1-17
Restrictions 1-17
Interface Support for Unified CME and Unified SRST 1-17
MGCP Gateways and SRST 1-17
IPv6 Support for Unified SRST SIP IP Phones 1-18

Feature Support for IPv6 in Unified SRST SIP IP Phones 1-18
Restrictions 1-19
Configure IPv6 Pools for SIP IP Phones 1-19
Examples for Configuring IPv6 Pools for SIP IP Phones 1-24
Support for Cisco Unified IP Phones and Platforms 1-25
Finding Cisco IOS Software Releases That Support Cisco Unified SRST 1-25
Cisco Unified IP Phone Support 1-26
Platform and Memory Support 1-26
Determining Platform Support Through Cisco Feature Navigator 1-26

iv
Contents
Availability of Cisco IOS Software Images 1-26

Cisco Unified Communications Manager Compatibility 1-27
Signal Support 1-27
Language Support 1-27
Switch Support 1-27
Multicast Music On Hold 1-27
Configure Multicast Music On Hold for Unified SRST 1-28
Where to Go Next 1-30
Additional References 1-30

Related Documents 1-31
Standards 1-33
MIBs 1-33
RFCs 1-33
Technical Assistance 1-33
Obtaining Documentation, Obtaining Support, and Security Guidelines 1-33
Cisco Unified SIP SRST on Cisco 4000 Series Integrated Services Router 2-35
Contents 2-35
Overview 2-36
Platform and Memory Support 2-36
Cisco IOS Software Releases that Support Unified SRST 2-36

Install Cisco IOS Software 2-36
Feature Support 2-38
Restrictions of Unified SRST on Cisco 4000 Series Integrated Services Routers 2-39
Unified IP Phone Support 2-39
Cisco Unified Communications Manager Compatibility 2-39
Installing Cisco Unified Communications Manager 2-39
Integrating Cisco Unified SIP SRST with Cisco Unified Communications Manager 2-40
Supported PSTN Trunk Connectivity 2-40
Language Support 2-40
Switch Support 2-40
Interface Support for Unified SRST 2-41
Licensing 2-41
Cisco Unified SRST Permanent License 2-41
Collaboration Professional Suite License 2-42
Cisco Smart License 2-42
Configure Call Home 2-44
Licensing Modes 2-45
Restrictions 2-46

v
Contents
Configure SIP Registrar Functionality for SIP Phones on Unified SRST 2-46
Configure Back up Registrar Service to SIP Phones 2-47
Prerequisites 2-48
Restrictions 2-48
Configure Backup Registrar Service to SIP Phones (Using Optional Commands) 2-50
Prerequisites 2-51
Verify SIP Registrar Configuration 2-54
Verify Proxy Dial-Peer Configuration 2-56
IPv6 Support for Unified SRST SIP IP Phones 2-59
Feature Support for IPv6 in Unified SRST SIP IP Phones 2-60
Restrictions 2-60
Configure IPv6 Pools for SIP IP Phones 2-60
Configure Unified SRST on Cisco 4000 Series Integrated Services Platform 2-65
Configure Voice Hunt Groups on Unified SRST 2-69

Prerequisites 2-69
Configure Feature Support on Unified SIP SRST 2-72
Configure SIP-to-SIP Call Forwarding 2-72
Configure Call Blocking Based on Time of Day, Day of Week, or Date 2-74
Verification 2-77
SIP Call Hold and Resume 2-78
Configure Music On Hold for Unified SRST 2-78
Enabling KPML for SIP Phones 2-79
Restrictions 2-79
Disabling SIP Supplementary Services for Call Forward and Call Transfer 2-80
Configuring Idle Prompt Status for SIP Phones 2-81
Examples 2-83
Example for Configuring Unified SIP SRST on Cisco 4000 Series Integrated Services
Routers 2-83
Example for Configuring Voice Hunt Groups in Unified SIP SRST 2-83
Examples for Configuring IPv6 Pools for SIP IP Phones 2-84
Example for Configuring Call Blocking Based on Time of Day, Day of Week, or
Date 2-85
Example for Configuring Music On Hold for Unified SIP SRST 2-85
Example for Configuring SIP-to-SIP Call Forwarding on Unified SRST 2-85
Example for Configuring Idle Prompt Status for SIP Phones 2-85
Example for Disabling SIP Supplementary Services for Call Forward and Call
Transfer 2-86
Cisco Unified Enhanced Survivable Remote Site Telephony 3-87
Contents 3-87

vi
Contents
Migration from Unified SRST Manager to Unified E-SRST 3-87

Benefits of Unified-ESRST 3-88
Unified E-SRST with Support for Voice Hunt Group 3-88
Support for B-ACD in Unified E-SRST 3-89
Recommendations for Configuring Voice Hunt Group on Unified E-SRST 3-89
Restrictions for Unified E-SRST, Release 12.2 3-90
SIP: Configure Unified E-SRST 3-90
Restrictions 3-91
Enable the E-SRST Mode 3-91
Configure SIP shared-line 3-92
Configure BLF 3-93
Enable a SIP Directory Number to be Watched 3-93
Enable BLF on a Voice Register Pool 3-93
Configure Unified E-SRST 3-94
DETAILED STEPS 3-96
Configure Voice Hunt Groups on Unified E-SRST 3-99
DETAILED STEPS 3-100
Example for configuring Unified E- SRST with Voice Hunt Group
Enhancements 3-101
Example for Configuring B-ACD with Unified E-SRST 3-102
Example for Configuring Shared Line with Voice Hunt Group on Unified
E-SRST 3-103
SCCP: Configure Unified E-SRST 3-106
Prerequisites 3-106
Configure Mixed Shared Lines with SCCP Phones 3-110
Configure BLF for SCCP Phones 3-111
Enable an SCCP Directory Number to be Watched 3-111
Enable BLF on an Ephone 3-112
Configure Digest Credentials On Unified Communications Manager 3-112
Configure Digest Credentials on Unified E-SRST for SIP 3-112
Configure Digest Credentials on Unified E-SRST for SCCP 3-113
Unified E-SRST Scale Support 3-113
Setting Up the Network 4-115
Contents 4-115
Information About Setting Up the Network 4-116
How to Set Up the Network 4-116

Enabling IP Routing 4-116
Enabling Cisco Unified SRST on an MGCP Gateway 4-116

vii
Contents
Configuring Cisco Unified SRST on an MGCP Gateway Prior to Cisco IOS Release
12.3(14)T 4-117
Configuring SRST on an MGCP Gateway Using Cisco IOS Release 12.3(14)T or Later
Releases 4-118
Restrictions 4-118
Configuration Example of Enabling SRST on a MGCP Gateway using Cisco IOS
Release 12.3(14)T 4-120
Configuring DHCP for Cisco Unified SRST Phones 4-122
Defining a Single DHCP IP Address Pool 4-122
Defining a Separate DHCP IP Address Pool for Each Cisco Unified IP Phone 4-123
Defining the DHCP Relay Server 4-124
Specifying Keepalive Intervals 4-125
Examples 4-126
Cisco Unified SIP SRST 4.1 5-127
Contents 5-127
Prerequisites for Cisco Unified SIP SRST 4.1 5-127
Restrictions for Cisco Unified SIP SRST 4.1 5-128
Information About Cisco Unified SIP SRST 4.1 5-128

Out-of-Dialog REFER 5-128
Digit Collection on SIP Phones 5-129
KPML Digit Collection 5-129
SIP Dial Plans 5-129
Caller ID Display 5-130
Idle Prompt Status 5-130
Enhanced 911 Services 5-130
How to Configure Cisco Unified SIP SRST 4.1 Features 5-131
Enabling KPML for SIP Phones 5-131
Restrictions 5-131
What to Do Next 5-132
Configuring Idle Prompt Status for SIP Phones 5-134
Prerequisites 5-134
Setting Up Cisco Unified IP Phones using SCCP 6-137
Contents 6-137
Information About Setting Up Cisco Unified IP Phones 6-137

viii
Contents
How to Set Up Cisco Unified IP Phones 6-138

Configuring Cisco Unified SRST to Support Phone Functions 6-138
Configuring Cisco Unified 8941 and 8945 SCCP IP Phones 6-140
Verifying That Cisco Unified SRST Is Enabled 6-141
Configuring IP Phone Clock, Date, and Time Formats 6-142
Example 6-143
Configuring IP Phone Language Display 6-144
Examples 6-145
Configuring Customized System Messages for Cisco Unified IP Phones 6-146
Examples 6-147
Configuring a Secondary Dial Tone 6-147
Examples 6-147
Configuring Dual-Line Phones 6-148
Examples 6-149
Configuring Eight Calls per Button (Octo-Line) 6-150
Prerequisites 6-150
Restrictions 6-150
Examples 6-152
Configuring the Maximum Number of Calls 6-152
Prerequisites 6-152
Troubleshooting 6-154
How to Set Up Cisco IP Communicator for Cisco Unified SRST 6-154
Prerequisites 6-154
Verifying Cisco IP Communicator 6-155
Troubleshooting Cisco IP Communicator 6-155
Setting Up Cisco Unified IP Phones using SIP 7-157
Contents 7-157
Prerequisites for Configuring the SIP Registrar 7-157
Restrictions for Configuring the SIP Registrar 7-157
Information About Configuring the SIP Registrar 7-157
How to Configure the SIP Registrar 7-158

Configuring the SIP Registrar 7-158
Configuring Backup Registrar Service to SIP Phones 7-160
Prerequisites 7-161
Restrictions 7-161

ix
Contents
Configuring Backup Registrar Service to SIP Phones (Using Optional Commands) 7-164
Prerequisites 7-164
Examples 7-167
Verifying SIP Registrar Configuration 7-167
Verifying Proxy Dial-Peer Configuration 7-169
Configuring Call Handling 8-175
Contents 8-175
Prerequisites for Configuring SIP SRST Features Using Back-to-Back User Agent
Mode 8-176
Restrictions for Configuring SIP SRST Features Using Back-to-Back User Agent
Mode 8-176
Information About Configuring SCCP SRST Call Handling 8-176
H.323 VoIP Call Preservation Enhancements for WAN Link Failures 8-176
Toll Fraud Prevention 8-177
Information About Configuring SIP SRST Features Using Back-to-Back User Agent
Mode 8-177
Cisco Unified SIP SRST and Cisco SIP Communications Manager Express Feature
Crossover 8-177
How to Configure Cisco Unified SCCP SRST 8-179
Configuring Incoming Calls 8-180
Configuring Call Forwarding During a Busy Signal or No Answer 8-180
Examples 8-181
Configuring Call Rerouting 8-182
Examples 8-185
Configuring Call Pickup 8-185
Examples 8-186
Configuring Consultative Transfer 8-187
Conference Calls 8-188
Configuring Transfer Digit Collection Method 8-188
Examples 8-189
Configuring Global Prefixes 8-189
Examples 8-191
Enabling Digit Translation Rules 8-191
Examples 8-192
Enabling Translation Profiles 8-192
Examples 8-195
Verifying Translation Profiles 8-195
Configuring Dial-Peer and Channel Hunting 8-196

x
Contents
Examples 8-197
Configuring Busy Timeout 8-197
Examples 8-198
Configuring the Ringing Timeout Default 8-198
Examples 8-199
Configuring Outgoing Calls 8-199
Configuring Local and Remote Call Transfer 8-199
Examples 8-200
Enabling Consultative Call Transfer and Forward Using H.450.2 and H.450.3 with
Cisco SRST 3.0 8-200
Examples 8-203
Enabling Analog Transfer Using Hookflash and the H.450.2 Standard with Cisco
SRST 3.0 or Earlier 8-204
Examples 8-208
Configuring Trunk Access Codes 8-208
Examples 8-209
Configuring Interdigit Timeout Values 8-209
Examples 8-210
Configuring Class of Restriction 8-210
Examples 8-212
Call Blocking (Toll Bar) Based on Time of Day and Day of Week or Date 8-214
Examples 8-215
How to Configure Cisco Unified SIP SRST 8-216
Configuring SIP Phone Features 8-216
Configuring SIP-to-SIP Call Forwarding 8-218
Configuring Call Blocking Based on Time of Day, Day of Week, or Date 8-220
Examples 8-223
Verification 8-223
SIP Call Hold and Resume 8-224
Examples 8-224
How to Configure Optional Features 8-226
Enabling Three-Party G.711 Ad Hoc Conferencing 8-226
Examples 8-227
Defining XML API Schema 8-228
Configuration Examples for Call Handling 8-228
Example: Monitoring the Status of Key Expansion Modules 8-228
Example: Configuring Voice Hunt Groups in Cisco Unified SIP SRST 8-229

xi
Contents
Configuring Secure SRST for SCCP and SIP 9-231
Contents 9-231
Prerequisites for Configuring Secure SRST 9-231
Restrictions for Configuring Secure SRST 9-232
Information About Configuring Secure SRST 9-234

Benefits of Secure SRST 9-234
Secure SIP SRST Support on Cisco 4000 Series Integrated Services Router 9-234
Secure Music On Hold for Unified Secure SRST (SIP) 9-235
Secure SCCP SRST on Cisco 4000 Series Integrated Services Router 9-235
Secure SCCP SRST for Analog Voice Gateways 9-235
Secure Music On Hold for Secure Unified SRST (SCCP) 9-236
Three-way Software Conferencing for Secure SCCP, Unified SRST Release
12.3 9-236
Feature Support for Secure SRST (SCCP), Unified SRST Release 12.3 9-237
Cisco IP Phones Clear-Text Fallback During Non-Secure SRST 9-238
Signaling Security on Unified SRST - TLS 9-238
SRST Routers and the TLS Protocol 9-238
Certificates Operation on Secure SRST 9-240
Certificates Transport from CUCM to Secure SRST 9-242
Media Security on Unified SRST - SRTP 9-242
Establishment of Secure Cisco Unified SRST to the Cisco Unified IP Phone 9-243
Secure SRST Authentication and Encryption 9-244
How to Configure Secure Unified SRST 9-245
9-245
Preparing the Cisco Unified SRST Router for Secure Communication 9-246
Configuring a Certificate Authority Server on a Cisco IOS Certificate Server 9-246
Autoenrolling and Authenticating the Secure Cisco Unified SRST Router to the CA
Server 9-248
Disabling Automatic Certificate Enrollment 9-252
Verifying Certificate Enrollment 9-253
Enabling Credentials Service on the Secure Cisco Unified SRST Router 9-255
Troubleshooting Credential Settings 9-257
Importing Phone Certificate Files in PEM Format to the Secure SRST Router 9-257
Cisco Unified Communications Manager 4.X.X and Earlier Versions 9-258
Cisco Unified Communications Manager 5.0 and Later Versions 9-258
Cisco Unified Communications Manager 6.0 and Later Versions 9-259
Authenticating the Imported Certificates on the Cisco Unified SRST Router 9-259
Examples 9-260

xii
Contents
Configuring Cisco Unified Communications Manager to the Secure Cisco Unified SRST
Router 9-265
Adding an SRST Reference to Cisco Unified Communications Manager 9-265
Configuring SRST Fallback on Cisco Unified Communications Manager 9-266
Configuring CAPF on Cisco Unified Communications Manager 9-268
Enabling SRST Mode on the Secure Cisco Unified SRST Router 9-268
Examples 9-270
Configuring Secure SCCP SRST 9-270
Prerequisites for Configuring Secure SCCP SRST 9-270
Restrictions for Configuring Secure SCCP SRST 9-270
Verifying Phone Status and Registrations 9-271
Configuration Examples for Secure SCCP SRST 9-278
Configuring Secure SIP Call Signaling and SRTP Media with Cisco SRST 9-284
Prerequisites for Configuring Secure SIP Call Signaling and SRTP Media with Cisco
SRST 9-284
Restrictions for Configuring Secure SIP Call Signaling and SRTP Media with Cisco
SRST 9-284
Information About Cisco Unified SIP SRST Support of Secure SIP Signaling and
SRTP Media 9-285
Configuring Cisco Unified Communications Manager 9-285
9-285
9-285
Configuring Phones 9-286
Configuring SIP options for Secure SIP SRST 9-287
Configuring SIP SRST Security Policy 9-288
Configuring SIP User Agent for Secure SIP SRST 9-289
Configuration Example for Cisco Unified SIP SRST 9-293
Additional References 9-299
Related Documents 9-300
Standards 9-300
MIBs 9-300
RFCs 9-300
Technical Assistance 9-301
Command Reference 9-301
Feature Information for Secure SCCP and SIP SRST 9-302
Configuring SIP Trunking on Unified SRST 10-303
Contents 10-303
Unified SRST and Unified Border Element Co-location 10-303

xiii
Contents
Configuration Recommendations for Unified SRST and Unified Border Element

Co-location 10-305
Restrictions 10-307
Examples 10-308
Feature Information for Configuring SIP Trunking on Unified SRST 10-319
Integrating Voicemail with Cisco Unified SRST 11-321
Contents 11-321
Information About Integrating Voicemail with

Cisco Unified SRST 11-321
How to Integrate Voicemail with Cisco Unified SCCP and SIP SRST 11-323
Configuring Direct Access to Voicemail 11-323
Examples 11-326
Configuring Message Buttons 11-326
Examples 11-328
Redirecting to Cisco Unified Communications Manager Gateway 11-329
Configuring Call Forwarding to Voicemail 11-329
Call Routing Instructions Using DTMF Digit Patterns 11-329
Prerequisites 11-331
Examples 11-333
Configuring Message Waiting Indication (Cisco Unified SRST Routers) 11-333
Configuring Message Waiting Indication (SIP Phones in SRST Mode) 11-335
Configuration Examples for Unified SRST 11-337
Configuring Local Voicemail System (FXO and FXS): Example 11-337
Configuring Central Location Voicemail System (FXO and FXS): Example 11-338
Configuring Voicemail Access over FXO and FXS: Example 11-338
Configuring Voicemail Access over BRI and PRI: Example 11-339
Message Waiting Indication for SIP SRST: Example 11-340
How to Configure DTMF Relay for SIP Applications and Voicemail 11-340
DTMF Relay Using SIP RFC 2833 11-340
Troubleshooting Tips 11-342
DTMF Relay Using SIP Notify (Nonstandard) 11-342
Troubleshooting Tips 11-343
Setting Video Parameters 12-345
Contents 12-345
Prerequisites for Setting Video Parameters 12-345
Restrictions for Setting Video Parameters 12-346
Information About Setting Video Parameters 12-346

xiv
Contents
Matching Endpoint Capabilities 12-347

Retrieving Video Codec Information 12-347
Call Fallback to Audio-Only 12-347
Call Setup for Video Endpoints 12-347
Call Setup Between Two Local SCCP Endpoints 12-348
Call Setup Between SCCP and H.323 Endpoints 12-348
Call Setup Between Two SCCP Endpoints Across an H.323 Network 12-348
Flow of the RTP Video Stream 12-348
How to Set Video Parameters for Cisco Unified SRST 12-349
Configuring Slow Connect Procedures 12-349
Verifying Cisco Unified SRST 12-350
Examples 12-351
Setting Video Parameters for Cisco Unified SRST 12-356
Examples 12-357
Troubleshooting Video for Cisco Unified SRST 12-358
Monitoring and Maintaining Cisco Unified SRST 13-359
Configuring Cisco Unified SIP SRST Features Using Redirect Mode A-1
Contents A-1
Prerequisites for Cisco Unified SIP SRST Features Using Redirect Mode A-1
Restrictions for Cisco Unified SIP SRST Features Using Redirect Mode A-1
Information About Cisco Unified SIP SRST Features Using Redirect Mode A-2
How to Configure Cisco Unified SIP SRST Features Using Redirect Mode A-2
Configuring Call Redirect Enhancements to Support Calls Between SIP IP Phones for
Cisco Unified SIP SRST A-2
Configuring Call Redirect Enhancements to Support Calls Globally A-3
Configuring Call Redirect Enhancements to Support Calls on a Specific VoIP Dial
Peer A-4
Configuring Sending 300 Multiple Choice Support A-5
Configuration Examples for Cisco Unified SIP SRST Features Using Redirect Mode A-6
Cisco Unified SIP SRST: Example A-7
Where to Go Next A-8
Integrating Cisco Unified Communications Manager and Cisco Unified SRST to Use Cisco
Unified SRST as a Multicast MOH Resource B-11
Contents B-11
Prerequisites for Using Cisco Unified SRST Gateways as a Multicast MOH Resource B-12
Restrictions for Using Cisco Unified SRST Gateways as a Multicast MOH Resource B-12

xv
Contents
Information About Using Cisco Unified SRST Gateways as a Multicast MOH Resource B-13
Cisco Unified SRST Gateways and Cisco Unified Communications Manager B-13
Codecs, Port Numbers, and IP Addresses B-14
Multicast MOH Transmission B-16
MOH from a Live Feed B-16
MOH from Flash Files B-17
How to Use Cisco Unified SRST Gateways as a Multicast MOH Resource B-18
Configuring Cisco Unified Communications Manager for Cisco Unified SRST Multicast
MOH B-18
Configuring the MOH Audio Source to Enable Multicasting B-19
Enabling Multicast on the Cisco Unified Communications Manager MOH Server and
Configuring Port Numbers and IP Addresses B-20
Creating an MRG and an MRGL, Enabling MOH Multicast, and Configuring
Gateways B-23
Creating a Region for the MOH Server B-25
Verifying Cisco Unified Communications Manager Multicast MOH B-26
Configuring Cisco Unified SRST for Multicast MOH from an Audio File B-26
Prerequisites B-27
Enabling Multicast MOH on the Cisco Unified SRST Gateway B-27
Verifying Basic Cisco Unified SRST Multicast MOH Streaming B-31
Verifying Cisco Unified SRST MOH to PSTN B-32
Verifying Cisco Unified SRST Multicast MOH to IP Phones B-36
Troubleshooting Tips B-36
Configuring Cisco Unified SRST for MOH from a Live Feed B-36
Prerequisites B-37
Restrictions B-37
Setting Up the Voice Port on the Cisco Unified SRST Gateway B-37
Setting Up the Directory Numbers on the Cisco Unified SRST Gateway B-39
Establishing the MOH Feed B-39
Verifying Cisco Unified SRST MOH Live Feed B-41
Configurations Examples for Cisco Unified SRST Gateways B-41
MOH Routed to Two IP Addresses: Example B-41
MOH Live Feed: Example B-42
Feature Information for Cisco Unified SRST as a Multicast MOH Resource B-42
Where to Go Next B-43

xvi
Preface
This preface describes the audience and conventions of the Cisco Unified SCCP and SIP SRST System
Administration Guide. It also describes the available product documentation and provides information
on how to obtain documentation and technical assistance.
• Audience, page i
• Conventions, page i
• Obtain Documentation and Submit a Service Request, page ii
Audience
This guide is intended primarily for network administrators and channel partners.
Conventions
This guide uses the following conventions:
Item Convention
Commands and keywords. boldface font
Variables for which you supply values. italic font
Optional command keywords. You do not [enclosed in brackets]
have to select any options.
Required command keyword to be selected {options enclosed in braces |
from a set of options. You must choose one separated by vertical bar}
option.
Displayed session and system information. screen font
Information you enter. boldface screen font
Variables you enter. italic screen font

Menu items and button names. boldface font
Choosing a menu item. Option > Network Preferences
Cisco Unified SCCP and SIP SRST System Administrator Guide

i
Obtain Documentation and Submit a Service Request
Note Means reader take note.
Tip Means the following information will help you solve a problem.
Caution Means reader be careful. In this situation, you might perform an action that could result in equipment
damage or loss of data.
Timesaver Means the described action saves time. You can save time by performing the action described in
the paragraph.
Warning Means reader be warned. In this situation, you might perform an action that could result in
bodily injury.
Obtain Documentation and Submit a Service Request

For information on obtaining documentation, using the Cisco Bug Search Tool (BST), submitting a service request, and
gathering additional information, see What’s New in Cisco Product Documentation.
To receive new and revised Cisco technical content directly to your desktop, you can subscribe to the What’s New in
Cisco Product Documentation RSS feed. The RSS feeds are a free service.

ii
CHAPTER iii
Cisco Unified Survivable Remote Site
Telephony Feature Roadmap
This chapter contains a list of Cisco Unified Survivable Remote Site Telephony (Cisco Unified SRST)
features and the location of feature documentation.
Use Cisco Feature Navigator to find information about platform support and Cisco IOS software image
support. Access Cisco Feature Navigator at http://www.cisco.com/go/fn. You must have an account on
Cisco.com. If you do not have an account or have forgotten your username or password, click Cancel at
the login dialog box and follow the instructions that appear.
Contents
• Documentation Organization, page iv
• Feature Roadmap, page v
• Information About New Features in Cisco Unified SRST, page xi
• Where to Go Next, page xlv

iii-iii
Chapter iii Cisco Unified Su rvivable Remote Site Telephony Feature Roadmap
Documentation Organization
Documentation Organization
This document consists of the following chapters or appendixes as shown in Table iii-1.
Table iii-1 Cisco Unified SRST Configuration Sequence
Chapter or Appendix Description

Cisco Unified SRST Feature Gives a brief description of Cisco Unified SRST and provides information on the
Overview, page 1 supported platforms and Cisco Unified IP Phones. In addition, it describes any
prerequisites or restrictions that should be addressed before Cisco Unified SIP SRST
is configured.
Setting Up the Network, page 115 Describes how to set up a Cisco Unified SRST system to communicate with your
network.
Cisco Unified Enhanced Survivable Describes how to configure the Cisco Unified Enhanced SRST feature in your network.
Remote Site Telephony, page 87
Cisco Unified SIP SRST 4.1, Describes the features for Cisco Unified SIP SRST Version 4.1 and provides the
page 127 associated configuration procedures.
Setting Up Cisco Unified IP Phones Describes how to set up the basic Cisco Unified SRST phone configuration.
using SCCP, page 137
Setting Up Cisco Unified IP Phones Describes features available in Version 3.0 that are also necessary for Version 3.4.
using SIP, page 157 Features include instructions on how to provide a backup to an external SIP call control
(IP-PBX) by providing basic registrar services. These services are used by a SIP IP
phone in the event of a WAN connection outage when the SIP phone is unable to
communicate with its primary SIP proxy.
Configuring Call Handling, Describes how to configure incoming and outgoing calls.
page 175
Configuring Secure SRST for SCCP Describes the Secure SRST security functionality to the Cisco Unified SRST.
and SIP, page 231
Integrating Voicemail with Cisco Describes how to set up voicemail.
Unified SRST, page 321
Setting Video Parameters, page 345 Describes how to set up video parameters.
Monitoring and Maintaining Cisco Provides a list of useful show commands for monitoring and maintaining Cisco Unified
Unified SRST, page 359 SRST.
Configuring Cisco Unified SIP Describes features using redirect mode, which applies to version 3.0 only.
SRST Features Using Redirect
Mode, page 1
Integrating Cisco Unified Describes how to configure Cisco Unified CM and Cisco Unified SRST to enable
Communications Manager and multicast music-on-hold (MOH).
Cisco Unified SRST to Use Cisco
Unified SRST as a Multicast MOH
Resource, page 11

iii-iv
Feature Roadmap
Feature Roadmap
Table iii-2 provides a feature history summary of Cisco Unified SRST features.
Table iii-2 Features by Cisco Unified SRST Software Version
Cisco Unified Cisco IOS

SRST Release Enhancements or Modifications
Version 12.3 Cisco IOS XE • Secure SCCP SRST on Cisco 4000 Series Integrated Services Router,
Fuji 16.9.1 page 235
Version 12.2 Cisco IOS XE • Unified E-SRST with Support for Voice Hunt Group, page 88
Fuji 16.8.1
Version 12.1 Cisco IOS XE • Cisco Smart License, page 14
Fuji 16.7.1
• Secure SIP SRST Support on Cisco 4000 Series Integrated Services Router,
page 234
• Unified SRST and Unified Border Element Co-location, page 303
Version 12.0 Cisco IOS XE • IPv6 Support for Unified SRST SIP IP Phones, page 18
Everest 16.6.1
Version 11.0 15.6(1)T • Support for Cisco IP Phone 7811
• Support for Cisco IP Phones 8811, 8831, 8841, 8845, 8865, 8851, 8851NR,
8861
• Support for Cisco ATA-190 Phones
Version 10.5 15.4(3)M • Unified E-SRST Scale Support, page 113
• Support for Cisco Unified DX650 SIP IP Phones, page xii
• Support for Cisco Unified 78xx SIP IP Phones, page xiii
• Support for Cisco IP Phones 88xx, 8941, 8945, and 8961
Version 10.0 15.3(3)M • Cisco Jabber for Windows, page xiii
• SIP: Configure Unified E-SRST, page 90
• To obtain an account on Cisco.com, go to www.cisco.com and click Register
at the top of the screen., page 4
Version 9.5 15.3(2)T • After-hour Pattern Blocking Support for Regular Expressions, page xiv
• Call Park Recall Enhancement, page xv
• Display Support for Name of Called Voice Hunt Groups, page xvi
• Preventing Local-Call Forwarding to Final Agent in Voice Hunt Groups,
page xvii
• Trunk-to-Trunk Transfer Blocking for Toll Fraud Prevention on Cisco
Unified SIP IP Phones, page xvii
Version 9.1 15.2(4)M • KEM Support for Cisco Unified SIP IP Phones, page xxvi
• Enhancement in Speed-Dial Support, page xxvi
• Voice Hunt Group Support, page xxvi

iii-v
Feature Roadmap
Table iii-2 Features by Cisco Unified SRST Software Version (continued)

Version 9.0 15.2(2)T • Support for Cisco Unified 6901 and 6911 SIP IP Phones, page xxvii
• Support for Cisco Unified 6921, 6941, 6945, and 6961 SIP IP Phones,
page xxviii
• Support for Cisco Unified 8941 and 8945 SIP IP Phones, page xxviii
• Multiple Calls Per Line, page xxviii
• Voice and Fax Support on Cisco ATA-187, page xxix
Version 8.8 15.2(1)T Support for Cisco Unified 6945, 8941, and 8945 SCCP IP Phones, page xxix
Version 8.6 15.1(4)M Support for Cisco Unified 8941 and 8945 SCCP IP Phones were introduced. For
more information, see Configuring Cisco Unified 8941 and 8945 SCCP IP
Phones, page 140.
Version 8.0 15.1(1)T Beginning with Cisco IP Phone firmware 8.5(3) and Cisco IOS Release 15.1(1)T,
Cisco SRST supports SIP signaling over UDP, TCP, and TLS connections,
providing both RTP and SRTP media connections based on the security settings
of the IP phone. For more information, see the following sections:
• Signaling Security on Unified SRST - TLS, page 238
• Media Security on Unified SRST - SRTP, page 242
• Configuring Secure SIP Call Signaling and SRTP Media with Cisco SRST,
page 284
Version 7.0/4.3 See Cisco Feature • Configuring Eight Calls per Button (Octo-Line), page 150
Navigator for
• Configuring Consultative Transfer, page 187
compatibility.
Version 4.2(1) See Cisco Feature Enhanced 911 Services, page 130
Navigator for The following new features are included:
compatibility.
• Assigning ERLs to zones to enable routing to the PSAP that is closest to the
caller.
• Customizing E911 by defining a default ELIN, identifying a designated
number if the 911 caller cannot be reached on callback, specifying the expiry
time for data in the Last Caller table, and enabling syslog messages that
announce all emergency calls.
• Expanding the E911 location information to include name and address.
• Adding new permanent call detail records.
Version 4.1 12.4(15)T • Enabling KPML for SIP Phones, page 131
• Disabling SIP Supplementary Services for Call Forward and Call Transfer,
page 130
• Configuring Idle Prompt Status for SIP Phones, page 134
• Enhanced 911 Services, page 130

iii-vi
Feature Roadmap

Version 4.0 12.4(4)XC • Cisco IP Communicator Support, page xxxi
• Fax Pass-through using SCCP and ATAs Support, page xxxi
• H.323 VoIP Call Preservation Enhancements for WAN Link Failures for
SCCP Phones, page xxxi
• Video Support, page xxxi
Version 3.4 12.4(4)T • Cisco SIP SRST 3.4, page xxxii
• Configuring Cisco Unified SIP SRST Features Using Redirect Mode, page 1
• Configuring Call Handling, page 175 (see Back-to-Back User Agent Mode)
Version 3.3 • Secure SRST, page xxxii.
• Cisco Unified IP Phone 7970G and Cisco Unified 7971G-GE Support,
page xxxiii
• Enhancement to the show ephone Command, page xxxiii
Version 3.2 12.3(11)T • Enhancement to the alias Command, page xxxiii
• Enhancement to the pickup Command, page xxxiv
• Enhancement to the user-locale Command, page xxxiv
• Increased the Number of Cisco Unified IP Phones Supported on the Cisco
3845, page xxxiv
• MOH Live-Feed Support, page xxxiv
• No Timeout for Call Preservation, page xxxv
• RFC 2833 DTMF Relay Support, page xxxv
• Translation Profile Support, page xxxv
Version 3.1 12.3(7)T • Cisco Unified IP Phone 7920 Support, page xxxvi
• Cisco Unified IP Phone 7936 Support, page xxxvi

iii-vii
Feature Roadmap

Version 3.0 12.2(15)ZJ • Additional Language Options for IP Phone Display, page xxxvi
12.3(4)T
• Consultative Call Transfer and Forward Using H.450.2 and H.450.3 for SCCP
Phones, page xxxvii
• Customized System Message for Cisco Unified IP Phones, page xxxvii
• Dual-Line Mode, page xxxvii
• E1 R2 Signaling Support, page xxxviii
• European Date Formats, page xxxix
• Huntstop for Dual-Line Mode, page xxxix
• Music-on-Hold for Multicast from Flash Files, page xxxix
• Ringing Timeout Default, page xxxix
• Secondary Dial Tone, page xxxix
• Enhancement to the show ephone Command, page xl
• System Log Messages for Phone Registrations, page xl
• Three-Party G.711 Ad Hoc Conferencing, page xl
• Support for Cisco VG248 Analog Phone Gateway 1.2(1) and Higher
Versions, page xl
Version 2.1 • Cisco Unified IP Phone 7902G Support, page xlii
• Cisco Unified IP Phone 7912G Support, page xliii
• Additional Language Options for IP Phone Display, page xlii
• Cisco SRST Aggregation, page xlii
• Cisco ATA 186 and ATA 188 Support, page xlii
• Cisco Unified IP Phone Expansion Module 7914 Support, page xliii
• Enhancement to the dialplan-pattern Command, page xliii
Version 2.02 • Cisco Unified IP Phone Conference Station 7935 Support, page xliv
• Increase in Directory Numbers, page xliv
• Cisco Unity Voicemail Integration Using In-Band DTMF Signaling Across
the PSTN and BRI/PRI, page xlv
• Cisco Unified SRST was implemented on the Cisco Catalyst 4500 access
gateway module and Cisco 7200 routers (NPE-225, NPE-300, and NPE400).
• Support was removed for the Cisco MC3810-V3 concentrator.
Version 2.01 • Cisco Unified SRST was implemented on the Cisco 1760 routers, and support
for the Cisco 1750 was removed.
• Support was added for additional connected Cisco IP phones.
• Support was added for additional directory numbers or virtual voice ports on
Cisco IP phones.

iii-viii
Feature Roadmap

Version 2.0 • Cisco Unified SRST was implemented on the Cisco 2600XM and Cisco 2691
routers.
• Cisco Unified SRST was integrated into Cisco IOS Release 12.2(8)T and
implemented on the Cisco 3725 and Cisco 3745 routers and the
Cisco MC3810-V3 concentrators.
• Cisco Unified SRST was implemented on the Cisco 1750 and Cisco 1751
routers.
• Huntstop support.
• Class of restriction (COR).
• Translation rule support.
• MOH and tone on hold.
• Distinctive ringing.
• Forward to a central voicemail or auto-attendant (AA) through PSTN during
Cisco Unified Communications Manager fallback.
• Phone number alias support during Cisco Unified Communications Manager
fallback: enhanced default destination support.
• List-based call restrictions for Cisco Unified Communications Manager
fallback.

iii-ix
Feature Roadmap

Version 1.0 • Support was added for 144 Cisco IP phones on the Cisco 3660 multiservice
routers.
• Cisco Unified SRST introduced on the Cisco 2600 series and
Cisco 3600 series multiservice routers and the Cisco IAD2420 series
integrated access devices.
• Cisco IP phones able to establish a connection with an SRST router in the
event of a WAN link to Cisco Unified Communications Manager failure.
• Dimming of all Cisco Unified IP Phone function keys that are not supported
during Cisco Unified SRST operation.
• Extension-to-extension dialing.
• Direct Inward Dialing (DID).
• Direct Outward Dialing (DOD).
• Calling party ID (Caller ID/ANI) display.
• Last number redial.
• Preservation of local extension-to-extension calls when WAN link fails.
• Preservation of local extension to PSTN calls when WAN link fails.
• Preservation of calls in progress when failed WAN link is re-established.
• Blind transfer of calls within IP network.
• Multiple lines per Cisco IP phone.
• Multiple-line appearance across telephones.
• Call hold (shared lines).
• Analog Foreign Exchange Station (FXS) and Foreign Exchange Office (FXO)
ports.
• BRI support for EuroISDN.
• PRI support for NET5 switch type.

iii-x
Information About New Features in Cisco Unified SRST
Information About New Features in Cisco Unified SRST

This section contains the following topics:
• New Features for Unified SRST Version 12.3, page xi
• New Features for Unified SRST Version 12.2, page xi
• New Features for Unified SRST Version 12.1, page xii
• New Feature for Unified SRST Version 12.0, page xii
• New Features for Cisco Unified SRST Version 11.0, page xii
• New Features for Cisco Unified SRST Version 10.5, page xii
• New Features in Cisco Unified SRST Version 10.0, page xiii
• New Features in Cisco Unified SRST Version 9.5, page xiv
• New Features in Cisco Unified SRST Version 9.1, page xxv
• New Features in Cisco Unified SRST Version 9.0, page xxvii
• New Features in Cisco Unified SRST Version 8.8, page xxix
• New Features in Cisco Unified SRST Version 8.0, page xxix
• New Features in Cisco Unified SRST Version 7.0/4.3, page xxix
• New Features in Cisco Unified SRST Version 4.2(1), page xxx
• New Features in Cisco Unified SRST Version 4.1, page xxx
• New Features in Cisco Unified SRST Version 4.0, page xxx
• New Features in Cisco Unified SRST Version 3.4, page xxxii
• New Features in Cisco SRST Version 3.3, page xxxii
• New Features in Cisco SRST Version 3.2, page xxxiii
• New Features in Cisco SRST Version 3.1, page xxxv
• New Features in Cisco SRST Version 3.0, page xxxvi
• New Features in Cisco SRST Version 2.1, page xl
• New Features in Cisco SRST Version 2.02, page xliv
New Features for Unified SRST Version 12.3

Unified SRST 12.3 Release introduces support for the following new feature:
• Secure SCCP SRST on Cisco 4000 Series Integrated Services Router, page 235

Unified SRST 12.2 Release introduces support for the following new feature:
• Unified E-SRST with Support for Voice Hunt Group, page 88

iii-xi

Unified SRST 12.1 introduces support for the following new features:
• Cisco Smart License, page 14
• Secure SIP SRST Support on Cisco 4000 Series Integrated Services Router, page 234
New Feature for Unified SRST Version 12.0

Cisco Unified SRST 12.0 introduces support for IPv6 protocols on SIP IP Phones. For more information
on IPv6 Support introduced for Unified SRST, see IPv6 Support for Unified SRST SIP IP Phones,
page 18.
New Features for Cisco Unified SRST Version 11.0

Cisco Unified SRST 11.0 supports the following new Cisco IP phones and adapters:
• Support for Cisco IP Phone 7811
• Support for Cisco IP Phones 8811, 8831, 841, 8851, 8851NR, 8861
• Support for Cisco ATA-190
For information on the phones supported in Cisco Unified SRST 11.0, see Phone Feature Support Guide
for Unified CME, Unified SRST, Unified E-SRST, and Unified Secure SRST.
New Features for Cisco Unified SRST Version 10.5

Cisco Unified SRST 10.5 supports the following features:
• Unified E-SRST Scale Support, page 113
For more information on the Cisco Unified SRST 10.5 supported feature, see the “SCCP: Configure
Unified E-SRST” section on page 106.
Cisco Unified SRST 10.5 supports the following new Cisco Unified SIP IP phones:
• Support for Cisco Unified DX650 SIP IP Phones, page xii
• Support for Cisco Unified 78xx SIP IP Phones, page xiii
Support for Cisco Unified DX650 SIP IP Phones

For information on feature support for the Cisco Unified DX650 SIP IP Phones in Cisco Unified SRST
10.5, see Phone Feature Support Guide for Unified CME, Unified SRST, Unified E-SRST, and Unified
Secure SRST.

iii-xii
New Features in Cisco Unified SRST Version 10.0
Support for Cisco Unified 78xx SIP IP Phones

For information on feature support for the Cisco Unified 78xx SIP IP Phones in Cisco Unified SRST
10.5, see Phone Feature Support Guide for Unified CME, Unified SRST, Unified E-SRST, and Unified
Secure SRST.

Cisco Unified SRST 10.0 supports the following new features:
• Cisco Jabber for Windows, page xiii
To obtain an account on Cisco.com, go to www.cisco.com and click Register at the top of the screen.
Cisco Jabber for Windows

Cisco Jabber for Windows client is supported from Cisco Unified CME Release 10 onwards.Cisco
Jabber for Windows supports the visual voicemail functionality integrated with the Cisco Unity
connection. Cisco Jabber for Windows is a SIP-based soft client with integrated Instant Messaging and
presence functionality, and uses the new Client Services Framework 2nd Generation (CSF2G)
architecture.
CSF is a unified communications engine that is reused by multiple Cisco PC-based clients. The Cisco
Jabber client has to be registered with a presence server such as cloud-based Cisco Webex server, or
Cisco Unified Presence server to avail the standard XMPP-based instant messaging functionalities. The
client is identified by a device ID name that can be configured under the voice register pool in Cisco
Unified CME. You should configure the username and password under voice register pool to identify the
user logging into Cisco Unified CME through Cisco Jabber for Windows client. The device discovery
process uses HTTPS connection. Therefore, you should configure the secure HTTP on Cisco Unified
CME. A new phone type, Jabber-Win has been added to configure the voice register pool for Cisco
Jabber for Windows client.
Restrictions
• The Cisco Jabber for Windows client version should be version 9.1.0 and later version.
• The Cisco Jabber for Windows client should register with a presence server such as cloud-based
Webex server, or a Cisco Unified Presence server to enable the telephony features on the Jabber
client.
• The Cisco Jabber for Windows client supports only the visual voicemail functionality using Internet
Message Access Protocol (IMAP) on the Cisco Unity Connection.
• The Cisco Jabber for Windows client does not support software-based conferencing and supports
only the softphone mode with Cisco Unified CME.
• Desk phone models are not supported.
For configuration information, see the “Cisco Jabber for Windows” section of Cisco Unified
Communications Manager Administration Guide.

iii-xiii
Version Negotiation for Cisco Unified SIP IP Phones

The version negotiation for Cisco Unified SIP IP Phones was introduced in Cisco Unified SRST 10.0
release. For more information on the Cisco Unified SRST 10.0 supported features, see the “SIP:
Configure Unified E-SRST” section on page 90.

• After-hour Pattern Blocking Support for Regular Expressions, page xiv
• Call Park Recall Enhancement, page xv
• Display Support for Name of Called Voice Hunt Groups, page xvi
• Preventing Local-Call Forwarding to Final Agent in Voice Hunt Groups, page xvii
• Trunk-to-Trunk Transfer Blocking for Toll Fraud Prevention on Cisco Unified SIP IP Phones,
page xvii
After-hour Pattern Blocking Support for Regular Expressions

In Cisco Unified SRST 9.5, support for afterhours pattern blocking is extended to regular expression
patterns for dial plans on Cisco Unified SIP and Cisco Unified SCCP IP phones. With this support, users
can add a combination of fixed dial plans and regular expression-based dial plans.
When a call is initiated after hours, the dialed number is matched against a combination of dial plans. If
a match is found, the call is blocked.
To enable regular expression patterns to be included when configuring afterhours pattern blocking, the
after-hours block pattern command is modified to include regular expressions as a value for the pattern
argument in the following command syntax:
after-hours block pattern pattern-tag pattern
This command is available in the following configuration modes:
• telephony-service—For both SCCP and SIP Phones.
• ephone-template—For SCCP phones only.
Note The maximum length of a regular expression pattern is 32 for both Cisco Unified SIP and Cisco Unified
SCCP IP phones.
If calls to the following numbers are to be blocked after hours:

• numbers beginning with ‘0’ and ‘00’
• numbers beginning with 1800, followed by four digits
• numbers 9876512340 to 9876512345
then the following configurations can be used:
• after-hours block pattern 1 0*
• after-hours block pattern 2 00*

iii-xiv
• after-hours block pattern 3 1800….

• after-hours block pattern 4 987651234[0-5]
Note There is no change in the number of afterhours patterns that can be added. The maximum number is still
100.
For more information on configuration examples, see the “Configuring Afterhours Block Patterns of
Regular Expressions: Example” section of Cisco Unified Communications Manager Administration
Guide.
For a summary of the basic Cisco IOS regular expression characters and their functions, see the “Cisco
Regular Expression Pattern Matching Characters“ section of Terminal Services Configuration Guide.
Call Park Recall Enhancement

Before Cisco Unified CME 9.5, a parked call could not be recalled by or transferred to the phone that
put the call in park or the original phone that transferred the call when the destination phone was offhook
or ringing.
In Cisco Unified CME 9.5, the recall force keyword is added to the call-park system command in
telephony-service configuration mode to allow a user to force the recall or transfer of a parked call to
the phone that put the call in park or the phone with the reserved-for number as its primary DN when the
destination phone is available to answer the call.
In Cisco Unified CME 10.5, a new ring tone is introduced for park recall to assist the phone user to
distinctly identify the type of call.
This feature is supported on all phone families for SCCP endpoints and on 89XX and 99XX phone
families for SIP endpoints. No configurations are required to activate this feature.
Examples
The following example configures the Call Park Recall:

Router# configure terminal
Router(config)# telephony-service
Router(config)# srst mode auto-provision all
Router(config-telephony)# call-park system ? recall Configure parameters for recall
Router(config-telephony)# call-park system recall ? force Force recall for busy call park
initiator
Router(config-telephony)# call-park system recall force
Park Monitor
In Cisco Unified CME 8.5 and later versions, the park monitor feature allows you to park a call and
monitor the status of the parked call until the parked call is retrieved or abandoned. When a Cisco
Unified SIP IP Phone 8961, 9951, or 9971 parks a call using the park soft key, the park monitoring
feature monitors the status of the parked call. The park monitoring call bubble is not cleared until the
parked call gets retrieved or is abandoned by the parkee. This parked call can be retrieved using the same
call bubble on the parker’s phone to monitor the status of the parked call.
Once a call is parked, Cisco Unified CME sends a SIP NOTIFY message to the parker phone indicating
the “parked” event along with the park slot number so that the parker phone can display the park slot
number as long as the call remains parked.

iii-xv
When a parked call is retrieved, Cisco Unified CME sends another SIP NOTIFY message to the parker
phone indicating the “retrieved” event so that the phone can clear the call bubble. When a parked call is
disconnected by the parkee, Cisco Unified CME sends a SIP NOTIFY message to the parker phone
indicating the “abandoned” event and the parker phone clears the call bubble upon cancellation of the
parked call.
When a parked call is recalled or transferred, Cisco Unified CME sends a SIP NOTIFY message to the
parker phone indicating the “forwarded” event so that parker phone can clear the call bubble during park,
recall, and transfer. You can also retrieve a parked call from the parker phone by directly selecting the
call bubble or pressing the resume soft key on the phone.
Display Support for Name of Called Voice Hunt Groups

A voice hunt group is associated with a pilot number. But because there is no association with the name
of the voice hunt group when calls are forwarded from the voice hunt group to the final number, the
forwarding number is sent without the name of the forwarding party. The final number can be in the form
of a voicemail, a Basic Automatic Call Distribution (BACD) script, or another extension.
In Cisco Unified SRST 9.5, the display of the name of the called voice-hunt-group pilot is supported by
configuring the following command in voice hunt-group or ephone-hunt configuration mode:
[no] name “primary pilot name” [secondary “secondary pilot name”]
The secondary name is optional and when the secondary pilot name is not explicitly configured, the
primary pilot name is applicable to both pilot numbers.
For configuration information, see the “Associating a Name with a Called Voice Hunt Group” section
Examples
The following example configures the primary pilot name for both the primary and secondary pilot
numbers:
name SALES
The following example configures different names for the primary and secondary pilot numbers:
name SALES secondary SALES-SECONDARY
Note Use quotes (") when input strings have spaces in between as shown in the next three examples.
The following example associates a two-word name for the primary pilot number and a one-word name
for the secondary pilot number:
name “CUSTOMER SERVICE” secondary CS
The following example associates a one-word name for the primary pilot number and a two-word name
for the secondary pilot number:
name FINANCE secondary “INTERNAL ACCOUNTING”
The following example associates two-word names for the primary and secondary pilot numbers:
name “INTERNAL LLER” secondary “EXTERNAL LLER”

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For configuration information, see the “Associating a Name with a Called Voice Hunt Group” section of
Cisco Unified Communications Manager Administration Guide.
For configuration examples, see the “Example: Associating a Name with a Called Voice Hunt Group”
section of Cisco Unified Communications Manager Administration Guide.
Restrictions
• Display support applies to Cisco Unified SCCP IP phones in voice hunt-group and ephone-hunt
configuration modes but are not supported in Cisco Unified SIP IP phones.
• Called name and called number information displayed on the caller’s phone follows existing
behavior, where the called names and called numbers are updated so that a sequential hunt reflects
the name and number of the ringing phone.
Preventing Local-Call Forwarding to Final Agent in Voice Hunt Groups

Local or internal calls are calls originating from a Cisco Unified SIP or Cisco Unified SCCP IP phone
in the same Cisco Unified CME system.
Before Cisco Unified CME 9.5, the no forward local-calls command was configured in ephone-hunt
group to prevent a local call from being forwarded to the next agent.
In Cisco Unified CME 9.5, local calls are prevented from being forwarded to the final destination using
the no forward local-calls to-final command in parallel or sequential voice hunt-group configuration
mode.
When the no forward local-calls to-final command is configured in sequential voice hunt-group
configuration mode, local calls to the hunt-group pilot number are sent sequentially only to the list of
members of the group using the rotary-hunt technique. In case all the group members of the voice hunt
group are busy, the caller hears a busy tone. If any of the group members are available but do not answer,
the caller hears a ringback tone and is eventually disconnected after the specified timeout. The call is not
forwarded to the final number.
When the no forward local-calls to-final command is configured in parallel voice hunt-group
configuration mode, local calls to the hunt-group pilot number are sent simultaneously to the list of
members of the group using the blast technique. In case all the group members of the voice hunt group
are busy, the caller hears a busy tone. If any of the group members are available but do not answer, the
caller hears a ringback tone and is eventually disconnected after the specified timeout.The call is not
forwarded to the final number. or configuration examples, see the “Preventing Local-Call Forwarding to
Final Agent in Voice Hunt Groups” section of” section of Cisco Unified Communications Manager
Administration Guide.
Trunk-to-Trunk Transfer Blocking for Toll Fraud Prevention on Cisco

Unified SIP IP Phones
In Cisco Unified Survivable Remote Site Telephony (SRST) 4.0, trunk-to-trunk transfer blocking for toll
bypass fraud prevention is supported on Cisco Unified Skinny Client Control Protocol (SCCP) IP
phones.
Table iii-3 lists the transfer-blocking commands and the appropriate configuration modes for Cisco
Unified CME and Cisco Unified SRST.

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Table iii-3 Configuration Modes for Transfer-Blocking Commands
Commands Cisco Unified SRST

transfer-pattern call-manager-fallback
transfer max-length voice register pool
transfer-pattern blocked voice register pool
conference transfer-pattern call-manager-fallback
conference max-length voice register pool or voice register template
conference-pattern blocked voice register pool or voice register template
Note The call transfer and conference restrictions apply when transfers or conferences are initiated toward
external parties, like a PSTN trunk, a SIP trunk, or an H.323 trunk. The restrictions do not apply to
transfers and conferences to local extensions.
transfer-pattern
The transfer-pattern command for Cisco Unified SIP IP phones functions like the transfer-pattern
command for Cisco Unified SCCP IP phones by allowing all, not just local, transfers to take place.
The transfer-pattern command specifies the directory numbers for call transfer. The command can be
configured up to 32 times using the following command syntax: transfer-pattern transfer-pattern
[blind].
Note The blind keyword in the transfer-pattern command applies only to Cisco Unified SCCP IP phones and
does not apply to Cisco Unified SIP IP phones.
With the transfer-pattern command configured, only call transfers to numbers that match the
configured transfer pattern are allowed to take place. With the transfer pattern configured, all or a subset
of transfer numbers can be dialed and the transfer to a remote party can be initiated.
The following are examples of configurable transfer patterns:
• .T—This configuration allows call transfers to any destinations with one or more digits, like 123,
877656, or 76548765.
• 919........—This configuration only allows call transfers to remote numbers beginning with “919”
and followed by eight digits, like 91912345678. However, call transfers to 9191234 or
919123456789 are not allowed.
Backward Compatibility
To maintain backward compatibility, all call transfers from Cisco Unified SIP IP phones to any number
(local or over trunk) are allowed when no transfer patterns are configured through the transfer-pattern,
transfer-pattern blocked, or transfer max-length commands.

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For Cisco Unified SCCP IP phones, call transfers over trunk continue to be blocked when no transfer
patterns are configured.
Dial Plans
Whatever dial plan is used for external calls, the same numbers should be configured as specific numbers
using the transfer-pattern command.
If a dial plan requires “9” to be dialed before an external call is made, then “9” should be a prefix of the
transfer-pattern number. For example, if 12345678 is an external number that requires “9” to be dialed
before the external call can be made, then the transfer-pattern number should be 912345678.
transfer max-length
The transfer max-length command is used to indicate the maximum length of the number being dialed
for a call transfer. When only a specific number of digits are to be allowed during a call transfer, a value
between 3 and 16 is configured.When the number dialed exceeds the maximum length configured, then
the call transfer is blocked.
For example, if the maximum length is configured as 5, then only call transfers from Cisco Unified SIP
IP phones up to a five-digit directory number are allowed. All call transfers to directory numbers with
more than five digits are blocked.
Note If only transfer max length is configured and conference max-length is not configured, then transfer
max-length takes effect for transfers and conferences.
transfer-pattern blocked
When the transfer-pattern blocked command is configured for a specific phone, no call transfers are
allowed from that phone over the trunk.
This feature forces unconditional blocking of all call transfers from the specific phone to any other
non-local numbers (external calls from one trunk to another trunk). No call transfers from this specific
phone are possible even when a transfer pattern matches the dialed digits for transfer.
Table iii-4 compares the behaviors of Cisco Unified SCCP and SIP IP phones for specific configurations.
Table iii-4 Behaviors of Cisco Unified IP Phones for Specific Configurations
Configuration Cisco Unified SCCP IP Phones Cisco Unified SIP IP Phones

No transfer patterns are All non-local call transfers are All non-local call transfers are
configured. blocked. allowed for backward
compatibility.

iii-xix
Table iii-4 Behaviors of Cisco Unified IP Phones for Specific Configurations (continued)

Specific transfer patterns are Call transfers to specific external Call transfers to specific external
configured. entities are allowed. entities are allowed.
The transfer-pattern blocked All non-local call transfers are All non-local call transfers are
command is configured. blocked. blocked.
Note The configuration Note The configuration
reverts to the default, unconditionally blocks
where no transfer all non-local call
patterns are configured. transfers. It does not
return to the default,
where all non-local call
transfers are allowed.
conference-pattern blocked
The conference-pattern blocked command is used to prevent extensions on a voice register pool from
initiating conferences.
The following table summarizes the behavior of the conference-pattern blocked command in relation
to no conference-pattern blocked, conference max-length, no conference max-length, and transfer
max-length commands.
conference max-length no conference max-length

Allowing/Blocking of Allowing/Blocking of conference call
conference call depends on depends on configured transfer
No conference-pattern configured conference max-length
blocked (default case) max-length
conference-pattern blocked Conference calls are not allowed on SIP and SCCP phones.
Max-length <= allowed Max-length > allowed

max-length max-length
Transfer Conference Transfer Conference
Transfer max-length + Y Y N N
No Conference max-length (use transfer
max-length for conference cases too, as
conference max-length not configured)
No transfer max-length + Conference Y Y Y N

max-length (conference max-length has
precedence over transfer max-length for
conference)

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Max-length <= allowed Max-length > allowed

max-length max-length
No transfer max-length + Conference Y Y N N
max-length (conference max-length has
precedence over transfer max-length for
conference)
No transfer max-length + No conference All transfer and conference calls are allowed
max-length
Configuring the Maximum Number of Digits for a Conference Call

This feature enables you to specify the maximum number of digits while making a conference call.
Prerequisites
• Cisco Unified SRST 10.5 or a later version.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice register pool pool-tag
4. conference max-length value
5. end

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DETAILED STEPS
Command or Action Purpose
Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.
Example:
Step 3 voice register pool pool-tag Enters voice register pool configuration mode and creates a
or pool configuration for a Cisco Unified SIP IP phone in Cisco
ephone phone-tag
Unified CME or for a set of Cisco Unified SIP IP phones in
Example: Cisco Unified SIP SRST.
Router(config)# voice register pool 25 • pool-tag—Unique number assigned to the pool. Range is
1 to 100.
or
Enters voice register template configuration mode and defines
a template of common parameters for Cisco Unified SIP IP
phones.
• template-tag—Declares a template tag. Range is 1 to 10.
or
Enters ephone configuration mode.
phone-tag—Unique sequence number that identifies this
ephone during configuration tasks. The maximum number of
ephones is version and platform-specific. Type ? to display
range.
Step 4 configure max-length value Allows the conference of calls from Cisco IP phones to
specified directory numbers of phones other than Cisco IP
phones.
Example:
Router(config-telephony)# conference • conference max-length—Allows conference call
max-lenght 6 depending on the configured conference max-length.
Range is 3 to 16.
Step 5 end Exits telephony-service configuration mode and enters
privileged EXEC mode.
Example:
Router(config-telephony)# end
Configuring Conference Blocking Options for Phones

This feature prevents extensions from making conference calls to directory numbers that are otherwise
allowed globally.

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Prerequisites
• Cisco Unified SRST 10.5 or a later version.
• The transfer-pattern command must be configured.
• The conference transfer-pattern command must be configured.
SUMMARY STEPS
1. enable
4. conference-pattern blocked
5. end

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DETAILED STEPS

Example:
Router> enable
Example:
Step 3 voice register pool pool-tag Enters voice register pool configuration mode and creates a
pool configuration for a Cisco Unified SIP IP phone in Cisco
Unified CME or for a set of Cisco Unified SIP IP phones in
Example: Cisco Unified SIP SRST.
Router(config)# voice register pool 25 • pool-tag—Unique number assigned to the pool. Range is
1 to 100.
or
Enters voice register template configuration mode and defines
a template of common parameters for Cisco Unified SIP IP
phones.
• template-tag—Declares a template tag. Range is 1 to 10.
or
Enters ephone configuration mode.
phone-tag—Unique sequence number that identifies this
ephone during configuration tasks. The maximum number of
ephones is version and platform-specific. Type? to display
range.
Step 4 conference-pattern blocked Allows the conference of calls from Cisco IP phones to
specified directory numbers of phones other than Cisco IP
Example: phones.
Router(config-telephony)# conference-pattern
blocked • conference-pattern blocked—No conference calls are
allowed.
Step 5 exit Exits telephony-service configuration mode and enters global
configuration mode.
Example:
Router(config-telephony)# exit
transfer-pattern blocked
When the transfer-pattern blocked command is configured for a specific phone, no call transfers are
allowed from that phone over the trunk.
This feature forces unconditional blocking of all call transfers from the specific phone to any other
non-local numbers (external calls from one trunk to another trunk). No call transfers from this specific
phone are possible even when a transfer pattern matches the dialed digits for transfer.

iii-xxiv
Table iii-5 compares the behaviors of Cisco Unified SCCP and SIP IP phones for specific configurations.
Table iii-5 Behaviors of Cisco Unified IP Phones for Specific Configurations

No transfer patterns are All non-local call transfers are All non-local call transfers are
configured. blocked. allowed for backward
compatibility.
Specific transfer patterns are Call transfers to specific external Call transfers to specific external
configured. entities are allowed. entities are allowed.
The transfer-pattern blocked All non-local call transfers are All non-local call transfers are
command is configured. blocked. blocked.
Note The configuration Note The configuration
reverts to the default, unconditionally blocks
where no transfer all non-local call
patterns are configured. transfers. It does not
return to the default,
where all non-local call
transfers are allowed.
conference transfer-pattern
When both the transfer-pattern and conference transfer-pattern commands are configured and dialed
digits match the configured transfer pattern, conference calls are allowed. However, when the dialed
digits do not match any of the configured transfer pattern, the conference call is blocked.
For information on provisioning Cisco Unified IP phones for secure access to web content using HTTPS,
see the “HTTPS Provisioning for Cisco Unified IP Phones” section of Cisco Unified Communications
Manager Express System Administrator Guide.
For configuration examples, see the “Configuring HTTPS Support for Cisco Unified CME:Example”
section of Cisco Unified Communications Manager Administration Guide.

• KEM Support for Cisco Unified SIP IP Phones, page xxvi
• Enhancement in Speed-Dial Support, page xxvi
• Voice Hunt Group Support, page xxvi
Note If you have older routers, such as the VG26nn and VG37nn platforms and Cisco Integrated Services
Router (ISR) Generation 1 platforms (Cisco ISR 1861, 2800, and 3800 Series), you need to upgrade to
Cisco ISR 881, 886VA, 887VA, 888, 888E, 1861E, 2900, 3900, and 3900E Series platforms to utilize
these new features.

iii-xxv
KEM Support for Cisco Unified SIP IP Phones

Cisco Unified IP Key Expansion Modules (KEMs) are supported on Cisco Unified 8851/51NR, 8861,
8961, 9951, and 9971 SIP IP phones from Cisco Unified SIP SRST 9.1.
For information on KEM support for Cisco Unified 8851/51NR, 8861, 8961, 9951, and 971 SIP IP
phones, see Phone Feature Support Guide for Unified CME, Unified SRST, Unified E-SRST, and
Unified Secure SRST.
Restrictions
• Bulk registration is not supported for KEMs in Cisco Unified SRST. Phones do not send bulk
registration requests but always use the User Datagram Protocol (UDP) port for registration.
• KEM is not supported for Cisco Unified SCCP IP Phones and Cisco Unified SIP IP Phones other
than the Cisco Unified 8851/51NR, 8861, 8961, 9951, and 9971 SIP IP phones.
• Features configured on keys are disabled when supported Cisco Unified SIP IP phones are in Cisco
Unified SIP SRST.
• All Cisco Unified 8851/51NR, 8861,8961, 9951, and 9971 SIP IP phone restrictions and limitations
apply to KEMs.
• All Cisco Unified SIP SRST feature restrictions and limitations apply to KEMs.
For more information on how the blf-speed-dial, number, and speed-dial commands, in voice register
pool configuration mode, have been modified, see Cisco Unified Communications Manager Express
Command Reference.
For information on installing KEMs on Cisco Unified IP Phone, see the “Installing a Key Expansion
Module on the Cisco Unified IP Phone” section of Cisco Unified IP Phone 8961, 9951, and 9971
Administration Guide for Cisco Unified Communications Manager 7.1 (3) (SIP).
For information on installing KEMs on Cisco Unified 8811, 8841, 8851, 8851NR, and 8861 Phones, see
the Cisco IP Phone Key Expansion Module section of Cisco IP Phone 8811, 8841, 8851, 8851NR, and
8861 Administration Guide for Cisco Unified Communications Manager.
Enhancement in Speed-Dial Support

In Cisco Unified SRST 9.1, the “,” or comma (pause indicator) is ignored to avoid a break in speed-dial
support.
Because the pause speed-dial feature (supported in Cisco Unified Communications Manager or Cisco
Unified CM) is not supported in Cisco Unified SRST, Cisco Unified CM and phones (Cisco Unified
SCCP IP phones and Cisco Unified SIP IP phones) registered in Cisco Unified SRST maintain backward
compatibility in Cisco Unified SRST mode. When phones failover to the Cisco Unified SRST router
during WAN outages and Cisco Unified CM failure, the phones only send the speed-dial numbers when
the pause speed-dial buttons are pressed. The comma pause indicator is ignored and the preconfigured
FAC, PIN, and DTMF are not sent.
For information on configuring speed-dial in Cisco Unified Communications Manager, see the “Device
setup” chapter of Cisco Unified Communications Manager Administration Guide.
Voice Hunt Group Support

Cisco Unified SIP SRST 9.1 supports voice hunt groups. Voice hunt groups allow a call placed to a single
(pilot) number to contact multiple destinations.

iii-xxvi
There are three different types of voice hunt groups. Each type uses a different strategy to determine the
first number that rings for successive calls to the pilot number until a number answers.
• Parallel Hunt Groups—Allows an incoming call to simultaneously ring all the numbers in the hunt
group member list.
• Sequential Hunt Groups—Allows an incoming call to ring all the numbers in the left-to-right order
in which they were listed when the hunt group was defined. The first number in the list is always the
first number tried when the pilot number is called. Maximum number of hops is not a configurable
parameter for sequential hunt groups.
• Longest-Idle Hunt Groups—Allows an incoming call to first go to the number that has been idle the
longest for the number of hops specified when the hunt group was defined. The longest-idle time is
determined from the last time that a phone registered, re-registered, or went on-hook.
While ephone hunt groups only support Cisco Unified SCCP IP phones, a voice hunt group supports
Cisco Unified SCCP IP phones, Cisco Unified SIP IP phones, or a mixture of Cisco Unified SCCP IP
phones and Cisco Unified SIP IP phones.
With the voice hunt group feature preconfigured in the Cisco Unified SIP SRST router, voice hunt groups
continue to be supported after phones fallback from Cisco Unified CM to the Cisco Unified SIP SRST
router.
Restrictions
• Hunt group statistics is not supported for voice hunt groups in Cisco Unified SRST.
• Hunt group nesting or setting the final number of one hunt group as the pilot of another hunt group
is not supported.

Cisco Unified SRST 9.0 supports the following new Cisco Unified SIP IP phones:
• Cisco Unified 6901 and 6911 SIP IP Phones
• Cisco Unified 6921, 6941, 6945, and 6961 SIP IP Phones
• Cisco Unified 8941 and 8945 SIP IP Phones
• Multiple Calls Per Line, page xxviii
• Voice and Fax Support on Cisco ATA-187, page xxix
Support for Cisco Unified 6901 and 6911 SIP IP Phones

For information on feature support for the Cisco Unified 6901 and 6911SIP IP Phones in Cisco Unified
SRST, see Phone Feature Support Guide for Unified CME, Unified SRST, Unified E-SRST, and Unified
Secure SRST.

iii-xxvii
Support for Cisco Unified 6921, 6941, 6945, and 6961 SIP IP Phones
For information on feature support for the Cisco Unified 6921, 6941, 6945, and 6961 SIP IP Phones in
Cisco Unified SRST, see Phone Feature Support Guide for Unified CME, Unified SRST, Unified
E-SRST, and Unified Secure SRST.
Support for Cisco Unified 8941 and 8945 SIP IP Phones

For information on feature support for the Cisco Unified 8941 and 8945 SIP IP Phones in Cisco Unified
SRST, see Phone Feature Support Guide for Unified CME, Unified SRST, Unified E-SRST, and Unified
Secure SRST.
Multiple Calls Per Line

Cisco Unified SRST 9.0 provides support for the Multiple Calls Per Line (MCPL) feature on Cisco
Unified 6921, 6941, 6945, and 6961 SIP IP phones and Cisco Unified 8941 and 8945 SCCP and SIP IP
phones.
Before Cisco Unified SRST 9.0, the maximum number of calls supported for every directory number
(DN) on Cisco Unified 8941 and 8945 SCCP IP phones was restricted to two.
With Cisco Unified SRST 9.0, the MCPL feature overcomes the limitation on the maximum number of
calls per line.
In Cisco Unified SRST 9.0, the MCPL feature is not supported on Cisco Unified 6921, 6941, 6945, and
6961 SCCP IP phones. The maximum number of calls allowed on these phones is two and the maximum
number of calls allowed on octo-line directory numbers on these phones before activating Call Forward
Busy or a busy tone is one.
Cisco Unified 8941 and 8945 SCCP IP Phones

Before Cisco Unified SRST 9.0, the values for the max-dn and timeouts busy commands were
hardcoded for Cisco Unified 8941 and 8945 SCCP IP phones.
In Cisco Unified SRST 9.0, you can configure the max-dn and timeouts busy commands in
call-manager-fallback configuration mode. Use the max-dn command to set the maximum number of
DNs that can be supported by the router and enable dual-line mode, octo-line mode, or both modes. Use
the timeouts busy command to set the timeout value for call transfers to busy destinations.
For configuration information, see the “Configuring the Maximum Number of Calls” section on
page 152.
Cisco Unified 6921, 6941, 6945, 6961, 8941, and 8945 SIP IP Phones
In Cisco Unified SRST 9.0, the maximum number of calls for Cisco Unified 6921, 6941, 6945, 6961,
8941, and 8945 SIP IP phones is controlled by the phones.
Prerequisites
• Cisco Unified SRST 9.0 and later versions.
• Correct firmware is installed:

iii-xxviii
– 9.2(1) or a later version for Cisco Unified 6921, 6941, 6945 and 6961 SIP IP phones.
– 9.2(2) or a later version for Cisco Unified 8941 and 8945 SIP IP phones.
Voice and Fax Support on Cisco ATA-187

Cisco ATA-187 is a SIP-based analog telephone adaptor that turns traditional telephone devices into IP
devices. Cisco ATA-187 can connect with a regular analog FXS phone or fax machine on one end, while
the other end is an IP side that uses SIP for signaling and registers as a Cisco Unified SIP IP phone.
Cisco ATA-187 functions as a Cisco Unified SIP IP phone that supports T.38 fax relay and fax
pass-through, enabling the real-time transmission of fax over IP networks. The fax rate is from 7.2 to
14.4 kbps.
For information on feature support for the Cisco ATA-187 in Cisco Unified SRST, see Phone Feature
Support Guide for Unified CME, Unified SRST, Unified E-SRST, and Unified Secure SRST.
For more information on Cisco ATA-187, see Cisco ATA 187 Analog Telephone Adaptor Administration
Guide for SIP.

Cisco Unified SRST 8.8 supports the following new Cisco Unified SCCP IP phones:
• Cisco Unified 6945 SCCP IP Phones
Support for Cisco Unified 6945, 8941, and 8945 SCCP IP Phones
For information on feature support for the Cisco Unified 6945, 8941, and 8945 SCCP IP Phones in Cisco
Unified SRST, see Phone Feature Support Guide for Unified CME, Unified SRST, Unified E-SRST, and
Unified Secure SRST.
For information on the Cisco Unified 6945 SCCP IP Phone, see Cisco Unified IP Phone 6945 User Guide
for Cisco Unified Communications Manager Express Version 8.8 (SCCP).
For information on the Cisco Unified 8941 and 8945 SCCP IP Phones, see Cisco Unified IP Phone 8941
and 8945 User Guide for Cisco Unified Communications Manager Express Version 8.8 (SCCP).

Beginning with Cisco IP Phone firmware 8.5(3) and Cisco IOS Release 15.1(1)T, Cisco SRST supports
SIP signaling over UDP, TCP, and TLS connections, providing both RTP and SRTP media connections
based on the security settings of the IP phone.
New Features in Cisco Unified SRST Version 7.0/4.3

Cisco Unified SRST 7.0/4.3 supports the following new features:

iii-xxix
New Features in Cisco Unified SRST Version 4.2(1)
• Configuring Eight Calls per Button (Octo-Line), page 150

• Configuring Consultative Transfer, page 187
New Features in Cisco Unified SRST Version 4.2(1)

Cisco Unified SRST Version 4.2(1) introduces the following new features:
• Enhancements for Enhanced 911 Services, page 130

Cisco Unified SRST Version 4.1 introduces the following new feature:

Cisco Unified SRST Version 4.0 has introduced the following new features:
• Additional Cisco Unified IP Phone Support, page xxx
• Cisco IP Communicator Support, page xxxi
• Fax Pass-through using SCCP and ATAs Support, page xxxi
• H.323 VoIP Call Preservation Enhancements for WAN Link Failures for SCCP Phones, page xxxi
• Video Support, page xxxi
Additional Cisco Unified IP Phone Support

The following IP phones are supported with Cisco Unified SRST systems:
• Cisco Unified IP Phone 7911G
• Cisco Unified IP Phone 7941G and Cisco Unified IP Phone 7941G-GE
• Cisco Unified IP Phone 7960G
• Cisco Unified IP Phone 7961G and Cisco Unified IP Phone 7961G-GE
In addition, the Cisco Unified IP Phone 7914 Expansion Module can attach to the Cisco 7941G-GE and
Cisco 7961G-GE. The Cisco 7914 Expansion Module adds additional features, such as adding 14 line
appearances or speed-dial numbers to your phone. You can attach one or two expansion modules to your
IP phone. When you use two expansion modules, you have 28 additional line appearances or speed-dial
numbers, or a total of 34 line appearances or speed-dial numbers. For more information, see Cisco IP
Phone 7914 Expansion Module Quick Start Guide.
No additional SRST configuration is required for these phones.
The show ephone command is enhanced to display the configuration and status of the new Cisco IP
Phones added to SRST Version 4.0. For more information, see the show ephone command in Cisco
Unified SRST and Cisco Unified SIP SRST Command Reference (All Versions).

iii-xxx
To determine compatible firmware, platforms, memory, and additional voice products that are associated
with Cisco Unified SRST 4.0, see Cisco Unified SRST 4.3 Supported Firmware, Platforms, Memory, and
Voice Products.
Cisco IP Communicator Support

Cisco IP Communicator is a software-based application that delivers enhanced telephony support on
personal computers. This SCCP-based application allows computers to function as IP phones, providing
high-quality voice calls on the road, in the office, or from wherever users may have access to the
corporate network. Cisco IP Communicator appears on a user's computer monitor as a graphical,
display-based IP phone with a color screen, a key pad, feature buttons, and soft keys.
Fax Pass-through using SCCP and ATAs Support

Fax pass-through mode is now supported using Cisco VG 224 voice gateways, Analog Telephone
Adaptors (ATA), and SCCP. ATAs ship with SIP firmware, so SCCP firmware must be loaded before this
feature can be used.
Note For ATAs that are registered to a Cisco Unified SRST system to participate in FAX calls, they must have
their ConnectMode parameter set to use the “standard payload type 0/8” as the RTP payload type in FAX
pass-through mode. For ATAs used with Cisco Unified SRST 4.0 and higher versions, this is done by
setting bit 2 of the ConnectMode parameter to 1 on the ATA. For more information, see the “Parameters
and Defaults” chapter in Cisco ATA 186 and Cisco ATA 188 Analog Telephone Adaptor Administrator's
Guide for SCCP.
H.323 VoIP Call Preservation Enhancements for WAN Link Failures for
SCCP Phones
H.323 VoIP call preservation enhancements for WAN link failures sustains connectivity for H.323
topologies where signaling is handled by an entity, such as Cisco Unified Communications Manager,
that is different from the other endpoint and brokers signaling between the two connected parties.
Call preservation is useful when a gateway and the other endpoint (typically a Cisco Unified IP phone)
are collocated at the same site and the call agent is remote and therefore more likely to experience
connectivity failures. H.323 VoIP call preservation enhancements does not support SIP Phones.
For configuration information see the “Configuring H.323 Gateways” chapter in
Cisco IOS H.323 Configuration Guide.
Video Support
This feature allows you to set video parameters for the Cisco Unified SRST to maintain close feature
parity with Cisco Unified CM. When the Cisco Unified SRST is enabled, Cisco Unified IP Phones do
not have to be reconfigured for video capabilities because all ephones retain the same configuration used
with Cisco Unified CM. However, you must enter call-manager-fallback configuration mode to set video
parameters for Cisco Unified SRST. The feature set for video is the same as that for Cisco Unified SRST
audio calls.

iii-xxxi
For more information, see the “Setting Video Parameters” section on page 345.

Cisco SRST V3.4 introduced the new features described in the following section:
• Cisco SIP SRST 3.4, page xxxii
Cisco SIP SRST 3.4

Cisco SIP SRST Version 3.4 describes SRST functionality for Session Initiation Protocol (SIP)
networks. Cisco SIP SRST Version 3.4 provides backup to an external SIP call control (IP-PBX) by
providing basic registrar and back-to-back user agent (B2BUA) services. These services are used by a
SIP IP phone in the event of a WAN connection outage when the SIP phone is unable to communicate
with its primary SIP proxy.
Cisco SIP SRST Version 3.4 can support SIP phones with standard RFC 3261 feature support locally and
across SIP WAN networks. With Cisco SIP SRST Version 3.4, SIP phones can place calls across SIP
networks in the same way as Skinny Client Control Protocol (SCCP) phones. For full information about
SIP SRST, Version 3.4, see Cisco SIP SRST Version 3.4 System Administrator Guide.
New Features in Cisco SRST Version 3.3

Cisco SRST V3.3 introduced the new features described in the following sections:
• Secure SRST, page xxxii
• Cisco Unified IP Phone 7970G and Cisco Unified 7971G-GE Support, page xxxiii
• Enhancement to the show ephone Command, page xxxiii
Secure SRST
Secure Cisco IP phones that are located at remote sites and that are attached to gateway routers can
communicate securely with Cisco Unified Communications Manager using the WAN. But if the WAN
link or Cisco Unified Communications Manager goes down, all communication through the remote
phones becomes nonsecure. To overcome this situation, gateway routers can now function in secure
SRST mode, which activates when the WAN link or Cisco Unified Communications Manager goes
down. When the WAN link or Cisco Unified Communications Manager is restored, Cisco Unified
Communications Manager resumes secure call-handling capabilities.
Secure SRST provides new SRST security features such as authentication, integrity, and media
encryption. Authentication provides assurance to one party that another party is whom it claims to be.
Integrity provides assurance that the given data has not been altered between the entities. Encryption
implies confidentiality; that is, that no one can read the data except the intended recipient. These security
features allow privacy for SRST voice calls and protect against voice security violations and identity
theft. For more information see the “Configuring Secure SRST for SCCP and SIP” section on page 231.

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Cisco Unified IP Phone 7970G and Cisco Unified 7971G-GE Support

The Cisco Unified IP Phones 7970G and 7971G-GE are full-featured telephones that provide voice
communication over an IP network. They function much like a traditional analog telephones, allowing
you to place and receive phone calls and to access features such as mute, hold, transfer, speed dial, call
forward, and more. In addition, because the phones are connected to your data network, they offer
enhanced IP telephony features, including access to network information and services, and customizable
features and services. The phones also support security features that include file authentication, device
authentication, signaling encryption, and media encryption.
The Cisco Unified IP Phones 7970G and 7971G-GE also provide a color touchscreen, support for up to
eight line or speed-dial numbers, context-sensitive online help for buttons and feature, and a variety of
other sophisticated functions. No configurations specific to SRST are necessary.
For more information, see the Cisco Unified IP Phone 7900 Series documentation index.
Note The Cisco Unified IP Phone 7914 Expansion Module can attach to your Cisco Unified IP Phones 7970G
and 7971G-GE. See the “Cisco Unified IP Phone Expansion Module 7914 Support” section on page xliii
for more information.
Enhancement to the show ephone Command

The show ephone command is enhanced to display the configuration and status of the Cisco Unified IP
Phone 7970G and Cisco Unified IP Phone 7971G-GE. For more information, see the show
ephone command in Cisco Unified SRST and Cisco Unified SIP SRST Command Reference (All
Versions).

• Enhancement to the alias Command, page xxxiii
• Enhancement to the cor Command, page xxxiv
• Enhancement to the pickup Command, page xxxiv
• Enhancement to the user-locale Command, page xxxiv
• Increased the Number of Cisco Unified IP Phones Supported on the Cisco 3845, page xxxiv
• MOH Live-Feed Support, page xxxiv
• No Timeout for Call Preservation, page xxxv
• RFC 2833 DTMF Relay Support, page xxxv
• Translation Profile Support, page xxxv
Enhancement to the alias Command

The alias command is enhanced as follows:
• The cfw keyword was added, providing call forward no-answer/busy capabilities.

iii-xxxiii
• The maximum number of alias commands used for creating calls to telephone numbers that are
unavailable during Cisco Unified Communications Manager fallback was increased to 50.
• The alternate-number argument can be used in multiple alias commands.
For more information, see the alias command in Cisco Unified SRST and Cisco Unified SIP SRST
Command Reference (All Versions).
Enhancement to the cor Command

The maximum number of cor lists has increased to 20.
For more information, see the cor command in Cisco Unified SRST and Cisco Unified SIP SRST
Enhancement to the pickup Command

The pickup command was introduced to enable the PickUp soft key on all Cisco Unified IP Phones,
allowing an external Direct Inward Dialing (DID) call coming into one extension to be picked up from
another extension during SRST.
For more information, see the pickup command in Cisco Unified SRST and Cisco Unified SIP SRST
Enhancement to the user-locale Command

The user-locale command is enhanced to display the Japanese Katakana country code. Japanese
Katakana is available in Cisco Unified Communications Manager V4.0 or later versions.
For more information, see the user-locale command in the
Cisco Unified SRST and Cisco Unified SIP SRST Command Reference (All Versions).
Increased the Number of Cisco Unified IP Phones Supported on the Cisco

3845
The Cisco 3845 now supports 720 phones and up to 960 ephone-dns or virtual voice ports.
MOH Live-Feed Support

Cisco Unified SRST is enhanced with the new moh-live command. The moh-live command provides
live-feed MOH streams from an audio device connected to an E&M or FXO port to Cisco IP phones in
SRST mode. If an FXO port is used for a live feed, the port must be supplied with an external third-party
adaptor to provide a battery feed. Music from a live feed is obtained from a fixed source and is
continuously fed into the MOH playout buffer instead of being read from a flash file. Live-feed MOH
can also be multicast to Cisco IP phones. See the “Integrating Cisco Unified Communications Manager
and Cisco Unified SRST to Use Cisco Unified SRST as a Multicast MOH Resource” section on page 11
for configuration instructions.

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No Timeout for Call Preservation

To preserve existing H.323 calls on the branch in the event of an outage, disable the H.225 keepalive
timer by entering the no h225 timeout keepalive command. This feature is supported in Cisco IOS
Releases 12.3(7)T1 and higher versions. See the “Cisco Unified SRST Feature Overview” section on
page 1 for more information.
H.323 is not supported with SIP phones.
RFC 2833 DTMF Relay Support

Cisco Skinny Client Control Protocol (SCCP) phones, such as those used with Cisco SRST systems,
provide only out-of-band DTMF digit indications. To enable SCCP phones to send digit information to
remote SIP-based IVR and voice-mail applications, Cisco SRST 3.2 and later versions provide
conversion from the out-of-band SCCP digit indication to the SIP standard for DTMF relay, which is
RFC 2833. You select this method in the SIP VoIP dial peer using the dtmf-relay rtp-nte command. See
the “How to Configure DTMF Relay for SIP Applications and Voicemail” section on page 340 for
configuration instructions.
To use voicemail on a SIP network that connects to a Cisco Unity Express system, use a nonstandard SIP
Notify format. To configure the Notify format, use the sip-notify keyword with the dtmf-relay
command. Using the sip-notify keyword may be required for backward compatibility with Cisco SRST
Versions 3.0 and 3.1.
Translation Profile Support

Cisco SRST 3.2 and later versions support translation profiles. Translation profiles allow you to group
translation rules together and to associate translation rules with the following:
• Called numbers
• Calling numbers
• Redirected called numbers
See the “Enabling Translation Profiles” section on page 192 for more configuration information. For
more information on the translation-profile command, see
Cisco Unified SRST and Cisco Unified SIP SRST Command Reference (All Versions).

Note For information about Cisco Unified IP phones, see the Cisco Unified IP Phone 7900 Series
documentation.

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Cisco Unified IP Phone 7920 Support

The Cisco Unified Wireless IP Phone 7920 is an easy-to-use IEEE 802.11b wireless IP phone that
provides comprehensive voice communications in conjunction with Cisco Unified CM and Cisco
Aironet 1200, 1100, 350, and 340 Series of Wi-Fi (IEEE 802.11b) access points. As a key part of the
Cisco AVVID Wireless Solution, the Cisco Unified Wireless IP Phone 7920 delivers seamless intelligent
services, such as security, mobility, quality of service (QoS), and management, across an end-to-end
Cisco network.
No configuration is necessary.
Cisco Unified IP Phone 7936 Support

The Cisco Unified IP Conference Station 7936 is an IP-based, hands-free conference room station that
uses VoIP technology. The IP Conference Station replaces a traditional analog conferencing unit by
providing business conferencing features—such as call hold, call resume, call transfer, call release,
redial, mute, and conference—over an IP network.

• Additional Language Options for IP Phone Display, page xxxvi
• Consultative Call Transfer and Forward Using H.450.2 and H.450.3 for SCCP Phones, page xxxvii
• Customized System Message for Cisco Unified IP Phones, page xxxvii
• Dual-Line Mode, page xxxvii
• E1 R2 Signaling Support, page xxxviii
• European Date Formats, page xxxix
• Huntstop for Dual-Line Mode, page xxxix
• Music-on-Hold for Multicast from Flash Files, page xxxix
• Ringing Timeout Default, page xxxix
• Secondary Dial Tone, page xxxix
• Enhancement to the show ephone Command, page xl
• System Log Messages for Phone Registrations, page xl
• Three-Party G.711 Ad Hoc Conferencing, page xl
• Support for Cisco VG248 Analog Phone Gateway 1.2(1) and Higher Versions, page xl
Additional Language Options for IP Phone Display

Displays for the Cisco Unified IP Phone 7940G and Cisco Unified IP Phone 7960G can be configured
with additional ISO-3166 codes for German, Danish, Spanish, French, Italian, Japanese, Dutch,
Norwegian, Portuguese, Russian, Swedish, United States.

iii-xxxvi
Note This feature is available only for Cisco Unified SRST running under Cisco Unified CM V3.2.
Consultative Call Transfer and Forward Using H.450.2 and H.450.3 for
SCCP Phones
Cisco SRST V1.0, Cisco SRST V2.0, and Cisco SRST V2.1 allow blind call transfers and blind call
forwarding. Blind calls do not give transferring and forwarding parties the ability to announce or consult
with destination parties. These three versions of Cisco SRST use a Cisco SRST proprietary mechanism
to perform blind transfers. Cisco SRST V3.0 adds the ability to perform call transfers with consultation
using the ITU-T H.450.2 (H.450.2) standard and call forwarding using the ITU-T H.450.3 (H.450.3)
standard for H.323 calls.
Cisco SRST V3.0 provides support for IP phones to initiate call transfer and forwarding with H.450.2
and H.450.3 by using the default session application. The built-in H.450.2 and H.450.3 support that is
provided by the default session application applies to call transfers and call forwarding initiated by IP
phones, regardless of the PSTN interface type.
Note All voice gateway routers in the VoIP network must support H.450. For H.450 support, routers with
Cisco SRST must run either Cisco SRST V3.0 and higher versions or Cisco IOS Release 12.2(15)ZJ and
later releases. Routers without Cisco SRST must run either Cisco SRST V2.1 and higher versions or
Cisco IOS Release 12.2(11)YT and later releases. SIP phones does not support this feature.
For more information about the default session application, see the Default Session Application
Enhancements document.
For configuration information, see the “Enabling Consultative Call Transfer and Forward Using H.450.2
and H.450.3 with Cisco SRST 3.0” section on page 200.
Customized System Message for Cisco Unified IP Phones

The display message that appears on Cisco Unified IP Phone 7905G, Cisco Unified IP Phone 7940G,
Cisco Unified IP Phone 7960G, and Cisco Unified IP Phone 7910 units when they are in fallback mode
can be customized. The new system message command allows you to edit these display messages on a
per-router basis. The custom system message feature supports English only.
For further information, see the “Configuring Customized System Messages for Cisco Unified IP
Phones” section on page 146.
Dual-Line Mode
A new keyword that was added to the max-dn command allows you to set IP phones to dual-line mode.
Each dual-line IP phone must have one voice port and two channels to handle two independent calls.
This mode enables call waiting, call transfer, and conference functions on a single ephone-dn (ephone
directory number). There is a maximum number of DNs available during Cisco SRST fallback. The
max-dn command affects all IP phones on a Cisco SRST router.
For configuration information, see the “Configuring Dual-Line Phones” section on page 148.

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E1 R2 Signaling Support
Cisco SRST V3.0 supports E1 R2 signaling. R2 signaling is an international signaling standard that is
common to channelized E1 networks; however, there is no single signaling standard for R2. The ITU-T
Q.400-Q.490 recommendation defines R2, but a number of countries and geographic regions implement
R2 in entirely different ways. Cisco Systems addresses this challenge by supporting many localized
implementations of R2 signaling in its Cisco IOS software.
The Cisco Systems E1 R2 signaling default is ITU, which supports the following countries: Denmark,
Finland, Germany, Russia (ITU variant), Hong Kong (ITU variant), and South Africa (ITU variant). The
expression “ITU variant” means there are multiple R2 signaling types in the specified country, but Cisco
supports the ITU variant.
Cisco Systems also supports specific local variants of E1 R2 signaling in the following regions,
countries, and corporations:
• Argentina
• Australia
• Bolivia
• Brazil
• Bulgaria
• China
• Colombia
• Costa Rica
• East Europe (includes Croatia, Russia, and Slovak Republic)
• Ecuador (ITU)
• Ecuador (LME)
• Greece
• Guatemala
• Hong Kong (uses the China variant)
• Indonesia
• Israel
• Korea
• Laos
• Malaysia
• Malta
• New Zealand
• Paraguay
• Peru
• Philippines
• Saudi Arabia
• Singapore
• South Africa (Panaftel variant)

iii-xxxviii
• Telmex Corporation (Mexico)

• Telnor Corporation (Mexico)
• Thailand
• Uruguay
• Venezuela
• Vietnam
European Date Formats

The date format on Cisco IP phone displays can be configured with the following two additional formats:
• yy-mm-dd (year-month-day)
• yy-dd-mm (year-day-month)
For configuration information, see the “Configuring IP Phone Clock, Date, and Time Formats” section
on page 142.
Huntstop for Dual-Line Mode

A new keyword was added to the huntstop command. The channel keyword causes hunting to skip the
secondary channel in dual-line configuration if the primary line is busy or does not answer.
For configuration information, see the “Configuring Dial-Peer and Channel Hunting” section on
page 196.
Music-on-Hold for Multicast from Flash Files

Cisco SRST can be configured to support continuous multicast output of MOH from a flash MOH file
in flash memory.
For more information, see the “Defining XML API Schema” section on page 228.
Ringing Timeout Default

A ringing timeout default can be configured for extensions on which no-answer call forwarding has not
been enabled. Expiration of the timeout causes incoming calls to return a disconnect code to the caller.
This mechanism provides protection against hung calls for inbound calls received over interfaces such
as Foreign Exchange Office (FXO) that do not have forward-disconnect supervision. For more
information, see the “Configuring the Ringing Timeout Default” section on page 198.
Secondary Dial Tone

A secondary dial tone is available for Cisco Unified IP Phones running Cisco SRST. The secondary dial
tone is generated when a user dials a predefined PSTN access prefix. An example would be the different
dial tone heard when a designated number is pressed to reach an outside line.

iii-xxxix
The secondary dial tone is created through the secondary dialtone command. For more information, see
the “Configuring a Secondary Dial Tone” section on page 147.
Enhancement to the show ephone Command

The show ephone command is enhanced to display the following:
• Configuration and status of additional phones (new keywords: 7905, 7914, 7935, ATA)
• Status of all phones with the call-forwarding all (CFA) feature enabled on at least one of their DNs
(new keyword: cfa)
For more information, see the show ephone command in Cisco Unified SRST and Cisco Unified SIP
SRST Command Reference (All Versions).
System Log Messages for Phone Registrations

Diagnostic messages are added to the system log whenever a phone registers or unregisters from Cisco
Unified SRST.
Three-Party G.711 Ad Hoc Conferencing

Cisco SRST supports three-party ad hoc conferencing using the G.711 coding technique. For
conferencing to be available, an IP phone must have a minimum of two lines connected to one or more
buttons.
For more information, see the “Enabling Three-Party G.711 Ad Hoc Conferencing” section on page 226.
Support for Cisco VG248 Analog Phone Gateway 1.2(1) and Higher
Versions
The Cisco VG248 Analog Phone Gateway is a mixed-environment solution, enabled by Cisco AVVID
(Architecture for Voice, Video and Integrated Data), that allows organizations to support their legacy
analog devices while taking advantage of the new opportunities afforded through the use of IP telephony.
The Cisco VG248 is a high-density gateway for using analog phones, fax machines, modems, voice-mail
systems, and speakerphones within an enterprise voice system based on Cisco Unified CM.
During Cisco Unified CM fallback, Cisco SRST considers the Cisco VG248 to be a group of Cisco
Unified IP Phones. Cisco Unified SRST counts each of the 48 ports on the Cisco VG248 as a separate
Cisco Unified IP Phone. Support for Cisco VG248 Version 1.2(1) and higher versions is also available
in Cisco Unified SRST Version 2.1.
For more information, see Cisco VG248 Analog Phone Gateway Data Sheet and
Cisco VG248 Analog Phone Gateway Version 1.2(1) Release Notes.

• Additional Language Options for IP Phone Display, page xlii

iii-xl
• Cisco SRST Aggregation, page xlii

• Cisco ATA 186 and ATA 188 Support, page xlii
• Cisco Unified IP Phone 7902G Support, page xlii

iii-xli
• Cisco Unified IP Phone Expansion Module 7914 Support, page xliii

• Enhancement to the dialplan-pattern Command, page xliii
Note For information about Cisco Unified IP phones, see the Cisco Unified IP Phone 7900 Series
documentation.
Additional Language Options for IP Phone Display

Displays for the Cisco Unified IP Phone 7940G and Cisco Unified IP Phone 7960G can be configured
with ISO-3166 codes for the following countries:
• France
• Germany
• Italy
• Portugal
• Spain
• United States
Note This feature is available only in Cisco Unified SRST running under Cisco Unified CM V3.2.
For configuration information, see the “Configuring IP Phone Language Display” section on page 144.
Cisco SRST Aggregation

For systems running Cisco Unified CM 3.3(2) and later versions, the restriction of running Cisco SRST
on a default gateway was removed. Multiple SRST routers can be used to support additional phones.
Note that dial peers and dial plans need to be carefully planned and configured for call transfer and
forwarding to work properly.
Cisco ATA 186 and ATA 188 Support

The Cisco ATA analog telephone adaptors are handset-to-Ethernet adaptors that allow regular analog
telephones to operate on IP-based telephony networks. Cisco ATAs support two voice ports, each with
an independent telephone number. The Cisco ATA 188 also has an RJ-45 10/100BASE-T data port.
Cisco SRST supports Cisco ATA 186 and Cisco ATA 188 using Skinny Client Control Protocol (SCCP)
for voice calls only.
Cisco Unified IP Phone 7902G Support

The Cisco Unified IP Phone 7902G is an entry-level IP phone that addresses the voice communications
needs of a lobby, laboratory, manufacturing floor, hallway, or other area where only basic calling
capability is required.

iii-xlii
The Cisco Unified IP Phone 7902G is a single-line IP phone with fixed feature keys that provide
one-touch access to the redial, transfer, conference, and voice-mail access features. Consistent with other
Cisco IP phones, the Cisco Unified IP Phone 7902G supports inline power, which allows the phone to
receive power over the LAN. This capability gives the network administrator centralized power control
and thus greater network availability.

The Cisco Unified IP Phone 7905G is a basic IP phone that provides a core set of business features. It
provides single-line access and four interactive soft keys that guide a user through call features and
functions via the pixel-based liquid crystal display (LCD). The graphic capability of the display presents
calling information, intuitive access to features, and language localization in future firmware releases.
The Cisco Unified IP Phone 7905G supports inline power, which allows the phone to receive power over
the LAN.

The Cisco Unified IP Phone 7912G provides core business features and addresses the communication
needs of a cubicle worker who conducts low to medium telephone traffic. Four dynamic soft keys
provide access to call features and functions. The graphic display shows calling information and allows
access to features.
The Cisco Unified IP Phone 7912G supports an integrated Ethernet switch, providing LAN connectivity
to a local PC. In addition, the Cisco Unified IP Phone 7912G supports inline power, which allows the
phone to receive power over the LAN. This capability gives the network administrator centralized power
control and thus greater network availability. The combination of inline power and Ethernet switch
support reduces cabling needs to a single wire to the desktop.
Cisco Unified IP Phone Expansion Module 7914 Support

The Cisco Unified IP Phone 7914 Expansion Module attaches to your Cisco Unified IP Phone 7960G,
adding 14 line appearances or speed-dial numbers to your phone. You can attach one or two expansion
modules to your IP phone. When you use two expansion modules, you have 28 additional line
appearances or speed-dial numbers or a total of 34 line appearances or speed-dial numbers.
Enhancement to the dialplan-pattern Command

A new keyword was added to the dialplan-pattern command. The extension-pattern keyword sets an
extension number’s leading digit pattern when it is different from the E.164 telephone number’s leading
digits defined in the pattern variable. This enhancement allows manipulation of IP phone abbreviated
extension number prefix digits. See the dialplan-pattern command in Cisco Unified SRST and Cisco
Unified SIP SRST Command Reference (All Versions).

iii-xliii

Cisco SRST Version 2.02 introduced the new features described in the following sections:
• Cisco Unified IP Phone Conference Station 7935 Support, page xliv
• Increase in Directory Numbers, page xliv
• Cisco Unity Voicemail Integration Using In-Band DTMF Signaling Across the PSTN and BRI/PRI,
page xlv
Cisco Unified IP Phone Conference Station 7935 Support

The Cisco IP Conference Station 7935 is an IP-based, full-duplex hands-free conference station for use
on desktops and offices and in small-to-medium-sized conference rooms. This device attaches a
Cisco Catalyst 10/100 Ethernet switch port with a simple RJ-45 connection and dynamically configures
itself to the IP network via the DHCP. Other than connecting the Cisco 7935 to an Ethernet switch port,
no further administration is necessary. The Cisco 7935 dynamically registers to
Cisco Unified CM for connection services and receives the appropriate endpoint phone number and any
software enhancements or personalized settings, which are preloaded within Cisco Unified CM.
The Cisco Unified IP Phone 7935 provides three soft keys and menu navigation keys that guide a user
through call features and functions. The Cisco Unified IP Phone 7935 also features a pixel-based LCD
display. The display provides features such as date and time, calling party name, calling party number,
digits dialed, and feature and line status. No configuration is necessary.
Increase in Directory Numbers

Table iii-6 shows the increases in directory numbers.
Table iii-6 Increases in Directory Numbers in Cisco IOS Release 12.2(11)T
Increase in Maximum Directory Number

Cisco Router Maximum Phones From To
Cisco 1751 24 96 120
Cisco 1760 24 96 120
Cisco 2600XM 24 96 120
Cisco 2691 72 216 288
Cisco 3640 72 216 288
Cisco 3660 240 720 960
Cisco 3725 144 432 576
Cisco 3745 240 720 960

iii-xliv
Where to Go Next
Cisco Unity Voicemail Integration Using In-Band DTMF Signaling Across

the PSTN and BRI/PRI
Cisco Unity voicemail and other voicemail systems can be integrated with Unified SRST. Voicemail
integration introduces six new commands:
• pattern direct
• pattern ext-to-ext busy
• pattern ext-to-ext no-answer
• pattern trunk-to-ext busy
• pattern trunk-to-ext no-answer
• vm-integration
Where to Go Next
Proceed to the “Setting Up the Network” section on page 115.

iii-xlv
Where to Go Next

iii-xlvi
CHAPTER 1
Cisco Unified SRST Feature Overview
This chapter describes Cisco Unified Survivable Remote Site Telephony (Cisco Unified SRST) and what
it does. It also includes information about support for Cisco Unified IP Phones and Platforms,
specifications, features, prerequisites, restrictions and where to find additional reference documents.
For the most up-to-date information about Cisco Unified IP Phone support, the maximum number of
Cisco Unified IP Phones, the maximum number of directory numbers (DNs) or virtual voice ports, and
memory requirements for Cisco Unified SRST and Cisco Unified SIP SRST, see Cisco Unified SRST
Supported Firmware, Platforms, Memory, and Voice Products.
Contents
• Cisco Unified SCCP SRST, page 1
• Cisco Unified SIP SRST, page 9
• Cisco Unified SRST Licenses
• Interface Support for Unified CME and Unified SRST, page 17
• MGCP Gateways and SRST, page 17
• Support for Cisco Unified IP Phones and Platforms, page 25
• IPv6 Support for Unified SRST SIP IP Phones, page 18
• Multicast Music On Hold, page 27
• Where to Go Next, page 30
• Additional References, page 30
• Obtaining Documentation, Obtaining Support, and Security Guidelines, page 33
Cisco Unified SCCP SRST

• Information About SCCP SRST, page 2
• Prerequisites for Configuring Cisco Unified SCCP SRST, page 4
• Restrictions for Configuring Cisco Unified SCCP SRST, page 7

1-1
Chapter 1 Cisco Unified SRST Feature Overview
Information About SCCP SRST

Cisco Unified SRST provides Cisco Unified CM with fallback support for Cisco Unified IP phones that
are attached to a Cisco router on your local network. Cisco Unified SRST enables routers to provide
call-handling support for Cisco Unified IP phones when they lose connection to remote primary,
secondary, or tertiary Cisco Unified CM installations or when the WAN connection is down.
Cisco Unified CM supports Cisco Unified IP phones at remote sites attached to Cisco multiservice
routers across the WAN. Prior to Cisco Unified SRST, when the WAN connection between a router and
the Cisco Unified CM failed or when connectivity with Cisco Unified CM was lost for some reason,
Cisco Unified IP phones on the network became unusable for the duration of the failure. Cisco Unified
SRST overcomes this problem and ensures that the Cisco Unified IP phones offer continuous (although
minimal) service by providing call-handling support for Cisco Unified IP phones directly from the Cisco
Unified SRST router. The system automatically detects a failure and uses Simple Network Auto
Provisioning (SNAP) technology to autoconfigure the branch office router to provide call processing for
Cisco Unified IP phones that are registered with the router. When the WAN link or connection to the
primary Cisco Unified CM is restored, call handling reverts back to the primary Cisco Unified CM.
When Cisco Unified IP phones lose contact with primary, secondary, and tertiary Cisco Unified CM,
they must establish a connection to a local Cisco Unified SRST router to sustain the call-processing
capability necessary to place and receive calls. The Cisco Unified IP phone retains the IP address of the
local Cisco Unified SRST router as a default router in the Network Configuration area of the Settings
menu. The Settings menu supports a maximum of five default router entries; however, Cisco Unified CM
accommodates a maximum of three entries. When a secondary Cisco Unified CM is not available on the
network, the local Cisco Unified SRST Router's IP address is retained as the standby connection for
Cisco Unified CM during normal operation.
Note Cisco Unified CM fallback mode telephone service is available only to those Cisco Unified IP
phones that are supported by a Cisco Unified SRST router. Other Cisco Unified IP phones on the
network remain out of service until they re-establish a connection with their primary, secondary,
or tertiary Cisco Unified CM.
Typically, it takes three times the keepalive period for a phone to discover that its connection to Cisco
Unified CM has failed. The default keepalive period is 30 seconds. If the phone has an active standby
connection established with a Cisco Unified SRST router, the fallback process takes 10 to 20 seconds
after connection with Cisco Unified CM is lost. An active standby connection to a Cisco Unified SRST
router exists only if the phone has the location of a single Cisco Unified CM in its Unified
Communications Manager list. Otherwise, the phone activates a standby connection to its secondary
Cisco Unified CM.
Note The time it takes for a Cisco Unified IP Phone to fallback to the SRST router can vary depending
on the phone type. Phones such as the Cisco 7902, Cisco 7905, and Cisco 7912 can take
approximately 2.5 minutes to fallback to SRST mode.
If a Cisco Unified IP phone has multiple Cisco Unified CM in its Cisco Unified CM list, it progresses
through its list of secondary and tertiary Cisco Unified CM before attempting to connect with its local
Cisco Unified SRST router. Therefore, the time that passes before the Cisco Unified IP phone eventually
establishes a connection with the Cisco Unified SRST router increases with each attempt to contact to a
Cisco Unified CM. Assuming that each attempt to connect to a Cisco Unified CM takes about 1 minute,
the Cisco Unified IP phone in question could remain offline for 3 minutes or more following a WAN link
failure.

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Note During a WAN connection failure, when Cisco Unified SRST is enabled, Cisco Unified IP
phones display a message informing you that they are operating in Cisco Unified CM fallback
mode. For example, the Cisco Unified IP Phone 7960G and Cisco Unified IP Phone 7940G
display a "CM Fallback Service Operating" message, and the Cisco Unified IP Phone 7910
displays a "CM Fallback Service" message when operating in Cisco Unified CM fallback mode.
When the Cisco Unified CM is restored, the message goes away and full Cisco Unified IP phone
functionality is restored.
While in Cisco Unified CM fallback mode, Cisco Unified IP phones periodically attempt to re-establish
a connection with Cisco Unified CM at the central office. Generally, the default time that Cisco Unified
IP phones wait before attempting to re-establish a connection to a remote Cisco Unified CM is 120
seconds. The time can be changed in Cisco Unified CM; see the "Device Pool Configuration Settings"
chapter in the appropriate Cisco Unified CM Administration Guide. A manual reboot can immediately
reconnect Cisco Unified IP phones to Cisco Unified CM.
When a connection is re-established with Cisco Unified CM, Cisco Unified IP phones automatically
cancel their registration with the Cisco Unified SRST Router. However, if a WAN link is unstable, Cisco
Unified IP phones can bounce between Cisco Unified CM and Cisco Unified SRST. A Cisco Unified IP
phone cannot re-establish a connection with the primary Cisco Unified CM at the central office if it is
currently engaged in an active call.
Cisco Unified SRST supports the following call combinations:

• SCCP phone to SCCP phone
• SCCP phone to PSTN/router voice-port
• SCCP phone to WAN VoIP using SIP or H.323
• SIP phone to SIP phone
• SIP phone to PSTN / router voice-port
• SIP phone to Skinny Client Control Protocol (SCCP) phone
• SIP phone to WAN VoIP using SIP
Figure 1-1 shows a branch office with several Cisco Unified IP phones connected to a Cisco Unified
SRST router. The router provides connections to both a WAN link and the PSTN. Typically, the
Cisco Unified IP phones connect to their primary Cisco Unified Communications Manager at the central
office via the WAN link. When the WAN connection is down, the Cisco Unified IP phones use the
Cisco Unified SRST router as a fallback for their primary Cisco Unified Communications Manager. The
branch office Cisco Unified IP phones are connected to the PSTN through the Cisco Unified SRST
router and are able to make and receive off-net calls.

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Figure 1-1 Branch Office Cisco Unifed IP Phones Connected to a Remote Central Cisco Unified
Communications Manage Operating in SRST Mode
Telephone Telephone
Central
Fax Cisco Unified
Communications
Manager
PSTN
V
Cisco Unified SRST IP
router network
WAN
IP IP IP Cisco IP phones disconnected
146613
PCs
On H.323 gateways for SCCP SRST, when the WAN link fails, active calls from Cisco Unified IP phones
to the PSTN are not maintained by default. Call preservation may work with the no h225 timeout
keepalive command.
Under default configuration, the H.323 gateway maintains a keepalive signal with Cisco Unified
Communications Manager and terminates H.323-to-PSTN calls if the keepalive signal fails, for
example, if the WAN link fails. To disable this behavior and help preserve existing calls from local
Cisco Unified IP phones, you can use the no h225 timeout keepalive command. Disabling the keepalive
mechanism only affects calls that will be torn down as a result of the loss of the H.225 keepalive signal.
For information regarding disconnecting a call when an inactive condition is detected, see the Media
Inactive Call Detection document.
Prerequisites for Configuring Cisco Unified SCCP SRST

Before configuring Cisco Unified SRST, you must do the following:
• An SRST feature license is required to enable the Cisco Unified SCCP SRST feature. Contact your
account representative if you have further questions. For more information about Licensing on
Unified SRST, refer Cisco Unified SRST Licenses, page 13.
• You have an account on Cisco.com to download software.
To obtain an account on Cisco.com, go to www.cisco.com and click Register at the top of the screen.

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Installing Cisco Unified Communications Manager

When installing Cisco Unified Communications Manager, consider the following:
• See the installation instructions for your version in the Cisco Unified Communications Manager
Install and Upgrade Guides.
• Integrate Cisco Unified SRST with Cisco Unified Communications Manager. Integration is
performed from Cisco Unified Communications Manager. See the “Integrating Cisco Unified
SCCP SRST with Cisco Unified Communications Manager” section on page 5.
Installing Cisco Unified SCCP SRST

Cisco Unified SRST versions have different installation instructions:
• Installing Cisco Unified SRST V3.0 and Later Versions, page 5
• Installing Cisco Unified SRST V2.0 and V2.1, page 5
• Installing Cisco Unified SRST V1.0, page 5
To update Cisco Unified SRST, follow the installation instructions described in this section.
Installing Cisco Unified SRST V3.0 and Later Versions
Install the Cisco IOS software release image containing the Cisco SRST or Cisco Unified SRST version
that is compatible with your Cisco Unified Communications Manager version. See the “Cisco Unified
Communications Manager Compatibility” section on page 27. Cisco IOS software can be downloaded
from the Cisco Software Center at http://www.cisco.com/public/sw-center/.
Cisco SRST and Cisco Unified SRST can be configured to support continuous multicast output of music-
on-hold (MOH) from a flash MOH file in flash memory. For more information, see the “Defining XML
API Schema” section on page 228. If you plan to use MOH, go to the Technical Support Software
Download site at http://www.cisco.com/cgi-bin/tablebuild.pl/ip-iostsp and copy the music-on-hold.au
file to the flash memory on your Cisco SRST or Cisco Unified SRST router.
Installing Cisco Unified SRST V2.0 and V2.1
Download and install Cisco SRST V2.0 or Cisco SRST V2.1 from the Cisco Software Center at
http://www.cisco.com/public/sw-center/.
Installing Cisco Unified SRST V1.0
Cisco SRST V1.0 runs with Cisco Communications Manager V3.0.5 only. It is recommended that you
upgrade to the latest Cisco Unified Communications Manager and Cisco Unified SRST versions.
Integrating Cisco Unified SCCP SRST with Cisco Unified Communications Manager
There are two procedures for integrating Cisco Unified SRST with Cisco Unified
Communications Manager. Procedure selection depends on the Cisco Unified
Communications Manager version that you have.
If You Have Cisco Communications Manager V3.3 or Later Versions
If you have Cisco Communications Manager V3.3 or later versions, you must create an SRST reference
and apply it to a device pool. An SRST reference is the IP address of the Cisco Unified SRST Router.

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Step 1 Create an SRST reference.

a. From any page in Cisco Unified Communications Manager, click System and SRST.
b. On the Find and List SRST References page, click Add a New SRST Reference.
c. On the SRST Reference Configuration page, enter a name in the SRST Reference Name field and
the IP address of the Cisco SRST router in the IP Address field.
d. Click Insert.
Step 2 Apply the SRST reference or the default gateway to one or more device pools.
a. From any page in Cisco Unified Communications Manager, click System and Device Pool.
b. On the Device Pool Configuration page, click on the required device pool icon.
c. On the Device Pool Configuration page, choose an SRST reference or “Use Default Gateway” from
the SRST Reference field’s menu.
If You Have Cisco Unified Communications Manager Version Prior to V3.3
If you have firmware versions that enable Cisco Unified SRST by default, no additional configuration is
required on Cisco Unified Communications Manager to support Cisco Unified SRST. If your firmware
versions disable Cisco Unified SRST by default, you must enable Cisco Unified SRST for each phone
configuration.
Step 1 Go to the Cisco Unified Communications Manager Phone Configuration page.

a. From any page in Cisco Unified Communications Manager, click Device and Phone.
b. In the Find and List Phones page, click Find.
c. After a list of phones appears, click on the required device name.
d. The Phone Configuration appears.
Step 2 In the Phone Configuration page, go to the Product Specific Configuration section at the end of the page,
choose Enabled from the Cisco Unified SRST field’s menu, and click Update.
Step 3 Go to the Phone Configuration page for the next phone and choose Enabled from the Cisco Unified
SRST field’s menu by repeating Step 1 and Step 2.

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Restrictions for Configuring Cisco Unified SCCP SRST

Table 1-1 provides a history of restrictions from Cisco SCCP SRST Version 1.0 to the present version of
Cisco Unified SCCP SRST.
Table 1-1 Restrictions from Cisco SCCP SRST from the Present Version to Version 1.0
Cisco Unified
SRST Cisco IOS
Version Release Restrictions
Version 4.1 12.4.(15)T • Enhanced 911 Services for Cisco Unified SRST does not interface with the Cisco
Emergency Responder.
• The information about the most recent phone that called 911 is not preserved after
a reboot of Cisco Unified SRST.
• Cisco Emergency Responder does not have access to any updates made to the
emergency call history table when remote IP phones are in Cisco Unified SRST
fallback mode. Therefore, if the PSAP calls back after the Cisco Unified IP phones
register back to Cisco Unified Communications Manager, Cisco Emergency
Responder will not have any history of those calls. As a result, those calls will not
get routed to the original 911 caller. Instead, the calls are routed to the default
destination that is configured on Cisco Emergency Responder for the
corresponding ELIN.
• For Cisco Unified Wireless IP Phone 7920 and 7921, a caller’s location can only be
determined by the static information configured by the system administrator. For
more information, see the Precautions for Mobile Phones in Configuring Enhanced
911 Services.
• The extension numbers of 911 callers can be translated to only two emergency
location identification numbers (ELINs) for each emergency response location
(ERL).
• Using ELINs for multiple purposes can result in unexpected interactions with
existing Cisco Unified SRST features. These multiple uses of an ELIN can include
configuring an ELIN for use as an actual phone number (ephone-dn, voice register
dn, or FXS destination-pattern), a Call Pickup number, or an alias rerouting
number. For more information, see the Multiple Usages of an ELIN in Configuring
Enhanced 911 Services.
• There are a number of other ways that your configuration of Enhanced 911 Services
can interact with existing Cisco Unified SRST features and cause unexpected
behavior. For a complete description of interactions between Enhanced 911
Services and existing Cisco Unified SRST features, see the Interactions with
Existing Cisco Unified CME Features in Configuring Enhanced 911 Services.

1-7
Table 1-1 Restrictions from Cisco SCCP SRST from the Present Version to Version 1.0 (continued)
Cisco Unified
SRST Cisco IOS
Version 4.0 12.4(4)XC • All of the restrictions in Cisco SRST Version 1.0.
Version 3.4 12.4(4)T • Caller-id display on supported Cisco Unified IP phones: SIP phones in fallback
Version 3.2 12.3(11)T mode displays the name and number of the caller. SCCP phones in fallback mode
display only the caller-id number assigned to the line; the caller-ID name
Version 3.1 12.3(7)T configuration for SCCP phones is not preserved during SRST fallback.
Version 3.0 12.2(15)ZJ • Call transfer is supported only on the following:
12.3(4)T – VoIP H.323, VoFR, and VoATM between Cisco gateways running Cisco IOS
Release 12.2(11)T and using the H.323 nonstandard information element
12.2(15)T – FXO and FXS loop-start (analog)
Version 2.1
12.2(13)T – FXO and FXS ground-start (analog)
Version 2.02
12.2(11)T – Ear and mouth (E&M) (analog) and DID (analog)
Version 2.01
12.2(8)T1 – T1 channel-associated signaling (CAS) with FXO and FXS ground-start
Version 2.0
12.2(8)T signaling
12.2(2)XT – T1 CAS with E&M signaling

– All PRI and BRI switch types
• The following Cisco Unified IP Phone function keys are dimmed because they are
not supported during SRST operation:
– MeetMe
– GPickUp (group pickup)
– Park
– Confrn (conference)
• Although the Cisco IAD2420 series integrated access devices (IADs) support the
Cisco Unified SRST feature, this feature is not recommended as a solution for
enterprise branch offices.
Version 1.0 12.2(2)XB • Does not support first generation Cisco Unified IP phones, such as Cisco IP Phone
12.2(2)XG 30 VIP and Cisco IP Phone 12 SP+.
• Does not support other Cisco Unified Communications Manager applications or
12.1(5)YD
services: Cisco IP SoftPhone, Cisco One: Voice and Unified Messaging
Application, or Cisco IP Contact Center.
• Does not support Centralized Automatic Message Accounting (CAMA) trunks on
the Cisco 3660 routers.
Note If you are in one of the states in the United States of America where there is a
regulatory requirement for CAMA trunks to interface to 911 emergency
services, and you would like to connect more than 48 Cisco Unified IP phones
to the Cisco 3660 multiservice routers in your network, contact your local Cisco
account team for help in understanding and meeting the CAMA regulatory
requirements.

1-8
Cisco Unified SIP SRST

• Information About SIP SRST, page 9
• Prerequisites for Configuring Cisco Unified SIP SRST, page 9
• Restrictions for Configuring Cisco Unified SIP SRST, page 9
Information About SIP SRST

This guide describes Cisco Unified SRST functionality for SIP networks. Cisco Unified SIP SRST
provides backup to an external SIP call control (IP-PBX) by providing basic registrar and redirect server
or back-to-back user agent (B2BUA) services. These services are used by a SIP IP phone in the event of
a WAN connection outage when the SIP phone is unable to communicate with its primary SIP proxy.
Cisco Unified SIP SRST can support SIP phones with standard RFC 3261 feature support locally and
across SIP WAN networks. With Cisco Unified SIP SRST, SIP phones can place calls across SIP
networks in the same way as SCCP phones.
Cisco Unified SIP SRST supports the following call combinations:
SIP proxy, registrar, and B2BUA servers are key components of a SIP VoIP network. These servers are
usually located in the core of a VoIP network. If SIP phones located at remote sites at the edge of the
VoIP network lose connectivity to the network core (because of a WAN outage), they may be unable to
make or receive calls. Cisco Unified SIP SRST functionality on a SIP PSTN gateway provides service
reliability for SIP-based IP phones in the event of a WAN outage. Cisco Unified SIP SRST enables the
SIP IP phones to continue to make and receive calls to and from the PSTN and also to make and receive
calls to and from other SIP IP phones.
To see a branch office Cisco Unifed IP Phones connected to a remote central Cisco Unified CM
Operating in SRST mode, see Figure 1-1.
Note Cisco Unity Express (CUE) interworking is not supported with secure SIP SRST.
Prerequisites for Configuring Cisco Unified SIP SRST

Before configuring Cisco Unified SIP SRST, you must do the following:
• An SRST feature license is required to enable the Cisco Unified SIP SRST feature. Contact your
account representative if you have further questions. For more information about Licensing on
Unified SRST, refer Cisco Unified SRST Licenses, page 13.
Restrictions for Configuring Cisco Unified SIP SRST

Table 1-2 provides a history of restrictions from Cisco SIP SRST Version 3.0 to the present version of
Cisco Unified SIP SRST.

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Table 1-2 Restrictions from Cisco SIP SRST from the Present Version to Version 3.0
Cisco Unified
SRST Cisco IOS
Version 8.0 15.1(1)T • SIP phones may be configured on the Cisco Unified CM with an Authenticated
device security mode. The Cisco Unified CM ensures integrity and authentication
for the phone using a TLS connection with NULL-SHA cipher for signaling. If
such an Authenticated SIP phone fails over to the Cisco Unified SRST device, and
if the Cisco Unified CM and SRST device are configured to support secure SIP
SRST, it will register using TCP instead of TLS/TCP, thus disabling the
Authenticated mode until the phone fails back to the Cisco Unified CM.

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Table 1-2 Restrictions from Cisco SIP SRST from the Present Version to Version 3.0 (continued)
Cisco Unified
SRST Cisco IOS
Version 4.1 12.4.(15)T • Cisco Unified SRST does not support BLF speed-dial notification, call forward all
synchronization, dial plans, directory services, or music-on-hold (MOH).
• Prior to SIP phone load 8.0, SIP phones maintained dual registration with both
Cisco Unified Communications Manager and Cisco Unified SRST simultaneously.
In SIP phone load 8.0 and later versions, SIP phones use keepalive to maintain a
connection with Cisco Unified SRST during active registration with
Cisco Unified Communications Manager. Every two minutes, a SIP phone sends a
keepalive message to Cisco Unified SRST. Cisco Unified SRST responds to this
keepalive with a 404 message. This process repeats until fallback to
Cisco Unified SRST occurs. After fallback, SIP phones send a keepalive message
every two minutes to Cisco Unified Communications Manager while the phones
are registered with Cisco Unified SRST. Cisco Unified SRST continues to support
dual registration for SIP phone loads older than 8.0.
• Enhanced 911 Services for Cisco Unified SRST does not interface with the
Cisco Emergency Responder.
• The information about the most recent phone that called 911 is not preserved after
a reboot of Cisco Unified SRST.
• Cisco Emergency Responder does not have access to any updates made to the
emergency call history table when remote IP Phones are in Cisco Unified SRST
fallback mode. Therefore, if the PSAP calls back after the Cisco Unified IP Phones
register back to Cisco Unified Communications Manager,
Cisco Emergency Responder will not have any history of those calls. As a result,
those calls will not get routed to the original 911 caller. Instead, the calls are routed
to the default destination that is configured on Cisco Emergency Responder for the
corresponding ELIN.
• For Cisco Unified Wireless 7920 and 7921 IP Phones, a caller’s location can only
be determined by the static information configured by the system administrator. For
more information, see Precautions for Mobile Phones in Configuring Enhanced
911 Services.
• The extension numbers of 911 callers can be translated to only two emergency
location identification numbers (ELINs) for each emergency response location
(ERL).
• Using ELINs for multiple purposes can result in unexpected interactions with
existing Cisco Unified SRST features. These multiple uses of an ELIN can include
configuring an ELIN for use as an actual phone number (ephone-dn, voice register
dn, or FXS destination-pattern), a Call Pickup number, or an alias rerouting
number. For more information, see Multiple Usages of an ELIN in Configuring
Enhanced 911 Services.
• There are a number of other ways that your configuration of Enhanced 911 Services
can interact with existing Cisco Unified SRST features and cause unexpected
behavior. For a complete description of interactions between Enhanced 911
Services and existing Cisco Unified SRST features, see the Interactions with
Existing Cisco Unified CME Features in Configuring Enhanced 911 Services..

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Table 1-2 Restrictions from Cisco SIP SRST from the Present Version to Version 3.0 (continued)
Cisco Unified
SRST Cisco IOS
Version 4.0 12.4(4)XC Not Supported
Version 3.4 12.4(4)T • MOH is not supported for a call hold invoked from a SIP phone. A caller hears only
silence when placed on hold by a SIP phone.
Version 3.2 12.3(11)T
• As of Cisco IOS Release 12.4(4)T, bridged call appearance, find-me, incoming call
Version 3.1 12.3(7)T screening, paging, SIP presence, call park, call pickup, and SIP location are not
Version 3.0 12.2(15)ZJ supported.
12.3(4)T • SIP-NAT is not supported.
• Cisco Unity Express is not supported.
• Transcoding is not supported.
Phone Features
• For call waiting to work on the Cisco ATA and Cisco IP Phone 7912 and Cisco
Unified IP Phone 7905G with a 1.0(2) build, the incoming call leg should be
configured with the G.711 codec.
Note Cisco Unified IP Phone 7905G, Cisco Unified IP Phone 7912G, and Cisco
Analog Telephone Adaptor (ATA) 186 are not capable of dual registration; thus
they are not supported and have limited functionality with Cisco Unified SIP
SRST.
General
• Call detail records (CDRs) are only supported by standard IOS RADIUS support;
CDRs are not supported otherwise.
• All calls must use the same codec, either G.729r8 or G.711.
• Calls that have been transferred cannot be transferred a second time.
• URL dialing is not supported. Only number dialing is supported.
• The SIP registrar functionality provided by Cisco Unified SIP SRST provides no
security or authentication services.
• SIP IP phones that do not support dual concurrent registration with both their
primary and their backup SIP proxy or registrar may be unable to receive incoming
calls from the Cisco Unified SIP SRST gateway during a WAN outage. These
phones may take a significant amount of time to discover that their primary SIP
proxy or registrar is unreachable before they initiate a fallback registration to their
backup proxy or registrar (the SIP SRST gateway).
• SIP-phone-to-SIP-trunk support requires Refer and 302/300 Redirection to be
supported by the SIP trunk (Version 3.0).

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Cisco Unified SRST Licenses

You should purchase a Cisco Unified SRST license that entitle you to use Unified SRST. You can
purchase:
• Cisco Unified SRST Permanent License or
• Collaboration Professional Suite License
• Cisco Smart License
Cisco Unified SRST Permanent License

When you purchase a Cisco Unified SRST permanent license, the permanent license is installed on the
device when the product is shipped to you. A permanent license never expires and you will gain access
to that particular feature set for the lifetime of the device across all IOS release. If you purchase a
permanent license for Cisco Unified SRST, you do not have to go through the Evaluation Right to Use
and Right To Use (RTU) licensing processes for using the features. If you want to purchase a CME-SRST
license for your existing device, you have to go through the RTU licensing process for using the features.
There is no change in the existing process for purchasing the license.
The Cisco Unified SRST permanent license is available in the form of an XML cme-locked3 file. You
should get the XML file and load it in the flash memory of the device. To install the permanent license
from the command prompt, use the license install flash0:cme-locked3 command. The cme-locked3 is
the xml file of the license.
Collaboration Professional Suite License

Collaboration Professional is a new suite of licenses. The Collaboration Professional suite can be
purchased either as a permanent license or an RTU license.
Collaboration Professional Suite Permanent License —When you purchase the Collaboration
Professional Suite license, by default, the Cisco Unified SRST licenses are delivered as part of the
Collaboration Professional Suite. You do not have to separately install and activate the Cisco Unified
SRST license. The Collaboration Professional Suite permanent license is available in the form of an
XML file. You should get the XML file and load it in the flash memory of the device. To install the
permanent license from the command prompt, use the license install flash:lic_name command.
Collaboration Professional Suite RTU License—When you purchase the Collaboration Professional
Suite RTU license, you do not have to go through the Evaluation Right to Use process. However, you
have to go through the RTU licensing process for using the Cisco Unified SRST features. To install the
Collaboration Professional Suite RTU license from the command prompt, use the license install
flash0:colla_pro command. To activate the license, use the license boot module c2951
technology-package collabProSuitek9 command.
Also, you have to:
• Choose an appropriate Cisco Unified SRST version. Each SRST version supports a specific set of
IP phones, memory requirements, features, and DNs. See the “Platform and Memory Support”
section on page 26 and the “Restrictions for Configuring Cisco Unified SCCP SRST” section on
page 7.

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• Choose an appropriate phoneload. SRST only supports certain phoneloads that have been tested with
the various Cisco Unified Communications Manager versions. For the most up-to-date phoneloads,
see the
Cisco Unified SRST Supported Firmware, Platforms, Memory, and Voice Products.
If you have Cisco Unified Communications Manager already installed, verify that your version of
Cisco Unified Communications Manager is compatible with your Cisco Unified SRST release. See the
“Cisco Unified Communications Manager Compatibility” section on page 27.
Cisco Smart License

From Release 12.1 onwards, Unified SRST supports Smart Licensing, apart from the existing CSL
licensing model. Smart Licensing is supported only on Cisco 4000 Series Integrated Services Router.
Depending on the technology package available on the router, licenses such as UCK9 and Security are
supported using Smart Licensing.
Smart Software Licensing is Cisco's new licensing model that gives you visibility into license ownership
and consumption. Smart licenses can be managed by a cloud-based deployment model, namely Cisco
Smart Software Manager (CSSM) or an on-prem software, Smart Software Manager satellite. Unified
SRST is supported by both CSSM and satellite. Your access to the customer Smart Account residing on
CSSM is authenticated using valid Cisco credentials. With the Smart Licensing support for Unified
SRST, your device can register with CSSM or Cisco Smart Software Manager satellite. You can access
your Licenses at the Cisco Software Central.
Unified SRST needs to register with CSSM or Cisco Smart Software Manager satellite to report license
consumption. You can register Unified SRST to a Virtual Account within a Smart Account by generating
a token ID from it, and pasting it to the underlying platform, Cisco 4000 Series Integrated Services
Router. Once the token is generated, it can be used to register many other product instances in your
network.
On the Unified SRST router, you need to ensure that the call home feature is not disabled. Also, Smart
Licensing should be enabled at the router using the CLI command license smart enable. Use the no form
of the command to disable Smart Licensing.
For more information on configuring Smart Licensing in your router, see Cisco 4000 Series ISRs
Software Configuration Guide. For more information on configuring Call Home for your devices, see
Configure Call Home, page 16. Once Smart Licensing is enabled, the router enters a 90-day evaluation
period that persist until it registers to CSSM or the Cisco Smart Software Manager satellite.
You can register the router to CSSM or Cisco Smart Software Manager satellite with the token ID. To
register the device (Unified SRST router) with CSSM or Cisco Smart Software Manager satellite, use
the CLI command license smart register idtoken. For information on registering the device with
CSSM, see Device Registration, Software Activation Configuration Guide, Cisco IOS Release 15M&T.
Upon successful registration, Unified SRST is in Registered status. As part of the registration process,
the router sends an authorization request, indicating the number of phone endpoints defined by the
max-pool, for SIP SRST, and max-ephone, for SCCP SRST. Based on the licenses in the Smart
Account, CSSM or Cisco Smart Software Manager satellite responds with one of the defined statuses
such as Authorized (using less than or equal to the number of licenses provisioned in CSSM or Cisco
Smart Software Manager satellite) or Out-of-Compliance (using more than it has licenses for).
The license limit on Unified SRST is restricted by the maximum platform limit defined for the Unified
SRST router (a cumulative sum of phones configured under max-pool and max-ephone). Hence, the
license usage count cannot exceed the platform limit set for the Unified SRST router even when the

1-14
cumulative sum of phones configured under max-pool and max-ephone exceeds the defined platform
limit. For more details on the platform limits defined for Unified SRST, see Cisco Unified
SRST/E-SRST 12.1 Supported Firmware, Platforms, Memory, and Voice Products.
CSSM or Smart Software Manager satellite reports license consumption submitted by the platform in its
User Interface (UI), and subtracts it from the available licenses in the Virtual Account within the Smart
Account. Unified SRST supports only one license entitlement to validate phones configured on Unified
SRST.
SRST_EP —This license type supports all phones configured on Unified SRST.
Note The SRST_EP license count reflects the total phone count of both the ephones and pools that are
configured in the Unified SRST irrespective of whether the phones are registered or not.
Unified SRST sends an authorization request when a license consumption changes or every 30 days to
let CSSM or Cisco Smart Software Manager satellite know it's still available and communicating. The
ID certificate issued to identify Unified SRST at time of registration is valid for one year, and is
automatically renewed every six months.
Note If the router does not communicate with CSSM or Cisco Smart Software Manager satellite for a period
of 90 days, the license authorization expires. When the license authorization expires, the devices
registered on Unified SRST change status to Out of Compliance.
The license count is evaluated for the number of phones configured across the routers. The CSSM
Licenses page reflects the total license count usage, the total number of licenses available for a type of
license (Quantity), number of licenses currently used (In Use), and the number of unused or over-used
licenses (Surplus/Shortage). If you do not have enough Cisco Smart licenses, you are in
Out-of-Compliance state.
For example, consider a smart account in CSSM with 50 SRST_EP licenses. If the user has a registered
Unified SRST with 20 phones configured, the CSSM licenses page reflects Quantity as 50, In Use as 20,
and Surplus as 30. For more information on Smart Software Manager, see Cisco Smart Software
Manager User Guide.
For more information on switching between CSL and Cisco Smart License, see Licensing Modes,
page 17.
The license entitlement for Unified SRST smart license is displayed on the router as follows:
Router# show license summary
Smart Licensing is ENABLED
Registration:
Status: REGISTERED
Smart Account: ABC
Virtual Account: XYZ
Export-Controlled Functionality: Not Allowed
Last Renewal Attempt: None
Next Renewal Attempt: Jun 07 12:08:10 2017 UTC
License Authorization:
Status: AUTHORIZED
Last Communication Attempt: SUCCESS
Next Communication Attempt: Apr 13 07:11:48 2017 UTC
License Usage:

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License Entitlement tag Count Status

-----------------------------------------------------------------------------
regid.2014-12.com.ci... (ISR_4351_UnifiedCommun..) 1 AUTHORIZED
regid.2016-10.com.ci... (SRST_EP) 4 AUTHORIZED
Configure Call Home

To configure the call home destination address and proxy server details for the HTTP proxy request,
perform the following steps.
Prerequisites
• Cisco Smart Software Licensing is enabled.
SUMMARY STEPS
2. call-home destination address http url
3. call-home http-proxy proxy_address port port number
4. end
DETAILED STEPS

Step 1 configure terminal Enters configuration mode.
Example:
Step 2 call-home destination address http url (Optional) Defines the destination URL to which
Call Home messages, including licensing requests
are sent. The destination URL can be the URL for
Example:
Router(config)# call-home destination address http
Transport Gateway or CSSM satellite.
http://10.22.183.117:8080/ddce/services/DDCEService The URL to the Cisco Smart Licensing production
server is set by default.
Step 3 call-home http-proxy proxy_address port port number (Optional) Specifies the proxy server for the HTTP
request.
Example:
Router(config)# call-home http-proxy 7.7.7.7 port
3218
Step 4 end Exits to privileged EXEC mode.
Example:
Router(config)# end

1-16
Interface Support for Unified CME and Unified SRST
Licensing Modes
From Unified SRST 12.1 onwards, both CSL and Smart Licensing modes are supported. That is,
customers can continue with CSL by not enabling Smart Licensing. Alternatively, they can enable Smart
Licensing and decide later to go back to CSL by disabling Smart Licensing with the no license smart
enable command. When you switch to CSL from the Smart Licensing mode, you need to ensure that the
End User License Agreement (EULA) is signed. CSL is not supported unless the EULA is signed. Use
the CLI command license accept end user agreement in global configuration mode to configure EULA.
To verify the status of the license issued to phones registered on Unified SRST, you can use the show
license command.
Router#show license ?
all Show license all information
status Show license status information
summary Show license summary
tech Show license tech support information
udi Show license udi information
usage Show license usage information
Restrictions
• For the Unified SRST license, the UCK9 technology package must be available if the Collaboration
Professional Suite package is not installed.
To purchase a license, go to http://www.cisco.com/cgi-bin/tablebuild.pl/ip-key. To activate cme-srst
feature license, see the Activating CME-SRST Feature License document.
Interface Support for Unified CME and Unified SRST

Unified CME and Unified SRST routers have multiple interfaces that are used for signaling and data
packet transfers. The two types of interfaces available on a Cisco router include the physical interface
and the virtual interface. The types of physical interfaces available on a router depends on its interface
processors or port adapters. Virtual interfaces are software-based interfaces that you create in the
memory of the networking device using Cisco IOS commands. When you need to configure a virtual
interface for connectivity, you can use the Loopback Interface for Unified CME and Unified SRST.
The following interfaces are supported on Unified CME and Unified SRST:
• Gigabit Ethernet Interface (IEEE 802.3z) (interface gigabitethernet)
• Loopback Interface (interface loopback)
• Fast Ethernet Interface (interface fastethernet)
MGCP Gateways and SRST

MGCP fallback is a different feature than SRST and, when configured as an individual feature, can be
used by a PSTN gateway. To use SRST as your fallback mode on an MGCP gateway, SRST and MGCP
fallback must both be configured on the same gateway. MGCP and SRST have had the capability to be
configured on the same gateway since Cisco IOS Release 12.2(11)T.

1-17
IPv6 Support for Unified SRST SIP IP Phones
To make outbound calls while in SRST mode on your MGCP gateway, two fallback commands must be
configured on the MGCP gateway. These two commands allow SRST to assume control over the voice
port and over call processing on the MGCP gateway. With Cisco IOS earlier than 12.3(14)T, the two
commands are the ccm-manager fallback-mgcp and call application alternate commands. With
Cisco IOS releases after 12.3(14)T, the ccm-manager fallback-mgcp and service commands must be
configured. A complete configuration for these commands is shown in the section the “Enabling Cisco
Unified SRST on an MGCP Gateway” section on page 116.
Note The commands listed above are ineffective unless both commands are configured. For instance, your
configuration will not work if you only configure the ccm-manager fallback-mgcp command.
For more information on the fallback methods for MGCP gateways, see the Configuring MGCP Gateway
Support for Cisco Unified Communications Manager document or the MGCP Gateway Fallback
Transition to Default H.323 Session Application document.

Internet Protocol version 6 (IPv6) is the latest version of the Internet Protocol (IP). IPv6 uses packets to
exchange data, voice, and video traffic over digital networks. Also, IPv6 increases the number of
network address bits from 32 bits in IPv4 to 128 bits. From Unified SRST Release 12.0 onwards, Unified
SRST supports IPv6 protocols for SIP IP phones.
IPv6 support in Unified SRST allows the network to behave transparently in a dual-stack (IPv4 and IPv6)
environment and provides additional IP address space to SIP IP phones that are connected to the
network. If you do not have a dual-stack configuration, configure the CLI command call service stop
under voice service voip configuration mode before changing to dual-stack mode. For an example of
switching to dual-stack mode, see Examples for Configuring IPv6 Pools for SIP IP Phones, page 24.
The Cisco IP Phone 7800 Series and 8800 Series are supported on IPv6 for Unified SRST.
For more information on configuring SIP IP phones for IPv6 source address, see Configure IPv6 Pools
for SIP IP Phones, page 19.
For an example of configuring IPv6 Support on Unified SRST, see Examples for Configuring IPv6 Pools
For more details about IPv6 deployment, see IPv6 Deployment Guide for Cisco Collaboration Systems
Release 12.0.
Feature Support for IPv6 in Unified SRST SIP IP Phones

The following basic features are supported for a IPv6 WAN down scenario:
• Basic SIP Line (IPv4 or IPv6) to SIP Line calls (IPv4 or IPv6) when Unified SRST is in dual-stack
no anat mode.
The following supplementary services are supported as part of IPv6 in Unified SRST IP Phones:
• Hold/Resume
• Call Forward
• Call Transfer
• Three-way Conference (with BIB conferencing only)

1-18
• Line to T1/E1 Trunk and Trunk to Line with Supplementary Service Features
• Fax to and from PSTN (IPv4 ATA to ISDN T1/E1) for both T.38 Fax Relay and Fax Passthrough
Restrictions
The following are the known restrictions for IPv6 support on Unified SRST:
• SIP Trunks are not supported on Unified SRST for IPv6 deployment. PSTN calls are supported only
through T1/E1 trunks.
• SCCP IP Phones are not supported in a deployment of IPv6 for Unified SRST.
• SIP Phones can be either in IPv4 only or IPv6 only mode (no anat).
• Trancoding and Transrating are not supported.
• H.323 trunks are not supported.
• Secure SIP lines or trunks are not supported.
• IPv6 on Unified SRST is not supported on the Cisco IOS platform. The support is restricted to Cisco
IOS XE platform with Cisco IOS Release 16.6.1 or later versions.
Configure IPv6 Pools for SIP IP Phones

Before You Begin
• Unified SRST 12.0 or a later version.
• IPv6 option only appears if protocol mode is dual-stack configured under sip-ua configuration mode
or IPv6.
• Cisco Unified SRST License must be configured for the gateway to function as a Unified SRST
gateway to support IPv6 functionality. For more information on licenses, see Cisco Unified SRST
Licenses, page 13.
• Cisco Unified Communications Manager (Unified Communications Manager) is provisioned with
the IPv6 address of Unified SRST. For information on configuration of Unified SRST on Unified
Communications Manager, see the section Survivable Remote Site Telephony Configuration in Cisco
Unified Communications Manager Administration Guide.
SUMMARY STEPS
5. enable
7. ipv6 unicast-routing
8. voice service voip
9. sip
10. no anat
11. call service stop
12. exit
13. exit

1-19
14. sip-ua
15. protocol mode {ipv4 | ipv6 | dual-stack [preference {ipv4 | ipv6}]}
16. exit
17. voice service {voip}
18. sip
19. no call service stop
20. exit
21. voice register global
22. default mode
23. max-dn max-directory-numbers
24. max-pool max-voice-register-pools
25. exit
27. id { network address mask mask | ip address mask mask | mac address }
28. end

1-20
DETAILED STEPS

Example:
Router> enable
Example:
Router #configure terminal
Step 3 ipv6 unicast-routing Enables the forwarding of IPv6 unicast datagrams.
Example:
Router(config)# ipv6 unicast-routing
Step 4 voice service voip Enters voice-service configuration mode to specify a voice
encapsulation type.
• voip—Specifies Voice over IP (VoIP) parameters.
Example:
Router (config)# voice service voip
Step 5 sip Enters SIP configuration mode.
Example:
Router(config-voi-serv)# sip
Step 6 no anat Disables Alternative Network Address Types (ANAT) on a

SIP trunk.
Example:
Router(config-serv-sip)# no anat
Step 7 call service stop Shuts down SIP call service.
Example:
Router(config-serv-sip)# call service stop
Step 8 exit Exits SIP configuration mode.
Example:
Router(config-serv-sip)# exit

1-21

Step 9 exit Exits voice service voip configuration mode.
Example:
Router(config-voi-sip)# exit
Step 10 sip-ua Enters SIP user-agent configuration mode.
Example:
Router(config)# sip-ua
Step 11 protocol mode {ipv4 | ipv6 | dual-stack Allows phones to interact with phones on IPv6 voice
[preference {ipv4 | ipv6}]} gateways. You can configure phones for IPv4 addresses,
IPv6 address es, or for a dual-stack mode.
• ipv4—Allows you to set the protocol mode as an IPv4
Example: address.
Router(config-sip-ua)# protocol mode dual-stack
preference ipv6 • ipv6—Allows you to set the protocol mode as an IPv6
address.
• dual-stack—Allows you to set the protocol mode for
both IPv4 and IPv6 addresses.
• preference—Allows you to choose a preferred IP
address family if protocol mode is dual-stack.
Example:
Router(config-sip-ua)# exit
Step 13 voice service {voip} Enters voice-service configuration mode to specify a voice
encapsulation type.
Example:
Example:

1-22

Step 15 no call service stop Activates SIP call service.
Example:
Example:
Step 17 voice register global Enters voice register global configuration mode to set
parameters for all supported SIP phones in Cisco Unified
CME.
Example:
Router(config)# voice register global
Step 18 default mode Enables mode for provisioning SIP phones in Unified
SRST. The default mode is Unified SRST itself.
Example:
Router(config-register-global)# default mode
Step 19 max-dn max-directory-numbers Limits number of directory numbers to be supported by this
router.
• Maximum number is platform and version-specific.
Example: Type ? for value.
Router(config-register-global)# max-dn 50
Step 20 max-pool max-voice-register-pools Sets maximum number of SIP phones to be supported by the
Unified SRST router.
Example:
Router(config-register-global)# max-pool 40
Step 21 exit Exits voice register global configuration mode.
Example:
Router(config-register-global)# exit
Step 22 voice register pool pool-tag Enters voice register pool configuration mode to set
phone-specific parameters for a SIP phone.
Example:
Router(config)# voice register pool 1

1-23

Step 23 id { network address mask mask | ip address Explicitly identifies a locally available individual SIP
mask mask | mac address } phone to support a degree of authentication.
Example:
Router(config-register-pool)# id network
2001:420:54FF:13::901:0/117
10.64.88.0 mask 255.255.255.0
Example:
Router(config)# end
Examples for Configuring IPv6 Pools for SIP IP Phones

The following example provides configuration of IPv6 pools for SIP IP Phones:
ipv6 unicast-routing
voice service voip
sip
no anat
call service stop
exit
exit
sip-ua
protocol mode dual-stack
exit
voice service voip
sip
no call service stop
exit
voice register global
default mode
max-dn 50
max-pool 40
exit
voice register pool 1
id network 2001:420:54FF:13::901:0/117
end

1-24
Support for Cisco Unified IP Phones and Platforms
The following example provides interface configuration for IPv6 supported on Unified SRST:
configure terminal
interface GigabitEthernet0/0/1
ip address 10.64.86.229 255.255.255.0
negotiation auto
ipv6 address 2001:420:54FF:13::312:82/119
ipv6 enable
The following example provides IP route configuration for IPv6 supported on Unified SRST:
ipv6 route 2001:420:54FF:13::312:0/119 2001:420:54FF:13::312:1

ipv6 route 2001:420:54FF:13::901:0/119 2001:420:54FF:13::312:1
The following example displays output when SIP call service is shut down with the call service stop
CLI command:
Router# show sip service

SIP service is shut
under 'voice service voip', 'sip' submode
The following example displays output when SIP call service is active with the no call service stop
CLI command:
Router# show sip-ua service

SIP Service is up

The following sections provide information about Cisco Feature Navigator and the histories of
Cisco Unified IP Phone, platform, and Cisco Unified CM support from Cisco SRST Version 1.0 to the
present version of Cisco Unified SRST.
• Finding Cisco IOS Software Releases That Support Cisco Unified SRST, page 25
• Cisco Unified IP Phone Support, page 26
• Platform and Memory Support, page 26
• Cisco Unified Communications Manager Compatibility, page 27
• Signal Support, page 27
• Language Support, page 27
• Switch Support, page 27
Finding Cisco IOS Software Releases That Support Cisco Unified SRST
Note With Cisco IOS Release 12.4(15)T, the number of SIP phones supported on each platform is now
equivalent to the number of SCCP phones supported. For example, 3845 now supports 720 phones
regardless of whether these are SIP or SCCP.

1-25
To access Cisco Feature Navigator, go to http://www.cisco.com/go/cfn. An account on Cisco.com is not

required.
See Cisco Unified CME and Cisco IOS Software Version Compatibility Matrix for related compatibility
information.
Cisco Unified IP Phone Support

For the most up-to-date information about Cisco Unified IP Phone support, see Compatibility
Information.
For ATAs that are registered to a Cisco Unified SRST system to participate in FAX calls, they must have
their ConnectMode parameter set to use the "standard payload type 0/8" as the RTP payload type in FAX
passthrough mode. For ATAs used with Cisco Unified SRST 4.0 and higher versions, this is done by
setting bit 2 of the ConnectMode parameter to 1 on the ATA. For more information, see the Parameters
and Defaults chapter in Cisco ATA 186 and Cisco ATA 188 Analog Telephone Adaptor Administrator's
Guide for SCCP.
During Cisco Unified CM fallback, Cisco Unified SRST considers the Cisco VG248 to be a group of
Cisco Unified IP phones. Cisco Unified SRST counts each of the 48 ports on the Cisco VG248 as a
separate Cisco Unified IP phone. Support for Cisco VG248 Version 1.2(1) and higher versions is
available as of Cisco SRST Version 2.1. For more information, see Cisco VG248 Analog Phone Gateway
Data Sheet and Cisco VG248 Analog Phone Gateway Version 1.2(1) Release Notes.
For IPv6 Support on Unified SRST, all the legacy IP Phones and Voice Gateways must be converted or
reconfigured to IPv4-Only SIP signaling from SCCP signaling, if applicable.
Platform and Memory Support

For the most up-to-date information about Platform and Memory Support, see Compatibility
Information.
Determining Platform Support Through Cisco Feature Navigator

Cisco IOS software is packaged in feature sets that are supported on specific platforms. To get updated
information regarding platform support for this feature, access Cisco Feature Navigator. Cisco Feature
Navigator dynamically updates the list of supported platforms as new platform support is added for the
feature.
Availability of Cisco IOS Software Images

Platform support for particular Cisco IOS software releases is dependent on the availability of the
software images for those platforms. Software images for some platforms may be deferred, delayed, or
changed without prior notice. For updated information about platform support and availability of
software images for each Cisco IOS software release, see the online release notes or, if supported,
Cisco Feature Navigator.
For the most up-to-date information about Cisco IOS software images, see Compatibility Information.

1-26
Multicast Music On Hold
Cisco Unified Communications Manager Compatibility

See Cisco Unified Communications Manager Compatibility Matrix.
Signal Support
Cisco Unified SRST supports FXS, FXO, T1, E1, and E1 R2 signals.
Language Support
See Cisco Unified Communications Manager Express Cisco Unified CME Localization Matrix.
Switch Support
Cisco SRST 3.2 and later versions support all PRI and BRI switches including the following:
• basic-1tr6
• basic-5ess
• basic-dms100
• basic-net3
• basic-ni
• basic-ntt NTT switch type for Japan
• basic-ts013
• primary-4ess Lucent 4ESS switch type for the United States
• primary-dms100 Northern Telecom DMS-100 switch type for the United States
• primary-net5 NET5 switch type for the United Kingdom, Europe, Asia, and Australia
• primary-ni National ISDN switch type for the United States
• primary-ntt NTT switch type for Japan
• primary-qsig QSIG switch type
• primary-ts014 TS014 switch type for Australia (obsolete)

For Unified SRST 3.0 and later versions, you can configure the MOH audio stream as a multicast source.
A Unified SRST router that is configured for multicast MOH also transmits the audio stream on the
physical IP interfaces of the specified router to permit access to the stream by external devices. Certain
IP phones do not support multicast MOH because they do not support IP multicast. You can disable
multicast MOH to individual phones that do not support multicast. Callers hear a repeating tone when
they are placed on hold.

1-27
Multicast MOH on Unified SRST is supported for both SIP and SCCP phones. Support is offered for
G.711 and G.729 codecs with multicast MOH on Unified SRST. Multicast MOH is supported on Cisco
Integrated Services Router Generation 2 (ISR G2) and the Cisco 4000 Series Integrated Services
Routers.
For SIP phones to play the Multicast MOH, you need to configure the CLI command moh enable-g711
filename (for example, moh enable-g711 "flash:en_bacd_music_on_hold.au" or moh g729
flash:SampleAudioSource.g729.wav). For SCCP phones to play Multicast MOH, you need to configure
the CLI command multicast moh ip-address port port-number [route ip-address-list] (for example,
multicast moh 239.1.1.1 port 2000), apart from the CLI command moh filename. If both the CLI
commands are not configured, SCCP phones will only play tone on hold.
For more information on supporting Multicast MOH with Unified SRST for a scenario where WAN is
available, see Information About Using Cisco Unified SRST Gateways as a Multicast MOH Resource,
page 13.
Configure Multicast Music On Hold for Unified SRST

To configure multicast MOH for Unified SRST, perform the following steps.
Prerequisites
• Unified SRST 3.0 or later versions.
• IP phones do not support multicast at 224.x.x.x addresses.
SUMMARY STEPS
1. enable
3. call-manager-fallback
4. moh filename
5. multicast moh ip-address port port-number [route ip-address-list]
6. exit
DETAILED STEPS

Example:
Router> enable
Example:
Step 3 call-manager-fallback Enters call-manager-fallback configuration mode.
Example:
Router(config)# call-manager-fallback

1-28

Step 4 moh filename Enables music on hold using the specified file.
• If you specify a file with this command and later
Example: want to use a different file, you must disable use
Router(config-cm-fallback)# moh enable-g711 of the first file with the no moh command before
"flash:en_bacd_music_on_hold.au" configuring the second file.
OR
Router(config-cm-fallback)# moh g729

flash:SampleAudioSource.g729.wav
Step 5 multicast moh ip-address port port-number [route Specifies that this audio stream is to be used for
ip-address-list] multicast and also for MOH.
Note This command is required to use MOH for
Example: internal calls and it must be configured after
Router(config-cm-fallback)# multicast moh 239.1.1.1 MOH is enabled with the moh command.
port 2000
• ip-address—Destination IP address for
multicast.
• port port-number—Media port for multicast.
Range is 2000 to 65535. We recommend
port 2000 because it is already used for normal
RTP media transmissions between IP phones and
the router.
Note Valid port numbers for multicast include even
numbers that range from 16384 to 32767.
(The system reserves odd values.)
• route—(Optional) List of explicit router

interfaces for the IP multicast packets.
• ip-address-list—(Optional) List of up to four
explicit routes for multicast MOH. The default is
that the MOH multicast stream is automatically
output on the interfaces that correspond to the
address that was configured with the ip
source-address command.
Note For MOH on internal calls, packet flow must
be enabled to the subnet on which the phones
are located.
Step 6 exit Exits call-manager-fallback configuration mode.
Example:
Router(config-cm-fallback)# exit

1-29
Where to Go Next
Where to Go Next
The next chapters of this book describe how to configure Cisco Unified SIP SRST. As shown in
Table 1-3, each chapter takes you through tasks in the order in which they need to be performed. The
first task for configuring Cisco Unified SRST is to ensure that the basic software and hardware in your
system are configured correctly for Cisco Unified SRST.
Table 1-3 Cisco Unified SRST Configuration Sequence
Task Where Task Is Described

7. Setting up a Cisco Unified SRST system Setting Up the Network, page 115
to communicate with your network
8. Configuring Version 4.1 features Cisco Unified SIP SRST 4.1, page 127
9. Setting up the basic Cisco Unified SRST Setting Up Cisco Unified IP Phones using SCCP,
phone configuration using SCCP page 137
10. Providing a backup to an external SIP call Setting Up Cisco Unified IP Phones using SIP,
control (IP-PBX) by supplying basic page 157
registrar services
11. Configuring incoming and outgoing calls Configuring Call Handling, page 175
12. Configuring optional security for SRST Configuring Secure SRST for SCCP and SIP, page 231
13. Setting up voicemail Integrating Voicemail with Cisco Unified SRST,
page 321
14. Setting up video parameters Setting Video Parameters, page 345
15. Monitoring and maintaining Cisco Monitoring and Maintaining Cisco Unified SRST,
Unified Survivable Remote page 359
Site Telephony (SRST)
Additional References
The following sections provide additional references related to Cisco Unified SIP SRST:
• Related Documents, page 31
• Standards, page 33
• MIBs, page 33
• RFCs, page 33
• Technical Assistance, page 33

1-30
Related Documents
Related Topic Documents
Cisco IOS voice product configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Voice Command Reference
• Cisco IOS Debug Command Reference
• Cisco IOS Tcl IVR and VoiceXML Application Guide
• Cisco IOS Survivable Remote Site Telephony Version 3.2 System
Administrator Guide
Configuring SRST and MGCP Fallback • Configuring MGCP Gateway Support for
Cisco Unified Communications Manager
• MGCP Gateway Fallback Transition to Default H.323 Session
Application
• Configuring SRS Telephony and MGCP Fallback
Cisco Unified Communications Manager user • Cisco Unified Communications Manager
documentation
• Cisco Unified Communications Manager Security Guide
• Cisco Unified Communications Operating System
Administration Guide
Cisco Unified IP Phones • Cisco 7900 Series Unified IP Phones End-User Guides
• Cisco IP Phone Authentication and Encryption for
Cisco Communications Manager
• Cisco Unified IP Phone 7970 Series Administration Guide for
Cisco Unified CallManager, Release 5.0 (for models 7970G and
7971G-GE) (SCCP), “Understanding Security Features for
Cisco IP Phones” section.
Cisco Unified SRST commands and specifications • Cisco Unified SRST and Cisco Unified SIP SRST Command
Reference (All Versions)
• Cisco Unified SRST 8.0 Supported Firmware, Platforms,
Memory, and Voice Products
• Cisco Unified SRST 4.3 Supported Firmware, Platforms,
Cisco Security Documentation • Media and Signaling Authentication and Encryption Feature for
Cisco IOS MGCP Gateways
• Cisco IOS Certificate Server
• Manual Certificate Enrollment (TFTP and Cut-and-Paste)
• Certification Authority Interoperability Commands
• Certificate Enrollment Enhancements
Cisco SIP SRST V3.4: Cisco IOS SIP Survivable • Cisco IOS SIP SRST Feature Roadmap
Remote Site Telephony Feature Roadmap
Cisco SIP functionality • Cisco IOS SIP Configuration Guide

1-31
Related Topic Documents

Cisco SRST command reference • Cisco IOS Survivable Remote Site Telephony Version 3.2
Command Reference
Command reference information for voice and • Cisco IOS Voice Command Reference
telephony commands
• Cisco IOS Debug Command Reference
DHCP • Cisco IOS DHCP Server
Media Inactive Call Detection • Media Inactive Call Detection
Phone documentation for Cisco Unified SRST • Cisco Unified IP Phones 7900 Series
• Survivable Remote Site Telephony
Standard Glossary • Cisco IOS Voice Configuration Library Glossary
Standard Preface • Cisco IOS Voice Configuration Library Preface

1-32
Obtaining Documentation, Obtaining Support, and Security Guidelines
Standards
Standard Title
ITU X. 509 Version 3 Public-Key and Attribute Certificate Frameworks
MIBs
MIB MIBs Link
No new or modified MIBs are supported by this To locate and download MIBs for selected platforms, Cisco IOS
feature, and support for existing MIBs has not been releases, and feature sets, use Cisco MIB Locator found at the
modified by this feature. following URL:
http://www.cisco.com/go/mibs
RFCs
RFC Title
RFC 2246 The Transport Layer Security (TLS) Protocol Version 1.0
RFC 2543 SIP: Session Initiation Protocol
RFC 3261 SIP: Session Initiation Protocol
RFC 3711 The Secure Real-Time Transport Protocol (SRTP)
Technical Assistance
Description Link
The Cisco Technical Support & Documentation http://www.cisco.com/techsupport
website contains thousands of pages of searchable
technical content, including links to products,
technologies, solutions, technical tips, and tools.
Registered Cisco.com users can log in from this page to
access even more content.
Obtaining Documentation, Obtaining Support, and Security

Guidelines
For information on obtaining documentation, obtaining support, providing documentation feedback,
security guidelines, and also recommended aliases and general Cisco documents, see the monthly
What’s New in Cisco Product Documentation, which also lists all new and revised Cisco technical
documentation, at
http://www.cisco.com/en/US/docs/general/whatsnew/whatsnew.html.

1-33
Obtaining Documentation, Obtaining Support, and Security Guidelines

1-34
CHAPTER 2
Cisco Unified SIP SRST on Cisco 4000 Series
Integrated Services Router
This chapter describes the support for Unified SIP SRST on the Cisco 4000 Series Integrated Services
platform.
Contents
• Overview, page 36
• Platform and Memory Support, page 36
• Cisco IOS Software Releases that Support Unified SRST, page 36
• Feature Support, page 38
• Unified IP Phone Support, page 39
• Cisco Unified Communications Manager Compatibility, page 39
• Supported PSTN Trunk Connectivity, page 40
• Language Support, page 40
• Switch Support, page 40
• Interface Support for Unified SRST, page 41
• Licensing, page 41
• Configure SIP Registrar Functionality for SIP Phones on Unified SRST, page 46
• IPv6 Support for Unified SRST SIP IP Phones, page 59
• Configure Unified SRST on Cisco 4000 Series Integrated Services Platform, page 65
• Configure Voice Hunt Groups on Unified SRST, page 69
• Configure Feature Support on Unified SIP SRST, page 72
• Enabling KPML for SIP Phones, page 79
• Disabling SIP Supplementary Services for Call Forward and Call Transfer, page 80
• Examples, page 83

2-35
Chapter 2 Cisco Unified SIP SRST on Cisco 4000 Series Integrated Services Router
Overview
Overview
This chapter describes Unified SRST functionality on Cisco 4000 Series Integrated Services Routers for
SIP phones. Unified SIP SRST provides backup to Unified Communications Manager when the IP
connectivity to Unified Communications Manager is down.
Cisco Unified SIP SRST supports the following during a WAN outage:
• Basic Registration of SIP phones.
• Basic call support on SIP phones.
• Basic supplementary services such as Call Transfer, MOH, and Conference
Platform and Memory Support

From Unified SRST Release 10.0 (Cisco IOS Release 15.3(3)M), Unified SIP SRST is supported on the
Cisco 4000 Series Integrated Services platform. As part of the Cisco IOS Release 15.3(3)M Release,
support was introduced on the Cisco 4451-X Integrated Services Router. From Unified SRST Release
10.5 (Cisco IOS Release 15.4(3)M, 15.4(3)S), SIP SRST is supported on all Cisco 4000 Series Integrated
Services Routers.
The following Cisco 4000 Series Integrated Services Router platforms are supported:
• Cisco ISR 4321 Integrated Services Routers
For more information on Platform and Memory Support, see Compatibility Information.
Cisco IOS Software Releases that Support Unified SRST

For information on the Unified SRST Release and the corresponding IOS Software, see Unified CME,
Unified SRST, and Cisco IOS Software Version Compatibility Matrix for related compatibility
information.
To access Cisco Feature Navigator, go to http://www.cisco.com/go/cfn. An account on Cisco.com is not
required.
Install Cisco IOS Software

To verify that the recommended software is installed on the Cisco router and if necessary, download and
install a Cisco IOS Voice or higher image, perform the following steps.

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Cisco IOS Software Releases that Support Unified SRST
Before You Begin

• The Cisco router is installed including sufficient memory, all Cisco voice services hardware, and
other optional hardware.
SUMMARY STEPS
1. Identify which Cisco IOS software release is installed on router.

2. Determine whether the Cisco IOS release supports the recommended Unified SRST.
3. Download and extract the recommended Cisco IOS IP Voice or higher image to flash memory
4. Use the reload command to reload the Unified SRST router with the new software.
DETAILED STEPS
Step 1 Identify which Cisco IOS software release is installed on router. Log in to the router and use the show
version EXEC command.
Example:
Router> show version
Cisco Internetwork Operating System Software
IOS (tm) 12.3 T Software (C2600-I-MZ), Version 12.3(11)T, RELEASE SOFTWARE
Step 2 Compare the Cisco IOS release installed on the Cisco router to the information in the
Cisco Unified CME, Unified SRST, and Cisco IOS Software Version Compatibility Matrix to determine
whether the Cisco IOS release supports the recommended Unified SRST.
Step 3 If necessary, download and extract the recommended Cisco IOS IP Voice or higher image to flash
memory in the router.
To find software installation information, access information located at www.cisco.com>Technical
Support & Documentation>Product Support> Cisco IOS Software>Cisco IOS Software Mainline release
you are using> Configuration Guides> Cisco IOS Configuration Fundamentals and Network
Management Configuration Guide>Part 2: File Management>Locating and Maintaining System Images.
Step 4 To reload the Unified SRST router with the new software after replacing or upgrading the Cisco IOS
release, use the reload privileged EXEC command.
Example:
Router# reload
System configuration has been modified. Save [yes/no]:

Y
Building configuration...
OK
Proceed with reload? Confirm.
11w2d: %Sys-5-RELOAD: Reload requested by console. Reload reason: reload command
.
System bootstrap, System Version 12.2(8r)T, RELEASE SOFTWARE (fc1)
...
Press RETURN to get started.
...
Router>

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Feature Support
Feature Support
The following features are known to be supported for Unified SIP SRST on Cisco 4000 Series Integrated
Services Platform:
• Auto-answer (If enabled on Unified Communications Manager)
• Alert/Semi-Consult/Attended/Consult Transfer
• Ad-hoc Software Conference
• Hold or Resume
• Headset Answer
• Caller ID Display
• Call Forward to Voice Hunt Group
• Call Transfer to a Voice Hunt Group
• Voicemail
• Message Waiting Indicator (MWI)
• Do Not Disturb (DND)
• DTMF
• Feature Button or Programmable Line Key (PLK) - If enabled on Unified Communications Manager
• Key Expansion Module (KEM - Supported only on the 8851/8851NR/8861 phones)
• Bulk Registration Support
• Enabling or Disabling KPML
• Alias Feature
• Call Forward (All, Busy, No Answer, Mailbox)
• Call Forward All Softkey on Phone
• Unicast MOH
• Audio codecs (G.722, G.711, G.729, iLBC)
• Translation Profile
• Conference Blocking
• Transfer Blocking
• COR
• Voice Class Codec
• SNMP/MIB (Supported only to get mode and number of registered phones)
• Speed Dial (If enabled on Unified Communications Manager)
• Call Waiting (If enabled on Unified Communications Manager)
• Forced Authorization Code
• Redial
• Speakerphone (Dialing, Answering)
• System Message
• After Hours

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Unified IP Phone Support
• SSH to Phone
• Span to PC (except Cisco IP Phone 8831)
• Web Access to Phone
• Voice Hunt Group (Support for Parallel, Sequential, Peer, and Longest-Idle hunt groups). Basic
features such as Call, Hold or Resume are only supported.)
Restrictions of Unified SRST on Cisco 4000 Series Integrated Services

Routers
• Multicast MOH for SIP is not supported on the Cisco 4000 Series Integrated Services Routers.
• Transcoding is not supported on the Unified SRST.
• Multi-VRF is supported only on the Cisco Integrated Services Routers Generation 2 (ISR G2).
• Shared lines and Mixed shared lines are not supported on the Unified SRST (supported on the
Unified E-SRST).
• Privacy (on hold) is not supported on the Unified SRST (supported on the Unified E-SRST).
• SNMP/MIB support is restricted to fetching information on mode and number of registered phones.
• Unified SRST supports only the basic voice hunt group features. To configure advanced voice hunt
group features, you must deploy the Cisco Unified Enhanced Survivable Remote Site Telephony.
• Video Calling is not supported on Unified SIP SRST.
Unified IP Phone Support

Unified SIP SRST on Cisco 4000 Series Integrated Services Platform is supported on the following
phone series:
• Cisco IP Phone 7800 Series (except Cisco IP Conference Phone 7832)
• Cisco IP Phone 8800 Series (except Cisco IP Conference Phone 8832)
For IPv6 Support on Unified SRST, all the legacy IP Phones and Voice Gateways must be converted or
reconfigured to IPv4-Only SIP signaling from SCCP signaling, if applicable.

For more information on Unified Communications Manager compatibility, see Cisco Unified
Communications Manager Compatibility Matrix.
Installing Cisco Unified Communications Manager

When installing Cisco Unified Communications Manager, consider the following:
• See the installation instructions for your version in the Cisco Unified Communications Manager
Install and Upgrade Guides.

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• Integrate Cisco Unified SRST with Cisco Unified Communications Manager. Integration is
performed from Cisco Unified Communications Manager. See the
Integrating Cisco Unified SIP SRST with Cisco Unified Communications

Manager
The procedure for integrating Unified SRST with Cisco Unified Communications Manager is as follows:
For Cisco Communications Manager integration with Unified SIP SRST, you must create an SRST
reference and apply it to a device pool. An SRST reference is the IP address of the Cisco Unified SRST
Router.
Step 1 Create an SRST reference.

a. From any page in Cisco Unified Communications Manager, click System and SRST.
b. On the Find and List SRST References page, click Add a New SRST Reference.
c. On the SRST Reference Configuration page, enter a name in the SRST Reference Name field and
the IP address of the Cisco SRST router in the IP Address field.
d. Click Insert.
Step 2 Apply the SRST reference or the default gateway to one or more device pools.
a. From any page in Cisco Unified Communications Manager, click System and Device Pool.
b. On the Device Pool Configuration page, click the required device pool icon.
c. On the Device Pool Configuration page, choose an SRST reference or “Use Default Gateway” from
the SRST Reference field’s menu.
Supported PSTN Trunk Connectivity

Unified SRST is supported on SIP trunks. Also, Unified SIP SRST supports the following trunk types:
• FXO/FXS
• Basic Rate ISDN
• Primary Rate ISDN (T1 or E1)
Language Support
For information on language support, see Localization Matrix.
Switch Support
Unified SRST supports all PRI and BRI switches including the following:
• basic-1tr6
• basic-5ess

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Interface Support for Unified SRST
• basic-dms100
• basic-net3
• basic-ni
• basic-ntt NTT switch type for Japan
• basic-ts013
• primary-dms100 Northern Telecom DMS-100 switch type for the United States
• primary-net5 NET5 switch type for the United Kingdom, Europe, Asia, and Australia
• primary-ni National ISDN switch type for the United States
• primary-ntt NTT switch type for Japan
• primary-qsig QSIG switch type
primary-ts014 TS014 switch type for Australia (obsolete)
Interface Support for Unified SRST

Unified SRST routers have multiple interfaces that are used for signaling and data packet transfers. The
two types of interfaces available on a Cisco router include the physical interface and the virtual interface.
The type of physical interfaces available on a router depends on its interface processors or port adapters.
Virtual interfaces are software-based interfaces that you create in the memory of the networking device
using Cisco IOS commands. To configure a virtual interface for connectivity, you can use the Loopback
Interface for Unified SRST.
The following interfaces are supported on Unified SRST:
• Gigabit Ethernet Interface (IEEE 802.3z) (interface gigabitethernet)
• Loopback Interface (interface loopback)
• Fast Ethernet Interface (interface fastethernet)
Licensing
Cisco Unified SRST Permanent License

When you purchase a Cisco Unified SRST permanent license, the permanent license is installed on the
device when the product is shipped to you. A permanent license never expires and you will gain access
to that particular feature set for the lifetime of the device across all IOS release. If you purchase a
permanent license for Cisco Unified SRST, you do not have to go through the Evaluation Right to Use
and Right To Use (RTU) licensing processes for using the features. If you want to purchase a CME-SRST
license for your existing device, you have to go through the RTU licensing process for using the features.
There is no change in the existing process for purchasing the license.

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Licensing
The Cisco Unified SRST permanent license is available in the form of an XML cme-locked3 file. You
must get the XML file and load it in the flash memory of the device. To install the permanent license
from the command prompt, use the license install flash0:cme-locked3 command. The cme-locked3 is
the XML file of the license.
Collaboration Professional Suite License

Collaboration Professional is a new suite of licenses. The Collaboration Professional suite can be
purchased either as a permanent license or an RTU license.
Collaboration Professional Suite Permanent License—When you purchase the Collaboration
Professional Suite license, by default, the Cisco Unified SRST licenses are delivered as part of the
Collaboration Professional Suite. You do not have to separately install and activate the Cisco Unified
SRST license. The Collaboration Professional Suite permanent license is available in the form of an
XML file. You must get the XML file and load it in the flash memory of the device. To install the
permanent license from the command prompt, use the license install flash:lic_name command.
Collaboration Professional Suite RTU License—When you purchase the Collaboration Professional
Suite RTU license, you do not have to go through the Evaluation Right to Use process. However, you
have to go through the RTU licensing process for using the Cisco Unified SRST features. To install the
Collaboration Professional Suite RTU license from the command prompt, use the license install
flash0:colla_pro command. To activate the license, use the license boot module c2951
technology-package collabProSuitek9 command.
Also, you have to:
• Choose an appropriate Cisco Unified SRST version. Each SRST version supports a specific set of
IP phones, memory requirements, features, and DNs. See the Platform and Memory Support,
page 36 and the Restrictions of Unified SRST on Cisco 4000 Series Integrated Services Routers,
page 39.
• Choose an appropriate phoneload. SRST only supports certain phoneloads that have been tested with
the various Cisco Unified Communications Manager versions. For the most up-to-date phoneloads,
see the
Cisco Unified SRST Supported Firmware, Platforms, Memory, and Voice Products.
If you have Cisco Unified Communications Manager already installed, verify that your version of
Cisco Unified Communications Manager is compatible with your Unified SRST release. See the Cisco
Unified Communications Manager Compatibility, page 39.
Cisco Smart License

From Release 12.1 onwards, Unified SRST supports Smart Licensing, apart from the existing CSL
licensing model. Smart Licensing is supported only on Cisco 4000 Series Integrated Services Router.
Depending on the technology package available on the router, licenses such as UCK9 and Security are
supported using Smart Licensing.
Smart Software Licensing is Cisco's new licensing model that gives you visibility into license ownership
and consumption. Smart licenses is managed by a cloud-based deployment model, namely Cisco Smart
Software Manager (CSSM), or an on-prem software, Smart Software Manager satellite. Unified SRST
supports both CSSM and satellite. Your access to the customer Smart Account residing on CSSM is
authenticated using valid Cisco credentials. With the Smart Licensing support for Unified SRST, your
device can register with CSSM or Cisco Smart Software Manager satellite. You can access your Licenses
at the Cisco Software Central.

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Licensing
Unified SRST must register with CSSM or Cisco Smart Software Manager satellite to report license
consumption. You can register Unified SRST to a Virtual Account within a Smart Account by generating
a token ID from it, and pasting it to the underlying platform, Cisco 4000 Series Integrated Services
Router. Once the token is generated, it can be used to register many other product instances in your
network.
On the Unified SRST router, you need to ensure that the call home feature is not disabled. Also, Smart
Licensing must be enabled at the router using the CLI command license smart enable. Use the no form
of the command to disable Smart Licensing.
For more information on configuring Smart Licensing in your router, see Cisco 4000 Series ISRs
Software Configuration Guide. For more information on configuring Call Home for your devices, see
Configure Call Home, page 44. Once Smart Licensing is enabled, the router enters a 90-day evaluation
period that persists until it registers to CSSM or the Cisco Smart Software Manager satellite.
You can register the router to CSSM or Cisco Smart Software Manager satellite with the token ID. To
register the device (Unified SRST router) with CSSM or Cisco Smart Software Manager satellite, use
the CLI command license smart register idtoken. For information on registering the device with
CSSM, see Device Registration, Software Activation Configuration Guide, Cisco IOS Release 15M&T.
Upon successful registration, Unified SRST is in Registered status. As part of the registration process,
the router sends an authorization request, indicating the number of phone endpoints defined by the
max-pool, for SIP SRST, and max-ephone, for SCCP SRST. Based on the licenses in the Smart
Account, CSSM or Cisco Smart Software Manager satellite responds with one of the defined statuses
such as Authorized (using less than or equal to the number of licenses provisioned in CSSM or Cisco
Smart Software Manager satellite) or Out-of-Compliance (using more than it has licenses for).
The license limit on Unified SRST is restricted by the maximum platform limit defined for the Unified
SRST router (a cumulative sum of phones configured under max-pool and max-ephone). Hence, the
license usage count cannot exceed the platform limit set for the Unified SRST router even when the
cumulative sum of phones configured under max-pool and max-ephone exceeds the defined platform
limit. For more details on the platform limit defined for Unified SRST, see Cisco Unified SRST/E-SRST
12.1 Supported Firmware, Platforms, Memory, and Voice Products.
CSSM or Smart Software Manager satellite reports license consumption submitted by the platform in its
User Interface (UI), and subtracts it from the available licenses in the Virtual Account within the Smart
Account. Unified SRST supports only one license entitlement to validate phones configured on Unified
SRST.
SRST_EP —This license type supports all phones configured on Unified SRST.
Note The SRST_EP license count reflects the total phone count of both the ephones and pools that are
configured in the Unified SRST irrespective of whether the phones are registered or not.
Unified SRST sends an authorization request when a license consumption changes or every 30 days to
let CSSM or Cisco Smart Software Manager satellite know it's still available and communicating. The
ID certificate issued to identify Unified SRST at time of registration is valid for one year, and is
automatically renewed every six months.
Note If the router does not communicate with CSSM or Cisco Smart Software Manager satellite for 90 days,
the license authorization expires. When the license authorization expires, the devices registered on
Unified SRST change status to Out of Compliance.

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Licensing
The license count is evaluated for the number of phones configured across the routers. The CSSM
Licenses page reflects the total license count usage. The total number of licenses available for a type of
license (Quantity), number of licenses currently used (In Use), and the number of unused or over-used
licenses (Surplus or Shortage). If you do not have enough Cisco Smart licenses, you are in
Out-of-Compliance state.
For example, consider a smart account in CSSM with 50 SRST_EP licenses. If you have a registered
Unified SRST with 20 phones configured, the CSSM licenses page reflects Quantity as 50, In Use as 20,
and Surplus as 30. For more information on Smart Software Manager, see Cisco Smart Software
Manager User Guide.
For more information on switching between CSL and Cisco Smart License, see Licensing Modes,
page 45.
The license entitlement for Unified SRST smart license is displayed on the router as follows:
Router# show license summary
Smart Licensing is ENABLED
Registration:
Status: REGISTERED
Smart Account: ABC
Virtual Account: XYZ
Export-Controlled Functionality: Not Allowed
Last Renewal Attempt: None
Next Renewal Attempt: Jun 07 12:08:10 2017 UTC
License Authorization:
Status: AUTHORIZED
Last Communication Attempt: SUCCESS
Next Communication Attempt: Apr 13 07:11:48 2017 UTC
License Usage:
License Entitlement tag Count Status

-----------------------------------------------------------------------------
regid.2014-12.com.ci... (ISR_4351_UnifiedCommun..) 1 AUTHORIZED
regid.2016-10.com.ci... (SRST_EP) 4 AUTHORIZED
Configure Call Home

To configure the call home destination address and proxy server details for the HTTP proxy request,
Prerequisites
• Cisco Smart Software Licensing is enabled.
SUMMARY STEPS
2. call-home destination address http url
3. call-home http-proxy proxy_address port port number
4. end

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Licensing
DETAILED STEPS

Step 1 configure terminal Enters configuration mode.
Example:
Step 2 call-home destination address http url Defines the destination URL to which Call Home
messages, including licensing requests are sent. The
destination URL can be the URL for Transport
Example:
Router(config)# call-home destination address http
Gateway or CSSM satellite.
http://10.22.183.117:8080/ddce/services/DDCEService The URL to the Cisco Smart Licensing production
server is set by default.
Step 3 call-home http-proxy proxy_address port port number Specifies the proxy server for the HTTP request.
Example:
Router(config)# call-home http-proxy 7.7.7.7 port
3218
Example:
Router(config)# end
Licensing Modes
From Unified SRST 12.1 onwards, both CSL and Smart Licensing modes are supported. That is, you can
continue with CSL by not enabling Smart Licensing. Alternatively, they can enable Smart Licensing and
decide later to go back to CSL by disabling Smart Licensing with the no license smart enable command.
When you switch to CSL from the Smart Licensing mode, you must ensure that the End User License
Agreement (EULA) is signed. CSL is not supported unless the EULA is signed. Use the CLI command
license accept end user agreement in global configuration mode to configure EULA.
To verify the status of the license issued to phones registered on Unified SRST, you can use the show
license command.
Router#show license ?
all Show license all information
status Show license status information
summary Show license summary
tech Show license tech support information
udi Show license udi information
usage Show license usage information

2-45
Configure SIP Registrar Functionality for SIP Phones on Unified SRST
Restrictions
• For the Unified SRST license, the UCK9 technology package must be available if the Collaboration
Professional Suite package is not installed.
To purchase a license, go to http://www.cisco.com/cgi-bin/tablebuild.pl/ip-key. To activate cme-srst
feature license, see the Activating CME-SRST Feature License document.
Configure SIP Registrar Functionality for SIP Phones on

Unified SRST
Session Initiation Protocol (SIP) registrar functionality in Cisco IOS software is an essential part of
Cisco Unified SIP Survivable Remote Site Telephony (SRST). According to RFC 3261, a SIP registrar
is a server that accepts Register requests.
Unified SIP SRST provides backup to Cisco Unified Communications Manager. The registrar
functionality is configured on the Unified SRST gateway so as to assist fallback of endpoints to Unified
SRST from Unified Communications Manager.
These services are used by a SIP IP phone if there is a WAN connection outage, and the SIP phone is
unable to communicate with its primary SIP call control (IP-PBX). The Unified SIP SRST device also
provides PSTN gateway access for placing and receiving PSTN calls.
SUMMARY STEPS
1. enable
4. allow-connections sip to sip
5. sip
6. registrar server [expires [max sec] [min sec]]
7. end
DETAILED STEPS

Example:
Router> enable
Example:

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Step 3 voice service voip Enters voice service configuration mode.
Example:
Router(config)# voice service voip
Step 4 allow-connections sip to sip Allows connections from SIP to SIP endpoints.
Example:
Router(config-voi-srv)# allow-connections sip
to sip
Example:
Router(config-voi-srv)# sip
Step 6 registrar server [expires [max sec] [min sec]] Enables SIP registrar functionality. The keywords and
arguments are defined as follows:
Example: • expires: (Optional) Sets the active time for an incoming
Router(conf-serv-sip)# registrar server expires registration.
max 600 min 60
• max sec: (Optional) Maximum expiration time for a
registration, in seconds. The range is from 600 to
86400. The default is 3600.
Note
Ensure that the registration expiration timeout is set
to a value smaller than the TCP connection aging
timeout to avoid disconnection from the TCP.
• min sec: (Optional) Minimum expiration time for a

registration, in seconds. The range is from 60 to 3600.
The default is 60.
Step 7 end Returns to privileged EXEC mode.
Example:
Router(conf-serv-sip)# end
Configure Back up Registrar Service to SIP Phones

Backup registrar service to SIP IP phones can be provided by configuring a voice register pool on SIP
gateways. The voice register pool configuration provides registration permission control and can be used
to configure some dial-peer attributes that are applied to the dynamically created VoIP dial peers when
SIP phone registrations match the pool. The following call types are supported:
• SIP IP phone to or from:
– Local PSTN
– Local analog FXS phones

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– Local SIP IP phone

The commands in the configuration provide registration permission control and set up a basic voice
register pool. The pool gives users control over which registrations are accepted by a Cisco Unified SIP
SRST device and which can be rejected. Registrations that match this pool create VoIP SIP dial peers
with the dial-peer attributes set to these configurations. Although only the id command is mandatory,
this configuration example shows basic functionality.
Prerequisites
• The SIP registrar must be configured before a voice register pool is set up.
Restrictions
• The id command identifies the individual SIP IP phone or sets of SIP IP phones that are to be
configured. Thus, the id command configured in Step 5 is required and must be configured before
any other voice register pool commands. For Unified SRST, It is recommended to configure id
ip/nework/device-id-name and avoid using id mac.
Note It is recommended that id mac command is not configured for Unified SRST, as the phones falling back
from Unified Communications Manager to Unified SRST do not mostly fall back on the same network.
Note The command proxy described in Step 7 is an optional configuration.
Note To monitor SIP proxies, the call fallback active command must be configured, as described in Step 3.
SUMMARY STEPS
1. enable
3. call fallback active
4. voice register pool tag
5. id [{network address mask mask | ip address mask mask | mac address}][device-id-name
devicename]
6. preference preference-order
7. proxy ip-address [preference value] [monitor probe {icmp-ping | rtr} [alternate-ip-address]]
(Optional)
8. voice-class codec tag
9. end

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DETAILED STEPS

Example:
Router> enable
Example:
Step 3 call fallback active (Optional) Enables a call request to fall back to alternate
dial peers if there is network congestion.
Example: • This command is used if you want to monitor the proxy
Router(config)# call fallback active dial peer and fallback to the next preferred dial peer.
For full information on the call fallback active
command, see PSTN Fallback Feature.
Step 4 voice register pool tag Enters voice register pool configuration mode for SIP
phones.
Example: • Use this command to control which registrations are
Router(config)# voice register pool 12 accepted or rejected by a Cisco Unified SIP SRST
device.
Step 5 id [{network address mask mask | ip address Explicitly identifies a locally available individual or set of
mask mask | mac address}] [device-id-name SIP IP phones. The keywords and arguments are defined as
devicename]
follows:
• network address mask mask: The network address
Example: mask mask keyword/argument combination is used to
accept SIP Register messages for the indicated phone
172.16.0.0 mask 255.255.0.0
numbers from any IP phone within the indicated IP
subnet.
• ip address mask mask: The ip address mask mask
keyword/argument combination is used to identify an
individual phone.
• mac address: MAC address of a particular
Cisco Unified IP Phone.
• device-id-name devicename: Defines the device name
to be used to download the phone’s configuration file.
Step 6 preference preference-order Sets the preference order for the VoIP dial peers to be
created. Range is from 0 to 10. Default is 0, which is the
highest preference.
Example:
Router(config-register-pool)# preference 2 • The preference must be greater (lower priority) than the
preference configured with the preference keyword in
the proxy command.

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Step 7 proxy ip-address [preference value] [monitor (Optional) Autogenerates additional VoIP dial peers to
probe {icmp-ping | rtr} [alternate-ip-address]] reach the main SIP proxy whenever a Cisco Unified SIP IP
Phone registers with a Cisco Unified SIP SRST gateway.
Example: The keywords and arguments are defined as follows:
Router(config-register-pool)# proxy • ip-address: IP address of the SIP proxy.
10.2.161.187 preference 1
• preference value: (Optional) Defines the preference of
the proxy dial peers that are created. The preference
must be less (higher priority) than the preference
configured with the preference command.
Range is from 0 to 10. The highest preference is 0.
There is no default.
• monitor probe: (Optional) Enables monitoring of
proxy dial peers.
• icmp-ping: Enables monitoring of proxy dial peers
using ICMP ping.
Note The dial peer on which the probe is configured will
be excluded from call routing only for outbound
calls. Inbound calls can arrive through this dial peer.
• rtr: Enables monitoring of proxy dial peers using RTR

probes.
• alternate-ip-address: (Optional) Enables monitoring of
alternate IP addresses other than the proxy address. For
example, to monitor a gateway front end to a SIP proxy.
Step 8 voice-class codec tag Sets the voice class codec parameters. The tag argument is
a codec group number between 1 and 10000.
Example:
Router(config-register-pool)# voice-class codec
15
Example:
Router(config-register-pool)# end
Configure Backup Registrar Service to SIP Phones (Using Optional

Commands)
The prior configurations set up a basic voice register pool. The configuration in this procedure adds
optional attributes to increase functionality. As part of this configuration, you can support:
• Translation Profile—Applies the translation profile to a specific directory number or to all directory
numbers on a SIP phone.
• Alias—Allows Cisco Unified SIP IP Phones to handle inbound PSTN calls to phone numbers that
are unavailable when the main SIP call control (IP-PBX) is not available.

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• Class of restriction (COR)—COR specifies which incoming dial peers can use which outgoing dial
peers to make a call. Each dial peer can be provisioned with an incoming and outgoing COR list.
Prerequisites
• Before configuring the 'alias' command, translation rules must be set using the translation-profile
outgoing (voice register pool) command.
SUMMARY STEPS
1. enable
4. translation-profile outgoing profile-tag
5. alias tag pattern to target [preference value]
6. cor {incoming | outgoing} cor-list-name {cor-list-number starting-number [- ending-number] |
default}
7. incoming called-number [number]
8. number tag number-pattern {preference value} [huntstop]
9. dtmf-relay [cisco-rtp] [rtp-nte] [sip-notify]
10. end
DETAILED STEPS

Example:
Router> enable
Example:
Step 3 voice register pool tag Enters voice register pool configuration mode.
• Use this command to control which registrations are
Example: accepted or rejected by a Cisco Unified SIP SRST
Router(config)# voice register pool 12 device.

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Step 4 translation-profile outgoing profile-tag Use this command to apply the translation profile to a
specific directory number or to all directory numbers on a
SIP phone.
Example:
Router(config-register-pool)# • Profile-tag: Translation profile name to handle
voice translation-rule 1 translation to outgoing calls.
rule 1 /1000/ /1006/
!
!
voice translation-profile 1
translate called 1
!
voice register pool xxx
translation-profile outgoing 1
Step 5 alias tag pattern to target [preference value] Allows Cisco Unified SIP IP Phones to handle inbound
PSTN calls to phone numbers that are unavailable when the
main proxy is not available. The keywords and arguments
Example:
Router(config-register-pool)# alias 1 94... to
are defined as follows:
91011 preference 8 • tag: Number from 1 to 5 and the distinguishing factor
when there are multiple alias commands.
• pattern: The prefix number; matches the incoming
phone number and may include wildcards.
• to: Connects the tag number pattern to the alternate
number.
• target: The target number; an alternate phone number to
route incoming calls to match the number pattern.
• preference value: (Optional) Assigns a dial-peer
preference value to the alias. The value argument is the
value of the associated dial peer, and the range is from
1 to 10. There is no default.

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Step 6 cor {incoming | outgoing} cor-list-name Configures a class of restriction (COR) on the VoIP dial
{cor-list-number starting-number [- peers associated with directory numbers. COR specifies
ending-number] | default}
which incoming dial peers can use which outgoing dial
peers to make a call. Each dial peer can be provisioned with
Example: an incoming and outgoing COR list. The keywords and
Router(config-register-pool)# cor incoming arguments are defined as follows:
call91 1 91011
• incoming: COR list to be used by incoming dial peers.
• outgoing: COR list to be used by outgoing dial peers.
• cor-list-name: COR list name.
• cor-list-number: COR list identifier. The maximum
number of COR lists that can be created is four,
comprised of incoming or outgoing dial peers.
• starting-number: Start of a directory number range, if
an ending number is included. Can also be a standalone
number.
• (Optional) Indicator that a full range is configured.
• ending-number: (Optional) End of a directory number
range.
• default: Instructs the router to use an existing default
COR list.
Step 7 incoming called-number [number] Applies incoming called parameters to dynamically created
dial peers. The number argument is optional and indicates a
sequence of digits that represent a phone number prefix.
Example:
Router(config-register-pool)# incoming
called-number 308
Step 8 number tag number-pattern {preference value} Indicates the E.164 phone numbers that the registrar permits
[huntstop] to handle the Register message from the Cisco Unified SIP
IP Phone. The keywords and arguments are defined as
Example: follows:
Router(config-register-pool)# number 1 50.. • tag: Number from 1 to 10 and the distinguishing factor
preference 2
when there are multiple number commands.
• number-pattern: Phone numbers (including wildcards
and patterns) that are permitted by the registrar to
handle the Register message from the SIP IP phone.
• preference value: (Optional) Defines the number list
preference order.
• huntstop: (Optional) Stops hunting if the dial peer is
busy.

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Step 9 dtmf-relay [cisco-rtp] [rtp-nte] [sip-notify] Specifies how a SIP gateway relays dual tone
multifrequency (DTMF) tones between telephony
interfaces and an IP network. The keywords are defined as
Example:
Router(config-register-pool)# dtmf-relay
follows:
rtp-nte • cisco-rtp: (Optional) Forwards DTMF tones by using
Real-Time Transport Protocol (RTP) with a Cisco
proprietary payload type.
• rtp-nte: (Optional) Forwards DTMF tones by using
RTP with the Named Telephone Event (NTE) payload
type.
• sip-notify: (Optional) Forwards DTMF tones using SIP
NOTIFY messages.
Example:
Verify SIP Registrar Configuration

To help you troubleshoot a SIP registrar and voice register pool, perform the following steps.
SUMMARY STEPS
1. debug voice register errors

2. debug voice register events
3. show sip-ua status registrar

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DETAILED STEPS

Step 1 debug voice register errors Use this command to debug errors that happen
during registration.
Example: If there are no voice register pools configured for a
Router# debug voice register errors particular registration request, the message “Contact
*Apr 22 11:52:54.523 PDT: VOICE_REG_POOL: Contact doesn’t match any pools” is displayed.
doesn't match any pools
*Apr 22 11:52:54.539 PDT: VOICE_REG_POOL: Register
request for (33015) from (10.2.152.39)
*Apr 22 11:52:54.539 PDT: VOICE_REG_POOL: Contact
doesn't match any pools.
request for (33017) from (10.2.152.39)
*Apr 22 11:53:04.559 PDT: VOICE_REG_POOL: Maximum
registration threshold for pool(3) hit
Step 2 debug voice register events Using the debug voice register events command
should suffice to display registration activity.
Registration activity includes matching of pools,
Example:
Router# debug voice register events
registration creation, and automatic creation of dial
Apr 22 10:50:21.731 PDT: VOICE_REG_POOL: Contact peers. For more details and error conditions, you can
matches pool 1 use the debug voice register errors command.
Apr 22 10:50:21.731 PDT: VOICE_REG_POOL: key(91011)
contact(192.168.0.2) add to contact table The phone number 91011 registered successfully,
Apr 22 10:50:21.731 PDT: VOICE_REG_POOL: key(91011) and type 1 is reported, which means there is a
exists in contact table pre-existing VoIP dial peer.
Apr 22 10:50:21.731 PDT: VOICE_REG_POOL:
contact(192.168.0.2) exists in contact table, ref
updated
Apr 22 10:50:21.731 PDT: VOICE_REG_POOL: Created
dial-peer entry of type 1
Registration successful for 91011, registration id
is 257
Step 3 show sip-ua status registrar Use this command to display all the SIP endpoints
currently registered with the contact address.
Example:
Router# show sip-ua status registrar
Line destination expires(sec) contact
======= =========== ============ =======
91021 192.168.0.3 227 192.168.0.3
91011 192.168.0.2 176 192.168.0.2
95021 10.2.161.50 419 10.2.161.50
95012 10.2.161.50 419 10.2.161.50
95011 10.2.161.50 420 10.2.161.50
95500 10.2.161.50 420 10.2.161.50
94011 10.2.161.40 128 10.2.161.40
94500 10.2.161.40 129 10.2.161.40

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Verify Proxy Dial-Peer Configuration

To use the icmp-ping keyword with the proxy command to assist in troubleshooting proxy dial peers,
SUMMARY STEPS
4. end
5. show voice register dial-peers
6. show dial-peer voice

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DETAILED STEPS

Step 1 configure terminal Use this command to enter global configuration
mode.
Example:
Step 2 voice register pool tag Use this command to enter voice register pool
configuration mode.
Example:
Step 3 proxy ip-address [preference value] [monitor probe Set the proxy command to monitor with icmp-ping.
{icmp-ping | rtr} [alternate-ip-address]]
Example:
Router(config-register-pool)# proxy 10.2.161.187
preference 1 monitor probe icmp-ping
Example:
Step 5 show voice register dial-peers Use this command to verify dial-peer
configurations, and notice that icmp-ping
monitoring is set.
Example:
Router# show voice register dial-peers
dial-peer voice 40035 voip
preference 5
destination-pattern 91011
session target ipv4:192.168.0.2
session protocol sipv2
voice-class codec 1

preference 1
voice-class codec 1
monitor probe icmp-ping 10.2.161.187

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Step 6 show dial-peer voice Use the show dial-peer voice command on dial peer
40036, and notice the monitor probe status.
Example:
Router# show dial-peer voice
VoiceOverIpPeer40036 Note Also highlighted is the output of the cor and
peer type = voice, information type = voice, incoming called-number commands.
description = `',
tag = 40036, destination-pattern = `91011',
answer-address = `', preference=1,
CLID Restriction = None
CLID Network Number = `'
CLID Second Number sent
source carrier-id = `', target carrier-id = `',
source trunk-group-label = `', target
trunk-group-label = `',
numbering Type = ùnknown'
group = 40036, Admin state is up, Operation state is
up,
incoming called-number = `', connections/maximum =
0/unlimited,
! Default output for incoming called-number command
DTMF Relay = disabled,
modem transport = system,
huntstop = disabled,
in bound application associated: 'DEFAULT'
out bound application associated: ''
dnis-map =
permission :both
incoming COR list:maximum capability
! Default output for cor command
outgoing COR list:minimum requirement
Translation profile (Incoming):
Translation profile (Outgoing):
incoming call blocking:
translation-profile = `'
disconnect-cause = `no-service'
advertise 0x40 capacity_update_timer 25 addrFamily 4
oldAddrFamily 4
type = voip, session-target = ìpv4:10.2.161.187',
technology prefix:
settle-call = disabled
ip media DSCP = ef, ip signaling DSCP = af31,
ip video rsvp-none DSCP = af41,ip video rsvp-pass
DSCP = af41
ip video rsvp-fail DSCP = af41,
UDP checksum = disabled,
session-protocol = sipv2, session-transport =
system,
req-qos = best-effort, acc-qos = best-effort,
req-qos video = best-effort, acc-qos video =
best-effort,
req-qos audio def bandwidth = 64, req-qos audio max
bandwidth = 0,
req-qos video def bandwidth = 384, req-qos video max
bandwidth = 0,

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RTP dynamic payload type values: NTE = 101
Cisco: NSE=100, fax=96, fax-ack=97, dtmf=121,
fax-relay=122
S=123, ClearChan=125, PCM switch over
u-law=0,A-law=8
RTP comfort noise payload type = 19
fax rate = voice, payload size = 20 bytes
fax protocol = system
fax-relay ecm enable
fax NSF = 0xAD0051 (default)
codec = g729r8, payload size = 20 bytes,
Media Setting = flow-through (global)
Expect factor = 0, Icpif = 20,
Playout Mode is set to adaptive,
Initial 60 ms, Max 300 ms
Playout-delay Minimum mode is set to default, value
40 ms
Fax nominal 300 ms
Max Redirects = 1, signaling-type = cas,
VAD = enabled, Poor QOV Trap = disabled,
Source Interface = NONE
voice class sip url = system,
voice class sip rel1xx = system,
monitor probe method: icmp-ping ip address:
10.2.161.187,
Monitored destination reachable
voice class perm tag = `'
Time elapsed since last clearing of voice call
statistics never
Connect Time = 0, Charged Units = 0,
Successful Calls = 0, Failed Calls = 0, Incomplete
Calls = 0
Accepted Calls = 0, Refused Calls = 0,
Last Disconnect Cause is "",
Last Disconnect Text is "",
Last Setup Time = 0.

Internet Protocol version 6 (IPv6) is the latest version of the Internet Protocol (IP). IPv6 uses packets to
exchange data, voice, and video traffic over digital networks. Also, IPv6 increases the number of
network address bits from 32 bits in IPv4 to 128 bits. From Unified SRST Release 12.0 onwards, Unified
SRST supports IPv6 protocols for SIP IP phones.
IPv6 support in Unified SRST allows the network to behave transparently in a dual-stack (IPv4 and IPv6)
environment and provides additional IP address space to SIP IP phones that are connected to the
network. If you do not have a dual-stack configuration, configure the CLI command call service stop
under voice service voip configuration mode before changing to dual-stack mode. For an example of
switching to dual-stack mode, see Examples for Configuring IPv6 Pools for SIP IP Phones, page 84.
The Cisco IP Phone 7800 Series and 8800 Series are supported on IPv6 for Unified SRST.

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For more information on configuring SIP IP phones for IPv6 source address, see Configure IPv6 Pools
For an example of configuring IPv6 Support on Unified SRST, see Examples for Configuring IPv6 Pools
For more details about IPv6 deployment, see IPv6 Deployment Guide for Cisco Collaboration Systems
Release 12.0.
Feature Support for IPv6 in Unified SRST SIP IP Phones

The following basic features are supported for a IPv6 WAN down scenario:
• Basic SIP Line (IPv4 or IPv6) to SIP Line calls (IPv4 or IPv6) when Unified SRST is in dual-stack
no anat mode.
The following supplementary services are supported as part of IPv6 in Unified SRST IP Phones:
• Hold/Resume
• Call Forward
• Call Transfer
• Three-way Conference (with BIB conferencing only)
• Line to T1/E1 Trunk and Trunk to Line with Supplementary Service Features
• Fax to and from PSTN (IPv4 ATA to ISDN T1/E1) for both T.38 Fax Relay and Fax Passthrough
Restrictions
The following are the known restrictions for IPv6 support on Unified SRST:
• SIP Trunks are not supported on Unified SRST for IPv6 deployment. PSTN calls are supported only
through T1/E1 trunks.
• SCCP IP Phones are not supported in a deployment of IPv6 for Unified SRST.
• SIP Phones can be either in IPv4 only or IPv6 only mode (no anat).
• Trancoding and Transrating are not supported.
• H.323 trunks are not supported.
• Secure SIP lines or trunks are not supported.
• IPv6 on Unified SRST is not supported on the Cisco IOS platform. The support is restricted to Cisco
IOS XE platform with Cisco IOS Release 16.6.1 or later versions.
Configure IPv6 Pools for SIP IP Phones

Before You Begin
• Unified SRST 12.0 or a later version.
• IPv6 option only appears if protocol mode is dual-stack configured under sip-ua configuration mode
or IPv6.
• Cisco Unified SRST License must be configured for the gateway to function as a Unified SRST
gateway to support IPv6 functionality. For more information on licenses, see Licensing, page 41.

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• Cisco Unified Communications Manager (Unified Communications Manager) is provisioned with

the IPv6 address of Unified SRST. For information on configuration of Unified SRST on Unified
Communications Manager, see the section Survivable Remote Site Telephony Configuration in Cisco
Unified Communications Manager Administration Guide.
SUMMARY STEPS
1. enable
3. ipv6 unicast-routing
5. sip
6. no anat
7. call service stop
8. exit
9. exit
10. sip-ua
11. protocol mode {ipv4 | ipv6 | dual-stack [preference {ipv4 | ipv6}]}
12. exit
13. voice service {voip}
14. sip
15. no call service stop
16. exit
18. default mode
21. exit
23. id { network address mask mask | ip address mask mask | mac address }
24. end

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DETAILED STEPS

Example:
Router> enable
Example:
Router #configure terminal
Step 3 ipv6 unicast-routing Enables the forwarding of IPv6 unicast datagrams.
Example:
Router(config)# ipv6 unicast-routing
Step 4 voice service voip Enters voice-service configuration mode to specify a voice
encapsulation type.
Example:
Example:
Step 6 no anat Disables Alternative Network Address Types (ANAT) on a

SIP trunk.
Example:
Router(config-serv-sip)# no anat
Step 7 call service stop Shuts down SIP call service.
Example:
Example:

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Step 9 exit Exits voice service voip configuration mode.
Example:
Router(config-voi-serv)# exit
Example:
Step 11 protocol mode {ipv4 | ipv6 | dual-stack Allows phones to interact with phones on IPv6 voice
[preference {ipv4 | ipv6}]} gateways. You can configure phones for IPv4 addresses,
IPv6 addresses, or for a dual-stack mode.
• ipv4—Allows you to set the protocol mode as an IPv4
Example: address.
Router(config-sip-ua)# protocol mode dual-stack
preference ipv6 • ipv6—Allows you to set the protocol mode as an IPv6
address.
• dual-stack—Allows you to set the protocol mode for
both IPv4 and IPv6 addresses.
• preference—Allows you to choose a preferred IP
address family if protocol mode is dual-stack.
Example:
Step 13 voice service {voip} Enters voice-service configuration mode to specify a voice
encapsulation type.
Example:
Example:

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Step 15 no call service stop Activates SIP call service.
Example:
Example:
parameters for all supported SIP phones in Unified SRST.
Example:
Example:
router.
Example:
Example:
Example:

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Configure Unified SRST on Cisco 4000 Series Integrated Services Platform

mask mask | mac address } phone to support a degree of authentication.
Example:
2001:420:54FF:13::901:0/117
10.64.88.0 mask 255.255.255.0
Example:
Router(config)# end
Configure Unified SRST on Cisco 4000 Series Integrated

Services Platform
For Unified SRST Release 10.5 and later, Unified SRST is supported on Cisco 4000 Series Integrated
Services Routers. A Unified SRST system supports SIP phones with standard-based RFC 3261 feature
support locally and across SIP WAN networks. With Cisco Unified SIP SRST, SIP phones can place calls
across SIP networks with similar features, as SCCP phones do. For example, most SCCP phone features
such as caller ID, speed dial, and redial are supported on SIP networks, that give users the opportunity
to choose SCCP or SIP.
Prerequisites
• Cisco IOS XE Denali 16.3.1 or a later release.

• Cisco IP Phones 7800 Series or 8800 Series.
• An appropriate feature license to support Unified SIP SRST on the router.
• You need to configure voice register global in your router.
• You need to ensure that your router is in default mode (for Unified SRST).
Restrictions
• For a list of restrictions for Unified SIP SRST support on Cisco 4000 Series Integrated Services
Routers, see Restrictions of Unified SRST on Cisco 4000 Series Integrated Services Routers,
page 39.
SUMMARY STEPS
1. enable

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4. allow-connections from-type to to-type
5. no supplementary-service sip moved-temporarily
6. no supplementary-service sip refer
7. supplementary-service media-renegotiate
8. sip
9. registrar server [expires[max sec][min sec]]
10. exit
11. exit
13. default mode
16. exit
18. id [network address mask mask | ip address mask mask]
19. dtmf-relay [rtp-nte]
20. codec codec
21. no vad
22. end
DETAILED STEPS

Example:
Router> enable
Example:
Step 3 voice service voip Enters voice-service configuration mode and specifies
voice-over-IP encapsulation.
Example: Enters voice register global configuration mode to set
Router(config)# voice service voip global parameters for all supported Cisco SIP IP phones in
a Cisco Unified SIP SRST environment.

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Step 4 allow-connections from-type to to-type Allows connections between specific types of endpoints in
a VoIP network.
Example:
Router(config-voi-serv)# allow-connections sip
to sip
Step 5 no supplementary-service sip moved-temporarily Disables supplementary service for call forwarding.
Example:
Router(config-voi-serv)# no
supplementary-service sip moved-temporarily
Step 6 no supplementary-service sip moved-temporarily Prevents the router from forwarding a REFER message to
the destination for call transfers.
Example:
Router(config-voi-serv)# no
supplementary-service sip refer
Step 7 supplementary-service media-renegotiate Enables mid-call media renegotiation for supplementary
services.
Example:
Router(config-voi-serv)# supplementary-service
media-renegotiate
• Required only if you perform the following step for
Example: enabling the SIP registrar function.
Step 9 registrar server [expires[max sec][min sec]] Enables SIP registrar functionality in Unified SRST.
• expires: (Optional) Sets the active time for an incoming
Example: registration.
Router(config-serv-sip)# registrar server
expires max 120 min 60
• max sec—(Optional) Maximum time for a registration
to expire, in seconds. Range: 600 to 86400. Default:
3600. Recommended value: 600.
The default is 60.
Example:
Step 11 exit Exits voice-service configuration mode.
Example:
Router(config-voi-serv)# exit
parameters for all supported SIP phones in Unified SRST.
Example:

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Example:
router.
Example:
Example:
mask mask } phone to support a degree of authentication.
Example:
10.64.88.0 mask 255.255.255.0
Step 19 dtmf-relay rtp-nte Forwards DTMF tones by using Real-Time Transport
Protocol (RTP) with the Named Telephone Event (NTE)
payload type and enables DTMF relay using the RFC 2833
Example: standard method.
rtp-nte
Step 20 no vad Disables voice activity detection (VAD) on the VoIP dial
peer.
Example: • VAD is enabled by default. Because there is no comfort
Router(config-register-pool)# no vad noise during periods of silence, the call may seem to be
disconnected. You may prefer to set no vad on the SIP
phone pool.

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Configure Voice Hunt Groups on Unified SRST

Step 21 codec codec-type [bytes] Specifies the codec supported by a single SIP phone or a
VoIP dial peer in a Cisco Unified SIP SRST environment.
The codec-type argument specifies the preferred codec and
Example:
Router(config-register-pool)# codec g729r8
can be one of the following:
• g711alaw: G.711 a–law 64,000 bps.
• g711ulaw: G.711 mu–law 64,000 bps.
• g729r8: G.729 8000 bps (default).
The bytes argument is optional and specifies the number of
bytes in the voice payload of each frame
Example:

To redirect calls for a specific number (pilot number) to a defined group of directory numbers on Cisco
Unified SCCP and SIP IP phones, perform the following steps.
Voice Hunt Group on Unified SRST is supported for Parallel, Sequential, Peer, and Longest-Idle hunt
groups. Only the basic call features such as Call, Hold or Resume are supported for Unified SRST on
Cisco 4000 Series Integrated Services Routers. For support of advanced features such as Auto Logout,
Members Logout, and supplementary call features, you need to configure Unified E-SRST. For more
information on Voice Hunt Group support on Unified E-SRST, see Unified E-SRST with Support for
Voice Hunt Group, page 88.
For a list of restrictions of Unified SRST on Cisco 4000 Series Integrated Services Routers, see
Restrictions of Unified SRST on Cisco 4000 Series Integrated Services Routers, page 39
Prerequisites
• Cisco IOS XE Denali 16.3.1 or later versions.
• Shared Lines are not supported on Unified SRST.
SUMMARY STEPS
1. enable
3. voice hunt-group hunt-tag [longest-idle | parallel | peer | sequential]
4. pilot number [secondary number]
5. list number
6. final number
7. preference preference-order [secondary secondary-order]
8. timeout seconds

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9. end
DETAILED STEPS

Example:
Router> enable
Example:
Step 3 voice hunt-group hunt-tag [longest-idle | Enters voice hunt-group configuration mode to define a
parallel | peer | sequential] hunt group.
• hunt-tag—Unique sequence number of the hunt group
Example: to be configured. Range is 1 to 100.
Router(config)# voice hunt-group 1 longest-idle
• longest idle—Hunt group in which calls go to the
directory number that has been idle for the longest time.
• parallel—Hunt group in which calls simultaneously
ring multiple phones.
• peer—Hunt group in which the first directory number
is selected round-robin from the list.
• sequential—Hunt group in which directory numbers
ring in the order in which they are listed, left to right.
• To change the hunt-group type, remove the existing
hunt group first by using the no form of the command;
then, recreate the group.
Step 4 pilot number [secondary number] Defines the phone number that callers dial to reach a voice
hunt group.
Example: • number—String of up to 16 characters that represents
Router(config-voice-hunt-group)# pilot number an E.164 phone number.
8100
• Number string may contain alphabetic characters when
the number is to be dialed only by the Unified SRST
router, as with an intercom number, and not from phone
keypads.
• secondary number—(Optional) Keyword and
argument combination defines the number that follows
as an additional pilot number for the voice hunt group.
• Secondary numbers can contain wildcards. A wildcard
is a period (.), which matches any entered digit.

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Step 5 list number Creates a list of extensions that are members of a voice hunt
group. To remove a list from a router configuration, use the
no form of this command.
Example:
Router(config-voice-hunt-group)# list 8000, • number—List of extensions to be added as members to
8010, 8020, 8030 the voice hunt group. Separate the extensions with
commas.
• Add or delete all extensions in a hunt-group list at one
time. You cannot add or delete a single number in an
existing list.
• There must be from 2 to 10 extensions in the
hunt-group list, and each number must be a primary or
secondary number.
• Any number in the list cannot be a pilot number of a
parallel hunt group.
Step 6 final number Defines the last extension in a voice hunt group.
• If a final number in one hunt group is configured as a
Example: pilot number of another hunt group, the pilot number of
Router(config-voice-hunt-group)# final 8888 the first hunt group cannot be configured as a final
number in any other hunt group.
Step 7 preference preference-order [secondary Sets the preference order for the directory number
secondary-order] associated with a voice hunt-group pilot number.
Note We recommend that the parallel hunt-group pilot
Example: number be unique in the system. Parallel hunt
Router(config-voice-hunt-group)# preference 6 groups may not work if there are more than one
partial or exact dial-peer match. For example, if the
pilot number is “8000” and there is another dial peer
that matches “8…”. If multiple matches cannot be
avoided, give parallel hunt groups the highest
priority to run by assigning a lower preference to the
other dial peers. Note that 8 is the lowest preference
value. By default, dial peers created by parallel hunt
groups have a preference of 0.
• preference-order—Range is 0 to 8, where 0 is the

highest preference and 8 is the lowest preference.
Default is 0.
• secondary secondary-order—(Optional) Keyword and
argument combination is used to set the preference
order for the secondary pilot number. Range is 1 to 8,
where 0 is the highest preference and 8 is the lowest
preference. Default is 7.

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Step 8 hops number For configuring a peer or longest-idle voice hunt group
only. Defines the number of times that a call can hop to the
next number in a peer or longest-idle voice hunt group
Example:
Router(config-voice-hunt-group)# hops 2
before the call proceeds to the final number.
• number—Number of hops. Range is 2 to 10, and the
value must be less than or equal to the number of
extensions specified by the list command.
• Default is the same number as there are destinations
defined under the list command.
Step 9 timeout seconds Defines the number of seconds after which a call that is not
answered is redirected to the next directory number in a
voice hunt-group list.
Example:
Router(config-voice-hunt-group)# timeout 100 • Default: 180 seconds.
Example:
Router(config-voice-hunt-group)# end
Configure Feature Support on Unified SIP SRST

This section provides configuration information for some of the features supported on Unified SIP
SRST.
Configure SIP-to-SIP Call Forwarding

SIP-to-SIP call forwarding (call routing) is available. Call forwarding is provided either by the phone or
by using a back-to-back user agent (B2BUA), which allows call forwarding on any dial peer. Calls into
a SIP device may be forwarded to other SIP or SCCP devices (including Cisco Unity, third-party
voice-mail systems, or an auto attendant or IVR system such as IPCC and IPCC Express). In addition,
SCCP IP phones may be forwarded to SIP phones.
Cisco Unity or other voice messaging systems connected by a SIP trunk or SIP user agent are able to
pass a message-waiting indicator (MWI) when a message is left. The SIP phone then displays the MWI
when indicated by the voice messaging system.
Note SIP-to-H.323 call forwarding is not supported.
To configure SIP-to-SIP call forwarding, you must first allow connections between specific types of
endpoints in a Cisco IP-to-IP gateway. The allow-connections command grants this capability. Once the
SIP-to-SIP connections are allowed, you can configure call forwarding under an individual SIP phone
pool. Any of the following commands can be used to configure call forwarding, according to your needs:
• Under voice register pool
– call-forward b2bua all directory-number

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– call-forward b2bua busy directory-number

– call-forward b2bua mailbox directory-number
– call-forward b2bua noan directory-number [timeout seconds]
In a typical Cisco Unified SIP SRST setup, the call-forward b2bua mailbox command is not used;
however, it is likely to be used in a Cisco Unified SIP Communications Manager Express (CME)
environment. Detailed procedures for configuring the call-forward b2bua mailbox command are found
in the Cisco Unified Communications Manager (CallManager) documentation on Cisco.com.
The command call-forward b2bua all needs to point towards the trunk.
SUMMARY STEPS
1. enable
4. call-forward b2bua all directory-number
5. call-forward b2bua busy directory-number
6. call-forward b2bua mailbox directory-number
7. call-forward b2bua noan directory-number timeout seconds
8. end
DETAILED STEPS

Example:
Router> enable
Example:
• Use this command to control which phone registrations
Example: are accepted or rejected by a Cisco Unified SIP SRST
Step 4 call-forward b2bua all directory-number Enables call forwarding for a SIP back-to-back user agent
(B2BUA) so that all incoming calls are forwarded to
another non-SIP station extension (that is, SIP trunk, H.323
Example:
Router(config-register-pool)# call-forward
trunk, SCCP device or analog/digital trunk).
b2bua all 5005 • directory-number: Phone number to which calls are
forwarded. Represents a fully qualified E.164 number.
Maximum length of the phone number is 32.

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Step 5 call-forward b2bua busy directory-number Enables call forwarding for a SIP B2BUA so that incoming
calls to a busy extension are forwarded to another extension.
Example: • directory-number: Phone number to which calls are
Router(config-register-pool)# call-forward forwarded. Represents a fully qualified E.164 number.
b2bua busy 5006 Maximum length of the phone number is 32.
Step 6 call-forward b2bua mailbox directory-number Controls the specific voice-mail box selected in a
voice-mail system at the end of a call forwarding exchange.
Example: • directory-number: Phone number to which calls are
Router(config-register-pool)# call-forward forwarded when the forwarded destination is busy or
b2bua mailbox 5007 does not answer. Represents a fully qualified E.164
number.
Step 7 call-forward b2bua noan directory-number Enables call forwarding for a SIP B2BUA so that incoming
timeout seconds calls to an extension that does not answer after a configured
amount of time are forwarded to another extension.
Example: This command is used if a phone is registered with a
Router(config-register-pool)# call-forward Cisco Unified SIP SRST router, but the phone is not
b2bua noan 5010 timeout 10
reachable because there is no IP connectivity (there is no
response to Invite requests).
• directory-number: Phone number to which calls are
• timeout seconds: Duration, in seconds, that a call can
ring with no answer before the call is forwarded to
another extension. Range is 3 to 60000.
The default value is 20.
Example:
Configure Call Blocking Based on Time of Day, Day of Week, or Date

This section applies to both SCCP and SIP SRST. Call blocking prevents the unauthorized use of phones
and is implemented by matching a pattern of up to 32 digits during a specified time of day, day of week,
or date. Cisco Unified SIP SRST provides SIP endpoints the same time-based call blocking mechanism
that is currently provided for SCCP phones. The call blocking feature supports all incoming calls,
including incoming SIP and analog FXS calls.
Note Pin-based exemptions and the “Login” toll-bar override are not supported in Cisco Unified SIP SRST.

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The commands used for SIP phone call blocking are the same commands that are used for SCCP phones
on your Cisco Unified SRST system. The Cisco SRST session application accesses the current
after-hours configuration under call-manager-fallback mode and applies it to calls originated by
Cisco SIP phones that are registered to the Cisco SRST router. The commands used in
call-manager-fallback mode that set block criteria (time/date/block pattern) are the following:
• after-hours block pattern pattern-tag pattern [7-24]
• after-hours day day start-time stop-time
• after-hours date month date start-time stop-time
When a user attempts to place a call to digits that match a pattern that has been specified for call blocking
during a time period that has been defined for call blocking, the call is immediately terminated and the
caller hears a fast busy.
In SRST (call-manager-fallback configuration mode), there is no phone- or pin-based exemption to
after-hours call blocking. However, in Cisco Unified SIP SRST (voice register pool mode), individual IP
phones can be exempted from all call blocking using the after-hours exempt command.
SUMMARY STEPS
1. enable
4. after-hours block pattern tag pattern [7-24]
5. after-hours day day start-time stop-time
6. after-hours date month date start-time stop-time
7. exit
9. after-hour exempt
10. end
DETAILED STEPS

Example:
Router> enable
Example:

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Example:
Step 4 after-hours block pattern tag pattern [7-24] Defines a pattern of outgoing digits to be blocked. Up to 32
patterns can be defined, using individual commands.
Example: • If the 7-24 keyword is specified, the pattern is always
Router(config-cm-fallback)# after-hours block blocked, 7 days a week, 24 hours a day.
pattern 1 91900
• If the 7-24 keyword is not specified, the pattern is
blocked during the days and dates that are defined using
the after-hours day and after-hours date commands.
Step 5 after-hours day day start-time stop-time Defines a recurring time period based on the day of the
week during which calls are blocked to outgoing dial
patterns that are defined using the after-hours block
Example:
Router(config-cm-fallback)# after-hours day mon
pattern command.
19:00 07:00 • day: Day of the week abbreviation. The following are
valid day abbreviations: sun, mon, tue, wed, thu, fri,
sat.
• start-time stop-time: Beginning and ending times for
call blocking, in an HH:MM format using a 24-hour
clock. If the stop time is a smaller value than the start
time, the stop time occurs on the day following the start
time. For example, “mon 19:00 07:00” means “from
Monday at 7 p.m. until Tuesday at 7 a.m.”
The value 24:00 is not valid. If 00:00 is entered as a
stop time, it is changed to 23:59. If 00:00 is entered for
both start time and stop time, calls are blocked for the
entire 24-hour period on the specified date.
Step 6 after-hours date month date start-time Defines a recurring time period based on month and date
stop-time during which calls are blocked to outgoing dial patterns that
are defined using the after-hours block pattern command.
Example: • month: Month abbreviation. The following are valid
Router(config-cm-fallback)# after-hours date month abbreviations: jan, feb, mar, apr, may, jun, jul,
jan 1 00:00 00:00
aug, sep, oct, nov, dec.
• date: Date of the month. Range is from 1 to 31.
clock. The stop time must be larger than the start time.

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Example:
Step 9 after-hour exempt Specifies that for a particular voice register pool, none of its
outgoing calls are blocked although call blocking is
enabled.
Example:
Router(config-register-pool)# after-hour exempt
Example:
Verification
To verify the feature’s configuration, enter one of the following commands:
• show voice register dial-peer: Displays all the dial peers created dynamically by phones that have
registered. This command also displays configurations for after hours blocking and call forwarding.
• show voice register pool <tag>: Displays information about a specific pool.
• debug ccsip message: Debugs basic B2BUA calls.
For more information about these commands, see Cisco Unified SRST and Cisco Unified SIP SRST

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SIP Call Hold and Resume

Unified SRST supports the ability for SIP phones to place calls on hold and to resume from calls placed
on hold. This also includes support for a consultative hold where A calls B, B places A on hold, B calls
C, and B disconnects from C and then resumes with A. Support for call hold is signaled by SIP phones
using “re-INVITE c=0.0.0.0” and also by the receive-only mechanism.
Configure Music On Hold for Unified SRST

Unified SRST supports the ability for SIP phones to play music for calls placed on hold. The following
is the recommended configuration for Music On Hold (MOH) on a SIP Phone that falls back to Unified
SRST.
SUMMARY STEPS
1. enable
3. no telephony-service
5. moh enable-g711 "flash:filename"
6. moh g729 "flash:filename"
7. end
DETAILED STEPS

Example:
Router> enable
Example:
Step 3 no telephony-service Removes all the configurations for IP phones configured
under the telephony-service configuration mode.
Example:
Router# no telephony-service

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Example:
Step 5 moh enable-g711 "bootflash:filename" Generates an audio stream from a router flash file that
supports G.711 codec for Music On Hold (MOH) in Unified
SRST.
Example:
Router(config-cm-fallback)# moh enable-g711
"bootflash:music-on-hold.au"
Step 6 moh g729 "bootflash:filename" Generates an audio stream from a router flash file that
supports G.729 codec for MOH in Unified SRST.
Example:
Router(config-cm-fallback)# moh g729
"flash:SampleAudioSource.g729.wav"
Example:
Router(config-cm-fallback)# end
Enabling KPML for SIP Phones

Perform the following steps to enable KPML digit collection on a SIP phone.
Restrictions
• A dial plan assigned to a phone has priority over KPML.
SUMMARY STEPS
1. enable
4. digit collect kpml
5. end
6. show voice register dial-peer

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DETAILED STEPS

Example:
Router> enable
Example:
Example: • pool-tag: Unique sequence number of the SIP phone to
Router(config)# voice register pool 4 be configured. Range is version and
platform-dependent; type ? to display range. You can
modify the upper limit for this argument with the
max-pool command.
Step 4 digit collect kpml Enables KPML digit collection for the SIP phone.
Note This command is enabled by default for supported
Example: phones in Cisco Unified CME and
Router(config-register-pool)# digit collect Cisco Unified SRST.
kpml
Example:
Step 6 show voice register dial-peers Displays details of all dynamically created VoIP dial peers
associated with the Cisco Unified CME SIP register
including the defined digit collection method.
Example:
Disabling SIP Supplementary Services for Call Forward and Call Transfer
Perform the following steps to disable REFER messages for call transfers and redirect responses for call
forwarding from being sent to the destination by Unified SRST. You can disable these supplementary
features if the destination gateway does not support them.
SUMMARY STEPS
1. enable
or
dial-peer voice tag voip

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4. no supplementary-service sip {moved-temporarily | refer}

5. end
DETAILED STEPS

Example:
Router> enable
Example:
Step 3 voice service voip Enters voice-service configuration mode to set global
or parameters for VoIP features.
dial-peer voice tag voip or
Enters dial peer configuration mode to set parameters for a
Example: specific dial peer.
or
Router(config)# dial-peer voice 99 voip
Step 4 no supplementary-service sip {moved-temporarily Disables SIP call forwarding or call transfer supplementary
| refer} services globally or for a dial peer.
• moved-temporarily: SIP redirect response for call
Example: forwarding.
Router(conf-voi-serv)# no supplementary-service
sip refer • refer: SIP REFER message for call transfers.
or • Sending REFER and redirect messages to the
Router(config-dial-peer)# no destination is the default behavior.
Note This command is supported for calls between SIP
phones and calls between SCCP phones. It is not
supported for a mixture of SCCP and SIP endpoints.
Example:
Router(config-voi-serv)# end
or
Router(config-dial-peer)# end
Configuring Idle Prompt Status for SIP Phones

Perform the following steps to customize the message that displays on SIP phones after the phones
failover to Cisco Unified SRST.

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Note You do not need to create new configuration files with the create profile command and restart the phones
after changing the idle status message in Cisco Unified SRST. Modifying the status message takes effect
immediately in Cisco Unified SRST.
SUMMARY STEPS
1. enable
4. system message string
5. end
6. show voice register global
DETAILED STEPS

Example:
Router> enable
Example:
global parameters for all supported SIP phones in a
Cisco Unified CME environment.
Example:
Step 4 system message string Defines a status message that displays on SIP phones
registered to Cisco Unified SRST.
Example: • string: Up to 32 alphanumeric characters. Default is
Router(config-register-global)# system message “CM Fallback Service Operating.”
fallback active
Example:
Router(config-register-global)# end
Step 6 show voice register global Displays all global configuration parameters associated
with SIP phones.
Example:
Router# show voice register global

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Examples
The following are sample configurations for supporting SIP SRST on Cisco 4000 Series Integrated
Services Router.
Example for Configuring Unified SIP SRST on Cisco 4000 Series

Integrated Services Routers
The following example shows how to configure Unified SIP SRST on Cisco 4000 Series Integrated
Services Routers.
!
voice service voip
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
supplementary-service media-renegotiate
sip
registrar server expires max 120 min 60
!
!
default mode
max-dn 40
max-pool 40
!
id network 8.55.0.0 mask 255.255.0.0
dtmf-relay rtp-nte
codec g711ulaw
no vad
!
!
Example for Configuring Voice Hunt Groups in Unified SIP SRST

The following example shows how to configure longest-idle hunt group 20 with pilot number 4701, final
number 5000, and 6 numbers in the list. After a call is redirected six times (makes 6 hops), it is redirected
to the final number 5000.
Router(config-voice-hunt-group)# pilot 4701
Router(config-voice-hunt-group)# list 4001, 4002, 4023, 4028, 4045, 4062
Router(config-voice-hunt-group)# final 5000
Router(config-voice-hunt-group)# timeout 20
Router(config-voice-hunt-group)# exit

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Examples for Configuring IPv6 Pools for SIP IP Phones

The following example provides configuration of IPv6 pools for SIP IP Phones:
ipv6 unicast-routing
voice service voip
sip
no anat
call service stop
exit
exit
sip-ua
protocol mode dual-stack
exit
voice service voip
sip
no call service stop
exit
default mode
max-dn 50
max-pool 40
exit
id network 2001:420:54FF:13::901:0/117
end
The following example provides interface configuration for IPv6 supported on Unified SRST:
configure terminal
ip address 10.64.86.229 255.255.255.0
negotiation auto
ipv6 address 2001:420:54FF:13::312:82/119
ipv6 enable
The following example provides IP route configuration for IPv6 supported on Unified SRST:
ipv6 route 2001:420:54FF:13::312:0/119 2001:420:54FF:13::312:1

ipv6 route 2001:420:54FF:13::901:0/119 2001:420:54FF:13::312:1
The following example displays output when SIP call service is shut down with the call service stop
CLI command:
Router# show sip service

SIP service is shut
The following example displays output when SIP call service is active with the no call service stop
CLI command:
Router# show sip-ua service

SIP Service is up

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Example for Configuring Call Blocking Based on Time of Day, Day of

Week, or Date
The following example defines several patterns of digits for which outgoing calls are blocked. Patterns 1
and 2, which block calls to external numbers that begin with 1 and 011, are blocked on Monday through
Friday before 7 a.m. and after 7 p.m. Pattern 3 blocks calls to 900 numbers 7 days a week, 24 hours a day.
call-manager-fallback
after-hours block pattern 1 91
after-hours block pattern 3 91900 7-24
after-hours day mon 19:00 07:00
after-hours day tue 19:00 07:00
after-hours day wed 19:00 07:00
after-hours day thu 19:00 07:00
after-hours day fri 19:00 07:00
The following example exempts a Cisco SIP phone pool from the configured blocking criteria:
after-hour exempt
Example for Configuring Music On Hold for Unified SIP SRST

The following example shows how to configure Music On Hold (MOH) for Unified SIP SRST on Cisco
4000 Series Integrated Services Routers.
enable
configure terminal
no telephony-service
moh enable-g711 "flash:music-on-hold.au"
moh g729 "flash:SampleAudioSource.g729.wav"
Example for Configuring SIP-to-SIP Call Forwarding on Unified SRST

The following is a sample configuration for SIP-to-SIP Call Forwarding on Unified SRST.
enable
configure terminal
call-forward b2bua busy 5006
call-forward b2bua mailbox 5007
call-forward b2bua noan 5010 timeout 8
Example for Configuring Idle Prompt Status for SIP Phones

The following is a sample configuration for idle prompt status for SIP phones on Unified SRST.
enable
configure terminal
system message fallback active
end

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show voice register global
Example for Disabling SIP Supplementary Services for Call Forward and
Call Transfer
The following is a sample configuration for disabling SIP supplementary services for call forward and
call transfer on Unified SRST.
enable
configure terminal
voice service voip
no supplementary-service sip {moved-temporarily | refer}
end

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CHAPTER 3
Cisco Unified Enhanced Survivable Remote
Site Telephony
This chapter describes the Unified Enhanced Survivable Remote Site Telephony (Unified E-SRST)
feature which is an enhancement of the SRST feature that provides advanced services compared to the
classic Unified SRST.
Contents
• Migration from Unified SRST Manager to Unified E-SRST, page 87
• Unified E-SRST with Support for Voice Hunt Group, page 88
• SCCP: Configure Unified E-SRST, page 106
• Configure Digest Credentials On Unified Communications Manager, page 112
• Unified E-SRST Scale Support, page 113
Migration from Unified SRST Manager to Unified E-SRST

From Cisco IOS XE Fuji 16.8.1 Release, Cisco Unified Survivable Remote Site Telephony Manager is
End-of-Life (EOL). Hence, provisioning for Unified E-SRST through Unified SRST Manager is not
supported for Unified E-SRST Release 12.2 and later releases. Unified E-SRST is provisioned only
using CLI commands (manual provisioning) to support fall back of phones registered to Unified
Communications Manager. For more information on configuring Unified E-SRST, see SIP: Configure
Unified E-SRST, page 90 and SCCP: Configure Unified E-SRST, page 106. For information on Cisco
Unified Survivable Remote Site Telephony Manager End-of-Life announcement, see Cisco Unified
Survivable Remote Site Telephony Manager Product Bulletin.
Unified SRST Manager is a GUI-based tool that helps to monitor, report, and troubleshoot remote sites.
It performs automatic sync up between the Unified Communications Manager and the Unified E-SRST
gateway that helps in adding, deleting, and modifying the users and phones including dial-plan mapping.
It also provides centralized management and control of all remote sites. For more information on the
Unified SRST Manager that is End-of-Life, see Administration Guide for Cisco Unified SRST Manager.

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Chapter 3 Cisco Unified Enhanced Survivable Remote Site Telephony
Unified E-SRST with Support for Voice Hunt Group
Benefits of Unified-ESRST
When you configure Unified E-SRST, it provides the following feature benefits in comparison to the
classic Unified SRST:
• Voice Hunt Group
– Shared Lines
– Mixed Shared Lines (SIP and SCCP Phones)
– Hunt Statistics Collection
– Mixed Deployment (SIP and SCCP Phones)
• Shared Line
• BLF
• Video
• B-ACD
For more information on configuring VHG with Unified E-SRST, see Unified E-SRST with Support for
Voice Hunt Group, page 88. For more information on configuring Shared Line, BLF, and Video with
Unified E-SRST, see SIP: Configure Unified E-SRST, page 90.

From Unified E-SRST Release 12.2, support is introduced for Voice Hunt Group with Cisco Unified
Enhanced Survivable Remote Site Telephony (Unified E-SRST). The deployment is supported on SIP
and SCCP phones. The Cisco IP Phone 7800 and 8800 Series are the SIP phones supported for this
deployment. The Unified E-SRST deployment for voice hunt group enhancement is introduced on the
Cisco 4000 Series Integrated Services Routers.
As part of the enhancement, voice hunt group features are supported in the E-SRST mode. Voice hunt
group deployments with Sequential, Parallel, Longest Idle, and Peer call blasting is supported with
Unified E-SRST 12.2 and onwards.
During a WAN outage, the SIP phones on the Cisco Unified Communications Manager (Unified
Communications Manager) fall back to Unified E-SRST router in mode esrst. The SIP phones are
logged in to the hunt group by default in this scenario. However, if the CLI command members logout
is configured under the voice hunt group configuration mode, the phones will be in logged out state. In
the Unified E-SRST mode, the phone that falls back on Unified E-SRST can toggle state and log in (or
log out) to the voice hunt group using HLog via Feature Access Code (FAC). The DN status (logged in
or logged out) is displayed on the phones registered with Unified E-SRST. The following FAC codes are
available as part of the enhancement introduced on Unified E-SRST:
• FAC Standard (Code: *5)
• FAC Custom (Code: Customizable, with a maximum character string length of 10. For example, *89,
• 8888888888)
When the user inputs FAC from a phone with multiple lines, the log out behavior is known to be different
across a deployment with common voice register pool configuration and individual voice register pool
configuration.
– Common Voice Register Pool Configuration — The DN's log out individually, and not at the
phone level.

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– Individual Voice Register Pool Configuration — The DN's log out at the phone level,
irrespective of the DN (primary, secondary, and so on) from which FAC input was provided by
the user.
When the WAN is available, the phones register back with Unified Communications Manager. For a
sample configuration of Unified E-SRST with voice hunt group enhancements, see Example for
configuring Unified E- SRST with Voice Hunt Group Enhancements, page 101.
The Unified E-SRST 12.2 Release introduces support for voice hunt group with shared lines and mixed
shared lines (SCCP and SIP phones). For a mixed shared line supported with voice hunt group, only
individual voice register pools can be configured. Common voice register pools are not supported. For a
sample configuration of mixed shared lines configured for a voice hunt group on Unified E-SRST, see
Example for Configuring Shared Line with Voice Hunt Group on Unified E-SRST, page 103.
Also, hunt statistic collection is supported for Unified E-SRST 12.2 and later releases.
A mixed deployment of SIP and SCCP phones is supported on Unified E-SRST, Release 12.2. Hunt
Group Logout from a mixed deployment of SIP and SCCP phones is supported using:
• FAC
• Feature Button, or
• DND
Line level logout and phone level logout is supported using FAC (*4).
Note Hunt Group logout is not supported for shared lines. Shared lines retain their logged in status.
Support for B-ACD in Unified E-SRST

B-ACD is supported as part of the Unified E-SRST enhancement introduced in Release 12.2. For SIP
phones that fall back to Unified E-SRST router in mode esrst, you need to ensure that the CLI command
members logout is configured. The Members Logout functionality handles login back from the phones
using FAC. It also supports call delivery to Voice Hunt Group from B-ACD.
For a sample configuration, see Example for Configuring B-ACD with Unified E-SRST, page 102.
Recommendations for Configuring Voice Hunt Group on Unified E-SRST

The Unified E-SRST Release with Support for voice hunt group has the following design characteristics:
• For all the directory numbers falling back from Unified Communications Manager, a common voice
register pool configuration as well as an individual voice register pool configuration is supported for
this deployment. An individual voice register pool configured with the CLI command id
device-id-name, along with voice register dn configuration, is recommended.
• Ensure that the CLI command mode esrst is configured under voice register global configuration
mode for phones to fall back to Unified E-SRST.
• Ensure that the CLI command id ip or id device-id-name is configured under voice register pool
configuration mode, along with voice register dn configuration, for a deployment with individual
voice register pool configuration. For a sample configuration, see Example for configuring Unified
E- SRST with Voice Hunt Group Enhancements, page 101.

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• Ensure that the CLI command id device-id-name is preferred over id ip as the CLI command to
configure under voice register pool configuration mode in scenarios where the IP address of the
phone might change due to the DHCP configured on the phone.
• Ensure that the CLI command id network is configured under voice register pool configuration
mode for a deployment with common voice register pool configuration. The recommended
configuration is id network 8.55.0.0 255.255.0.0 so as to facilitate registration of phones falling
back on Unified E-SRST from Unified Communications Manager.
• Ensure that the CLI command members logout is configured under voice hunt-group configuration
mode. The CLI is applied by default when the SIP phones fall back to Unified E-SRST from Unified
Communications Manager.
• Ensure that the CLI command fac standard is configured under telephony-service configuration
mode. If you want to configure a FAC code other than *5, you need to configure the CLI command
fac custom under telephony-service configuration mode.
• Ensure that the CLI commands call-park system application and hunt-group logout hlog are
configured under telephony-service configuration mode. The CLI commands are mandatory
configuration for FAC functionality to work.
For steps on configuring voice hunt groups on Unified E-SRST, see Configure Voice Hunt Groups on
Unified E-SRST, page 99.
For a sample configuration of voice hunt groups on Unified E-SRST, see Example for configuring
Unified E- SRST with Voice Hunt Group Enhancements, page 101.
Restrictions for Unified E-SRST, Release 12.2

The Unified E-SRST deployment with voice hunt group is known to have the following restrictions:
• Auto Logout is not supported
• Programmable Line Keys (PLK) are not supported
• HLog Softkey is not supported
SIP: Configure Unified E-SRST

The Enhanced SRST for Cisco Unified SIP IP Phones feature supports version negotiation between the
SIP phones and ESRST to enable more features in the Cisco Unified ESRST mode. In the current
scenario, when the SIP phones fall back to the SRST mode, features such as Shared-Line,
Busy-Lamp-Field (BLF), and Video call are disabled on the phones because the features are not
supported in the SRST mode. However, with the Enhanced Survivable Remote Site Telephony (E-SRST)
deployment, you can enable the basic and supplementary call features. Also, you can enable the
following features using version negotiation:
• Shared-Line
• Busy-Lamp-Field (BLF)
• Video Calls

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Table 3-1 contains a list of supported features and the expected behavior of the features in the E-SRST
mode.
Table 3-1 Supported features in the E-SRST mode
Expected Behavior in the

Feature Supported Features E-SRST Mode
Shared-Line cBarge Not Supported (After the
failover, the phone does not
retain the key.)
Privacy-on-hold Supported
Transfer Supported
Conference Supported
BLF BLF dn monitoring Supported
BLF device-based Not supported (Not supported in
monitoring RT phones)
BLF call-list Supported
monitoring
Monitoring of Call-park Not supported
slot
Monitoring of Paging Not supported
dn
Monitoring of Not supported
Conference dn
• To enable version negotiation feature between ESRST & phone, user needs to configure "mode
esrst" under voice register global mode.
• It is recommended to use SRST manager to automate the CLI provisioning of ESRST branch
routers.
For more information on SRST, see the Cisco Unified SRST Manager Administration Guide.
Restrictions
• The Version Negotiation feature is supported only on the Cisco Unified 9951, 9971, 8961 SIP IP
phones, Cisco IP Phone 7800 and 8800 Series.
• The phone firmware version should be Version 9.4.1 or later versions.
• This feature supports video calls only between the local Cisco Unified SIP IP phones and the No
Time-Division Multiplexing (TDM) video calls during the SRST failovers.
• To enable phone specific features like shared-line & BLF work, individual voice register pools need
to be configured.
Enable the E-SRST Mode

To enable the version negotiation feature in the Unified E-SRST mode, perform the following procedure.

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SUMMARY STEPS
1. enable
4. mode esrst
5. exit
DETAILED STEPS

Step 1 enable Enables the privileged EXEC mode. Enter your password if
prompted.
Example:
Router> enable
Step 2 configure terminal Enters the global configuration mode.
Example:
Step 3 voice register global Enters the voice register global configuration mode to set
the parameters for all the supported SIP phones in Cisco
Unified CME.
Example:
Step 4 mode esrst Configures the E-SRST mode under the voice register
global mode.
Example:
Router(config-register-global)# mode esrst
Step 5 exit Exits the voice register-global configuration mode.
Example:
Configure SIP shared-line

To configure SIP shared-line, perform the following procedure:
1. enable
3. voice register dn dn-tag
4. number number
5. shared-line [max-calls number-of-calls]
6. huntstop channel number-of-channels
7. end

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Configure BLF
To configure BLF, perform the following procedure:
1. enable
3. sip-ua
4. presence enable
5. exit
6. presence
7. max-subscription number
8. presence call-list
9. end
Enable a SIP Directory Number to be Watched

To enable a directory number to be watched, perform the following procedure:
2. number number
3. allow watch
4. end
Enable BLF on a Voice Register Pool

To enable BLF on a voice register pool, perform the following steps:
1. enable
4. number tag dn dn-tag
5. blf-speed-dial tag number label string [device]
6. presence call-list (To enable Presence feature for all the missed/received/placed calls)
7. end
For configuration information, see the Cisco Unified Communications Manager Administration Guide.
Example: ESRST mode
The following example shows how to enable the E-SRST mode:


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Example: Configuring Shared Line
The following example shows how to configure shared-line:

Router(config)#voice register dn 1
Router (config-register-dn)#number 1111
Router (config-register-dn)#shared-line max-calls 7
Router(config)#voice register pool 1

Router(config-register-pool)#Id mac 002D.264E.54FA
Router(config-register-pool)#type 9971
Router(config-register-pool)#number 1 dn 1

Router(config-register-pool)#id mac 000D.39F9.3A58
Example: Configuring BLF
The following example shows how to configure BLF:

Router (config-register-dn)#allow watch

Router(config-register-pool)#id mac 0015.6247.EF90

Router(config-register-pool)#id mac 0012.0007.8D82
Router(config-register-pool)#blf-speed-dial 1 1111 label "1111"
Note If the phone and the Unified E-SRST router are in different subnets and you are using id mac in the voice
register pool configuration mode, then the user must configure digest credentials on Unified
Communications Manager, and username password configuration under voice register pool on Unified
E-SRST. Digest Configuration is not required with the id device-id-name CLI command introduced in
Unified SRST Release 12.2.
Configure Unified E-SRST

The mode esrst under telephony-service and voice register global configuration mode provisions
SCCP and SIP phones respectively to enable the enhanced services in Unified E-SRST mode. While
Unified SRST supports only the basic voice hunt group features, Unified E-SRST supports advanced
voice hunt group features such as HLog, shared lines, and B-ACD. To configure the basic Unified
E-SRST, perform the following procedure:
SUMMARY STEPS
1. enable

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3. telephony-service
4. mode esrst
5. max-ephones max-phones
7. ip source-address ip-address [port port] [any-match | strict-match]
8. call-park system application
9. hunt-group logout HLog
10. transfer-system full-consult
11. transfer-pattern transfer-pattern
12. fac standard
13. create cnf-files
14. exit
16. mode esrst
18. max-pool max-phones
19. exit
21. number number
22. exit
24. id [network address mask mask | ip address mask mask][device-id-name devicename]
25. dtmf-relay rtp-nte
26. exit

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DETAILED STEPS

prompted.
Example:
Router> enable
Example:
Step 3 telephony-service Enters telephony-service configuration mode.
Example:
Step 4 mode esrst Configures the E-SRST mode under the telephony-service
configuration mode.
Example:
Step 5 max-ephones max-phones Configures the maximum number of IP phones that can be
supported by the router. The default is 0.
The maximum number is platform dependent.
Example:
Router(config-telephony)# max-ephones 40
Step 6 max-dn max-directory-numbers Sets the maximum number of directory numbers (DNs) that
can be supported by the router.
• max-directory-numbers: Maximum number of
directory numbers (dns) or virtual voice ports
supported by the router. The maximum number is
Example: platform-dependent. The default is 0.
Router(config-telephony)# max-dn 15
Step 7 ip source-address ip-address [port port] Enables the router to receive messages from the Cisco IP
[any-match | strict-match] phones through the specified IP addresses and supports
strict IP address verification. The default port number is
Example: 2000.
Router(config-telephony)# ip source-address
8.39.23.24 port 2000
Step 8 call-park system {application |redirect} Defines system parameters for the Call Park feature.
• application — Enables the Call Park features supported
Example: in Unified SRST.
Router(config-telephony)# call-park system
application

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Step 9 hunt-group logout {DND | HLog} Sets the hunt-group logout options with Hlog in
telephony-service configuration mode.
Example:
Router(config-telephony)# hunt-group logout
HLog
Step 10 transfer-system full-consult Specifies the call transfer method.
• full-consult—Calls are transferred with consultation
Example: using H.450.2 standard methods and a second phone
Router(config-telephony)# transfer-system line if available. Calls fall back to full-blind if the
full-consult second line is unavailable.
Step 11 transfer-pattern transfer-pattern Allows transfer of telephone calls by Cisco Unified IP
phones to specified phone number patterns. If no transfer
pattern is set, the default is that transfers are permitted only
Example:
Router(config-telephony)# transfer-pattern .T
to other local IP phones.
• transfer-pattern—A string of digits for permitted call
transfers.
Step 12 fac { standard | custom { alias alias-tag | Enables all standard feature access codes (FACs) or creates
feature } } and enables individual custom FACs in telephony-service
configuration mode.
Example:
Router(config-telephony)# fac standard
Step 13 create cnf-files Builds the eXtensible Markup Language (XML)
configuration files that are required for IP phones, in
telephony-service configuration mode.
Example:
Router(config-telephony)# create cnf-files
version-stamp
Step 14 exit Exits the telephony-service configuration mode.
Example:
Step 15 voice register global Enters the voice register global configuration mode.
Example:
Step 16 mode esrst Configures the E-SRST mode under the voice register
global mode.
Example:
Step 17 max-dn max-directory-numbers Sets the maximum number of SIP phone directory numbers
(extensions) that are supported by a Cisco router in voice
register global configuration mode.
Example:

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Step 18 max-pool max-phones Sets maximum number of SIP phones to be supported by the
Example: • Version- and platform-dependent; type ? for range.
Step 19 exit Exits the voice register global configuration mode.
Example:
Step 20 voice register dn dn-tag Enters voice register dn configuration mode to define a
directory number for a SIP phone.
Example: The voice register dn configured in Unified E-SRST must
Router(config)# voice register dn 17 be the same directory number (dn) configured in Unified
Communications Manager.
Step 21 number number Defines a valid number for a directory number.
Example:
Router(config-register-dn)# number 7001
Step 22 exit Exits the voice register dn configuration mode.
Example:
Router(config-register-dn)# exit
Step 23 voice register pool pool-tag Enters the voice register pool configuration mode to set
Example:
Step 24 id [{network address mask mask | ip address Explicitly identifies a locally available individual or set of
mask mask | mac address}] [device-id-name SIP IP phones. The keywords and arguments are defined as
devicename]
follows:
• network address mask mask: The network address
Example: mask mask keyword/argument combination is used to
8.55.0.0 mask 255.255.0.0
subnet.
individual phone.
• device-id-name devicename: Defines the device name
to be used to download the phone’s configuration file.

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Protocol (RTP) with the Named Telephone Event (NTE)
payload type and enables DTMF relay using the RFC 2833
Example: standard method.
rtp-nte
Step 26 exit Exits the voice register pool configuration mode.
Example:
Router(config-register-pool)# exit
Configure Voice Hunt Groups on Unified E-SRST

To configure Voice Hunt Group feature on Unified E-SRST, perform the following procedure:
SUMMARY STEPS
1. enable
3. voice hunt-group hunt-tag {longest-idle | parallel | peer | sequential}
4. members logout (optional)
5. list number [, number...]
6. timeout seconds
7. statistics collect
8. pilot 111

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DETAILED STEPS

prompted.
Example:
Router> enable
Example:
Step 3 voice hunt-group hunt-tag {longest-idle | Enters voice hunt-group configuration mode to define a
parallel | peer | sequential} hunt group.
• hunt-tag—Unique sequence number of the hunt group
to be configured. Range is 1 to 100.
• longest idle—Hunt group in which calls go to the
directory number that has been idle for the longest time.
• sequential—Hunt group in which directory numbers
ring in the order in which they are listed, left to right.
• parallel—Hunt group in which all directory numbers
ring simultaneously.
• peer—Hunt group in which the call placed to a
directory number rings for the next directory number in
line.
To change the hunt-group type, remove the existing hunt
Example: group first by using the no form of the command; then,
Router(config)# voice hunt-group 1 sequential recreate the group.
Step 4 members logout (optional) Configures a Unified SRST system for all
non-shared static members or agents in a voice hunt group
with the Hlogout initial state.
Example:
Router(config-voice-hunt-group)# members logout
Step 5 list number [, number...] Defines a list of extensions that are members of a voice hunt
group.
Example:
Router(config-voice-hunt-group)# list 1812,
1813, 1814
Step 6 timeout seconds Defines the number of seconds after which a call that is not
answered is redirected to the next number in a voice
hunt-group list.
Example:

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Step 7 statistics collect Enables the collection of call statistics for a voice hunt
group.
Example:
Router(config-voice-hunt-group)# statistics
collect
Step 8 exit Exits the voice hunt group configuration mode.
Example:
Example for configuring Unified E- SRST with Voice Hunt Group Enhancements
The following is a sample configuration for Unified E-SRST Release 12.2 under telephony-service,
voice register global, voice register pool, and voice hunt-group configuration modes, for a deployment
with common voice register pool configuration.
Router#
telephony-service
call-park system application
hunt-group logout HLog
transfer-system full-consult
fac standard
Router#sh run | sec global

mode esrst
max-dn 40
max-pool 40
Router#
id network 8.55.0.0 mask 255.255.0.0
dtmf-relay rtp-nte
Router#
telephony-service
max-ephones 40
max-dn 50
ip source-address 8.39.23.24 port 2000
transfer-pattern .T
fac standard
create cnf-files version-stamp Jan 01 2002 00:00:00
Router#sh run | sec hunt

voice hunt-group 1 sequential
members logout
list 1812,1813,1814
timeout 30
statistics collect
pilot 1111
The following is a sample configuration for Unified E-SRST Release 12.2, for a deployment with
individual voice register pool configuration, with the CLI command id ip configured.

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voice register dn 2
number 4000
!
voice register dn 3
number 4002
!
busy-trigger-per-button 2
id ip 8.55.0.241 mask 255.255.0.0
type 8811
number 1 dn 2
dtmf-relay rtp-nte
codec g711ulaw
!
id ip 8.55.0.242 mask 255.255.0.0
type 7861
number 1 dn 3
dtmf-relay rtp-nte
codec g711ulaw
The following is a sample configuration for Unified E-SRST Release 12.2, for a deployment with
individual voice register pool configuration, with the CLI command id device-id-name configured.
voice register dn 2
number 4000
!
voice register dn 3
number 4002
!
id device-id-name SEP00EBD5CD77ED
type 8811
number 1 dn 2
dtmf-relay rtp-nte
codec g711u;aw

id device-id-name SEP0076861A7EDC
type 7861
number 1 dn 3
dtmf-relay rtp-nte
codec g71ulaw
Example for Configuring B-ACD with Unified E-SRST

The following is a sample configuration for B-ACD functionality supported with Unified E-SRST:
application
service aa-bcd bootflash:/app-b-acd-aa-3.0.0.4_thd_v4.tcl
paramspace english index 0
param second-greeting-time 60
param welcome-prompt _bacd_welcome.au
param call-retry-timer 8
param voice-mail 1811
paramspace english language en
param max-time-call-retry 16

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param service-name callq

param number-of-hunt-grps 2
param handoff-string aa-bcd
paramspace english location flash:
param max-time-vm-retry 2
param aa-pilot 1117
!
service clid_col_npw_npw
param uid-length 4
!
service aa-ccd bootflash:/app-b-acd-aa-3.0.0.4_thd_v4.tcl
paramspace english index 0
param drop-through-prompt _bacd_welcome.au
param second-greeting-time 60
paramspace english language en
param call-retry-timer 8
param voice-mail 1811
param max-time-call-retry 16
param service-name callq
param drop-through-option 1
paramspace english location flash:
param handoff-string aa-ccd
param max-time-vm-retry 2
param aa-pilot 1118
!
service callq bootflash:/imanage-b-acd-3.0.0.4_Q60.tcl
param queue-len 1
param aa-hunt1 1111
param queue-manager-debugs 1
!
Example for Configuring Shared Line with Voice Hunt Group on Unified E-SRST
The following is a sample configuration of Unified E-SRST, Release 12.2 with support for mixed shared
lines (SIP and SCCP Phones) in a voice hunt group deployment.
Router# sh run | sec global
mode esrst
no allow-hash-in-dn
max-dn 40
max-pool 40
Router# sh run | sec pool

max-pool 40
id device-id-name SEP00CCFC4AA4DC
type 8811
number 1 dn 1
number 2 dn 21
dtmf-relay rtp-nte
username xxxx password uvwx
codec g711ulaw
no vad
id device-id-name SEP00CCFC177A4E
type 8841

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number 1 dn 2
dtmf-relay rtp-nte
codec g711ulaw
no vad
id device-id-name SEP0076861ADEF0
type 7841
number 1 dn 3
number 2 dn 22
dtmf-relay rtp-nte
codec g711ulaw
no vad
id device-id-name SEP00EBD5CD270C
type 8811
number 1 dn 4
number 2 dn 22
dtmf-relay rtp-nte
codec g711ulaw
no vad
id device-id-name SEP94D4692A2553
type 8841
number 1 dn 5
dtmf-relay rtp-nte
codec g711ulaw
no vad
id device-id-name SEP00CAE540C4B5
type 8811
number 1 dn 6
number 2 dn 21
dtmf-relay rtp-nte
codec g711ulaw
no vad
alias exec pool show voice register pool all br
Router# sh run | sec dn

no allow-hash-in-dn
max-dn 40
voice register dn 1
voice-hunt-groups login
number 1811
voice register dn 2
number 1812
voice register dn 3
number 1813
voice register dn 4
number 1814
voice register dn 5

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number 1815
voice register dn 6
number 1816
voice register dn 21
number 1821
shared-line
voice register dn 22
number 1822
shared-line
Router# sh run | sec ephone

max-ephones 40
ephone-dn 11
number 1911
ephone-dn 12
number 1912
ephone-dn 13
number 1913
ephone-dn 14
number 1914
ephone-dn 21
number 1921
ephone-dn 22
number 1822
shared-line sip
ephone 11
device-security-mode none
mac-address 1111.1111.1911
feature-button 1 HLog
type 7970
button 1:11
ephone 12
mac-address 1111.1111.1912
type 7970
button 1:12 2:21
ephone 13
mac-address 1111.1111.1913
type 7970
button 1:13 2:21
ephone 14
mac-address 1111.1111.1914
type 7970
button 1:14 2:22
alias ephone show ephone summary brief
alias exec ephone show ephone summary brief
Router# sh run | sec tele

telephony-service
conference transfer-pattern
mode esrst
max-ephones 40
max-dn 50
service phone sshAccess 0

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service phone webAccess 0

max-conferences 8 gain -6
hunt-group logout HLog
fac standard
SCCP: Configure Unified E-SRST

You need to configure mode esrst under telephony-service to enable ESRST mode for SCCP Phones.
Prerequisites
• Cisco Unified CME 10.5 or later version
• The telephony-services command must be configured
Note For SCCP phones, CME-as-SRST mode is provisioned using the srst mode auto-provision command.
From 10.5 release onwards, this command will be deprecated. When you try to configure CME-as-SRST
mode, the following message will be displayed:
“Note: This configuration is being deprecated. Please configure "mode esrst" to use the enhanced SRST
mode”.
SUMMARY STEPS
1. enable
3. telephony-service
4. mode esrst
6. max-dn max-directory-numbers [preference preference-order] [no-reg primary | both]
7. ip source-address ip-address port port [any-match | strict-match]
8. exit
9. ephone-dn dn-tag [dual-line]
10. number number [secondary number] [no-reg [both | primary]]
11. name name
12. exit
13. ephone phone-tag
14. mac-address [mac-address]
15. type phone-type [addon 1 module-type [2 module-type]]
16. button button-number{separator}dn-tag [,dn-tag...] [button-number{x}overlay-button-number]
[button-number...]

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17. end

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DETAILED STEPS

prompted.
Example:
Router> enable
Example:
Step 3 telephony-service Enters telephony-service configuration mode.
Example:
Step 4 mode esrst Configures the E-SRST mode under the telephony-service
mode.
Example:
Router(config-telephony)# mode esrst
Step 5 max-ephones max-phones Sets the maximum number of phones that can register to
Unified E-SRST.
Example: • Maximum number is platform and version-specific. Type
Router(config-telephony)# max-ephones 24 ? for range.
Step 6 max-dn max-directory-numbers [preference Limits number of directory numbers to be supported by this
preference-order] [no-reg primary | both] router.
• Maximum number is platform and version-specific. Type
Example: ? for value.
Router(config-telephony)# max-dn 24 no-reg
primary
Step 7 ip source-address ip-address [port port] Identifies the IP address and port number that the Unified
[any-match | strict-match] SRST router uses for IP phone registration.
• port port—(Optional) TCP/IP port number to use for
Example: SCCP. Range is 2000 to 9999. Default is 2000.
Router(config-telephony)# ip source-address
192.168.11.1 port 2000 • any-match—(Optional) Disables strict IP address
checking for registration. This is the default.
• strict-match—(Optional) ) Instructs the router to reject
IP phone registration attempts if the IP server address used
by the phone does not exactly match the source address.
Step 8 exit Exits telephony-service configuration mode.
Example:

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Step 9 ephone-dn dn-tag [dual-line] Enters ephone dn configuration mode to define a directory
number for an IP phone, intercom line, voice port, or a mes-
sage-waiting indicator (MWI).
Example: • dn-tag—Identifies a particular directory number during
Router(config)# ephone-dn 1
configuration tasks. Range is 1 to the maximum number of
directory numbers allowed on the router platform. Type ?
to display the range.
Step 10 number number [secondary number] [no-reg [both Associates an extension number with this directory number.
|primary]]
• number—String of up to 16 digits that represents an
extension or E.164 telephone number.
Example:
Router(config-ephone-dn)# number 1001
Step 11 name name (Optional) Associates a name with this directory number.
• Name is used for caller-ID displays and in the local
Example: directory listings.
Router(config-ephone-dn)# name Smith, John
• Must follow the name order that is specified with the
directory command.
Step 12 exit Exits ephone-dn configuration mode.
Example:
Step 13 ephone phone-tag Enters ephone configuration mode to set ephone specific
parameters.
Example: • phone-tag—Unique sequence number that identifies the
Router(config)# ephone 1 phone. Range is version and platform-dependent; type ? to
display range.
Step 14 mac-address [mac-address] Associates the MAC address of a Cisco IP phone with an
ephone configuration in a Unified E-SRST system.
Example: • mac-address—Identifying MAC address of an IP phone,
Router(config-ephone)# mac-address which is found on a sticker located on the bottom of the
0022.555e.00f1 phone.
Step 15 type phone-type [addon 1 module-type [2 Specifies the type of phone.
module-type]]
Example:
Router(config-ephone)# type 7960

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Step 16 button button-number{separator}dn-tag Associates a button number and line characteristics with an
[,dn-tag...][button-number{x}overlay-button-num ephone-dn. Maximum number of buttons is determined by
ber] [button-number...]
phone type.
Example:
Router(config-ephone)# button 1:7
Example:
Example: Enhanced SRST mode configuration
The following example shows the status of the device in E-SRST mode:
show telephony-service
CONFIG (Version=10.5)
=====================
Version 10.5
Max phoneload sccp version 17
Max dspfarm sccp version 18
Cisco Unified Enhanced SRST
Note For SCCP phones, switching the mode from CME to ESRST and vice versa, results in wiping out the
entire CME or ESRST configurations (including ephone, DNs, templates etc.).
Configure Mixed Shared Lines with SCCP Phones

To configure mixed shared lines between SCCP and SIP IP Phones on Unified E-SRST, perform the
following procedure:
SUMMARY STEPS
1. enable
3. ephone-dn dn-tag [dual-line]
4. number number [secondary number] [no-reg [both | primary]]
5. shared-line sip
6. end

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DETAILED STEPS
prompted.
Example:
Router> enable
Example:
Step 3 ephone-dn dn-tag [dual-line] Enters ephone dn configuration mode to define a directory
number for an IP phone, intercom line, voice port, or a mes-
sage-waiting indicator (MWI).
• dn-tag—Identifies a particular directory number during
configuration tasks. Range is 1 to the maximum number of
Example: directory numbers allowed on the router platform. Type ?
Router(config)# ephone-dn 1 to display the range.
Step 4 number number [secondary number] [no-reg [both Associates an extension number with this directory number.
|primary]]
• number—String of up to 16 digits that represents an
extension or E.164 telephone number.
Example:
Router(config-ephone-dn)# number 1001
Step 5 shared-line sip Adds an ephone-dn as a member of a shared directory
number for a mixed shared line between Unified SIP and
Unified SCCP IP phones.
Example:
Router(config-ephone-dn)# shared-line sip
Example:
Router(config-ephone-dn)# end
Configure BLF for SCCP Phones

1. enable
3. presence
4. max-subscription number
6. end
Enable an SCCP Directory Number to be Watched

To enable a directory number to be watched, perform the following procedure:
1. ephone-dn dn-tag

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2. number number
3. allow watch
4. end
Enable BLF on an Ephone

To enable BLF on an ephone, perform the following steps:
1. enable
3. ephone ephone-tag
4. button button-number{separator}dn-tag [,dn-tag...] [button-number{x}overlay-button-number]
[button-number...]
5. blf-speed-dial tag number label string [device]
7. end
Configure Digest Credentials On Unified Communications Manager

To configure the username and password with Digest Authentication on Unified Communications
Manager, perform the following steps:
Step 1 Login to Cisco Unified Communications Manager.

Step 2 Go to System>Security->Phone Security Profile
a. Edit the existing configuration, or create a new configuration and associate with the phone
b. Check the Enable Digest Authentication box
Step 3 Go to User Management > End User
a. Create a new user
b. Add the User ID, and digest credentials
Step 4 Go to the Phone Settings page and associate the user in the Digest User field.
Configure Digest Credentials on Unified E-SRST for SIP

To configure credentials under a specific voice register pool, perform the following procedure:
SUMMARY STEPS
1. enable
3. voice register pool <pool-tag>
4. username <username> password <password>
5. end

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Example: Configuring Digest Credentials on ESRST
The following example shows how to configure digest credentials on ESRST:

Router# conf terminal
Router (config-register-pool)# username abc password xyz
Configure Digest Credentials on Unified E-SRST for SCCP

To configure credentials under a specific ephone, perform the following procedure:
SUMMARY STEPS
1. enable
3. ephone ephone-tag
4. username username password password
5. end
Unified E-SRST Scale Support

For Unified E-SRST 10.5 to 12.0, the scale of Unified E-SRST mode is increased to match the scale of
Classic SRST for both SIP and SCCP Phones.
Table 3-2 lists the scale for number of phones and DNs supported in ESRST mode for Release 10.5 to
12.0.
Table 3-2 ESRST Scales Supported on All Platforms

Platform New Phone Scale New DN Scale
Cisco 2901 35 200
Cisco 2911 50 300
Cisco 2921 100 400
Cisco 2951 250 500
Cisco 3925 730 1000
Cisco 3945 1200 1800
Cisco 3925E 1350 2000
Cisco 3945E 1500 2500
Cisco 4451-X 1500 2500
ISR 4321 35 200
ISR 4331 100 400
ISR 4351 730 1000
ISR 4431 1200 1800
ISR 4451 1500 2500

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Note The increase in scale mentioned in the table is only for basic calls. For enhanced feature support such as
Shared-line, BLF, Video, and VHG, these numbers are not applicable.
Example: ESRST Scale Increase
The following example shows the increase in scale support in the E-SRST mode for ISR 3945E platform:
ESRST_3945e(config-telephony)#max-dn ?
<1-2500> Maximum single/dual/octo line directory numbers supported
ESRST_3945e(config-telephony)#max-ephones ?
<1-1500> Maximum phones to support
Where to Go Next
Proceed to the “Setting Up the Network” section on page 115.

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CHAPTER 4
Setting Up the Network
This chapter describes how to configure your Cisco Unified Survivable Remote Site Telephony (SRST)
router to run DHCP and to communicate with the IP phones during Cisco Unified Communications
Manager fallback.
Contents
• Information About Setting Up the Network, page 116
• How to Set Up the Network, page 116

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Chapter 4 Setting Up the Network
Information About Setting Up the Network
Information About Setting Up the Network

When the WAN link fails, the Cisco Unified IP Phones detect that they are no longer receiving keepalive
packets from Cisco Unified CM. The Cisco Unified IP Phones then register with the router. The Cisco
Unified SRST software is automatically activated and builds a local database of all Cisco Unified IP
Phones attached to it (up to its configured maximum). The IP phones are configured to query the router
as a backup call-processing source when the central Cisco Unified CM does not acknowledge keepalive
packets. The Cisco Unified SRST router now performs call setup and processing, call maintenance, and
call termination.
Cisco Unified Communications Manager uses DHCP to provide Cisco Unified IP Phones with the IP
address of Cisco Unified Communications Manager. In a remote branch office, DHCP service is
typically provided either by the SRST router itself or through the Cisco Unified SRST router using
DHCP relay. Configuring DHCP is one of two main tasks in setting up network communication. The
other task is configuring the Cisco Unified SRST router to receive messages from the Cisco IP phones
through the specified IP addresses. Keepalive intervals are also set at this time.
How to Set Up the Network

This section contains the following tasks:
• Enabling IP Routing, page 116 (Required)
• Enabling Cisco Unified SRST on an MGCP Gateway (Required)
• Configuring DHCP for Cisco Unified SRST Phones, page 122 (Required)
• Specifying Keepalive Intervals, page 125 (Optional)
Enabling IP Routing
To initiate SRST service, you need to enable IP routing command and configure an interface that you
want to use or bind. For information about enabling IP routing, see Configuring IP Addressing.
Enabling Cisco Unified SRST on an MGCP Gateway

To use SRST as your fallback mode with an MGCP gateway, SRST and MGCP fallback must both be
configured on the same gateway. The configuration below allows SRST to assume control over the voice
port and over call processing on the MGCP gateway. Due to command changes that were made in
Cisco IOS Release 12.3(14)T, use the configuration task that corresponds with the Cisco IOS Release
you have installed.
Note The commands described in the configuration below are ineffective unless both commands are
configured. For instance, your configuration will not work if you only configure the ccm-manager
fallback-mgcp command.

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Note When an MGCP-controlled PRI goes into SRST mode, do not make or save configuration changes to the
NVRAM on the router. If configuration changes are made and saved in SRST mode, the
MGCP-controlled PRI fails when normal MGCP operation is restored.
Configuring Cisco Unified SRST on an MGCP Gateway Prior to Cisco IOS Release 12.3(14)T
Perform this task to enable SRST on a MGCP Gateway if you are using a software release prior to
Cisco IOS Release 12.3(14)T.
SUMMARY STEPS
1. enable
3. ccm-manager fallback-mgcp
4. call application alternate [application-name]
or
service [alternate | default] service-name location
5. exit
DETAILED STEPS

• Enter your password when prompted.
Example:
Router> enable
Example:
Step 3 ccm-manager fallback-mgcp Enables the gateway fallback feature and allows an MGCP
voice gateway to provide call processing services through
SRST or other configured applications when
Example:
Router(config)# ccm-manager fallback-mgcp
Cisco Unified Communications Manager is unavailable.

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Step 4 call application alternate [application-name] The call application alternate command specifies that
or the default voice application takes over if the MGCP
service [alternate | default] service-name
location
application is not available. The application-name
argument is optional and indicates the name of the specific
voice application to use if the application in the dial peer
Example: fails. If a specific application name is not entered, the
Router(config)# call application alternate gateway uses the DEFAULT application.
or
Router(config)# service default Or
The service command loads and configures a specific,
standalone application on a dial peer. The keywords and
arguments are as follows:
• alternate (Optional). Alternate service to use if the
service that is configured on the dial peer fails.
• default (Optional). Specifies that the default service
(“DEFAULT”) on the dial peer is used if the alternate
service fails.
• service-name: Name that identifies the voice
application.
• location: Directory and filename of the Tcl script or
VoiceXML document in URL format. For example,
flash memory (flash:filename), a TFTP
(tftp://../filename), or an HTTP server
(http://../filename) are valid locations.
Step 5 exit Exits global configuration mode and returns to privileged
EXEC mode.
Example:
Router(config)# exit
Configuring SRST on an MGCP Gateway Using Cisco IOS Release 12.3(14)T or Later
Releases
Perform this task to enable SRST on an MGCP Gateway if you are using Cisco IOS Release 12.3(14)T
or later version.
Restrictions
Effective with Cisco IOS Release 12.3(14)T, the call application alternate command is replaced by the
service command. The service command can be used in all releases after Cisco IOS Release 12.3(14)T.

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SUMMARY STEPS
1. enable
3. ccm-manager fallback-mgcp
4. application [application-name]
5. global
6. service [alternate | default] service-name location
7. exit
DETAILED STEPS

• Enter your password when prompted.
Example:
Router> enable
Example:
Step 3 ccm-manager fallback-mgcp Enables the gateway fallback feature and allows an MGCP
voice gateway to provide call processing services through
SRST or other configured applications when
Example:
Router(config)# ccm-manager fallback-mgcp
Cisco Unified Communications Manager is unavailable.
Step 4 application [application-name] The application-name argument is optional and indicates
the name of the specific voice application to use if the
application in the dial peer fails. If a specific application
Example:
Router(config) application app-xfer
name is not entered, the gateway uses the DEFAULT
application.
Step 5 global Enters global configuration mode.
Example:
Router(config)# global

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Step 6 service [alternate | default] service-name Loads and configures a specific, standalone application on
location a dial peer.
• alternate (Optional). Alternate service to use if the
Example: service that is configured on the dial peer fails.
Router(config) service myapp
https://myserver/myfile.vxml • default (Optional). Specifies that the default service
(“DEFAULT”) on the dial peer is used if the alternate
service fails.
• service-name: Name that identifies the voice
application.
• location: Directory and filename of the Tcl script or
VoiceXML document in URL format. For example,
flash memory (flash:filename), a TFTP
Step 7 exit Exits global configuration mode and returns to privileged
EXEC mode.
Example:
Router(config)# exit
Configuration Example of Enabling SRST on a MGCP Gateway using Cisco IOS Release
12.3(14)T
The following is an example of configuring SRST on an MGCP Gateway if you are using Cisco IOS
Release 12.3(14)T or later release:
isdn switch-type primary-net5
!
!
ccm-manager fallback-mgcp
ccm-manager mgcp
ccm-manager config
mta receive maximum-recipients 0
!
controller E1 1/0
pri-group timeslots 1-12,16 service mgcp
!
controller E1 1/1
!
!
!
interface Ethernet0/0
ip address 10.48.80.9 255.255.255.0
half-duplex
!
interface Serial1/0:15
no ip address
no logging event link-status
isdn incoming-voice voice
isdn bind-l3 ccm-manager
no cdp enable
!

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!
!
call rsvp-sync
!
call application alternate DEFAULT
!--- For Cisco IOS® Software Release 12.3(14)T or later,

this command was replaced by the service command
in global application configuration mode.
application
global
service alternate Default
!
voice-port 1/0:15
!
mgcp
mgcp dtmf-relay voip codec all mode cisco
mgcp package-capability rtp-package
mgcp sdp simple
!
mgcp profile default
!
!
!
dial-peer cor custom
!
!
!
dial-peer voice 10 pots
application mgcpapp
incoming called-number
destination-pattern 9T
direct-inward-dial
port 1/0:15
!
!
limit-dn 7960 2
max-ephones 10
max-dn 32
dialplan-pattern 1 704.... extension-length 4
keepalive 20
default-destination 5002
alias 1 5003 to 5002
call-forward busy 5002
call-forward noan 5002 timeout 12
time-format 24
!
!
line con 0
exec-timeout 0 0
line aux

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Configuring DHCP for Cisco Unified SRST Phones

To perform this task, you must have your network configured with DHCP. For further details about
DHCP configuration, see the Cisco IOS DHCP Server document and see your Cisco Unified
Communications Manager documentation.
When a Cisco IP phone is connected to the Cisco Unified SRST system, it automatically queries for a
DHCP server. The DHCP server responds by assigning an IP address to the Cisco IP phone and
providing the IP address of the TFTP server through DHCP option 150. Then, the phone registers with
the Cisco Unified Communications Manager system server and attempts to get configuration and phone
firmware files from the Cisco Unified Communications Manager TFTP server address provided by the
DHCP server.
When setting up your network, configure your DHCP server local to your site. You may use your SRST
router to provide DHCP service (recommended). If your DHCP server is across the WAN and there is an
extended WAN outage, the DHCP lease times on your Cisco Unified IP Phones may expire. This may
cause your phones to lose their IP addresses, resulting in a loss of service. Rebooting your phones when
there is no DHCP server available after the DHCP lease has expired will not reactivate the phones,
because they will be unable to obtain an IP address or other configuration information. Having your
DHCP server local to your remote site ensures that the phones can continue to renew their IP address
leases in the event of an extended WAN failure.
Choose one of the following tasks to set up DHCP service for your Cisco UnifiedIP Phones:
• Defining a Single DHCP IP Address Pool, page 122:Use this method if the Cisco Unified SRST
router is a DHCP server and if you can use a single shared address pool for all your DHCP clients.
• Defining a Separate DHCP IP Address Pool for Each Cisco Unified IP Phone, page 123: Use this
method if the Cisco Unified SRST router is a DHCP server and you need separate pools for
non-IP-phone DHCP clients.
• Defining the DHCP Relay Server, page 124: Use this method if the Cisco Unified SRST router is
not a DHCP server and you want to relay DHCP requests from IP phones to a DHCP server on a
different router.
Defining a Single DHCP IP Address Pool

This task creates a large shared pool of IP addresses in which all DHCP clients receive the same
information, including the option 150 TFTP server IP address. The benefit of selecting this method is
that you set up only one DHCP pool. However, defining a single DHCP IP address pool can be a problem
if non-IP phone clients need to use a different TFTP server address.
SUMMARY STEPS
1. ip dhcp pool pool-name

2. network ip-address [mask | prefix-length]
3. option 150 ip ip-address
4. default-router ip-address
5. exit

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DETAILED STEPS

Step 1 ip dhcp pool pool-name Creates a name for the DHCP server address pool
and enters DHCP pool configuration mode.
Example:
Router(config)# ip dhcp pool mypool
Step 2 network ip-address [mask | prefix-length] Specifies the IP address of the DHCP address pool
and the optional mask or number of bits in the
address prefix, preceded by a forward slash.
Example:
Router(config-dhcp)# network 10.0.0.0 255.255.0.0
Step 3 option 150 ip ip-address Specifies the TFTP server address from which the
Cisco IP phone downloads the image configuration
file. This needs to be the IP address of Cisco Unified
Example:
Router(config-dhcp)# option 150 ip 10.0.22.1
CM.
Step 4 default-router ip-address Specifies the router to which the
Cisco Unified IP phones are connected directly.
Example: • This router should be the Cisco Unified SRST
Router(config-dhcp)# default-router 10.0.0.1 router because this is the default address that is
used to obtain SRST service in the event of a
WAN outage. As long as the Cisco IP phones
have a connection to the Cisco Unified SRST
router, the phones are able to get the required
network details.
Step 5 exit Exits DHCP pool configuration mode.
Example:
Router(config-dhcp)# exit
Defining a Separate DHCP IP Address Pool for Each Cisco Unified IP Phone
This task creates a name for the DHCP server address pool and specifies IP addresses. This method
requires you to make an entry for every Cisco Unified IP phone.
SUMMARY STEPS
1. ip dhcp pool pool-name

2. host ip-address subnet-mask
3. option 150 ip ip-address
4. default-router ip-address
5. exit

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DETAILED STEPS

Step 1 ip dhcp pool pool-name Creates a name for the DHCP server address pool
and enters DHCP pool configuration mode.
Example:
Router(config)# ip dhcp pool pool2
Step 2 host ip-address subnet-mask Specifies the IP address that you want the phone to
use.
Example:
Router(config-dhcp)# host 10.0.0.0 255.255.0.0
Step 3 option 150 ip ip-address Specifies the TFTP server address from which the
Cisco IP phone downloads the image
configuration file. This needs to be the IP address
Example:
Router(config-dhcp)# option 150 ip 10.0.22.1
of Cisco Unified CM.
Step 4 default-router ip-address Specifies the router to which the Cisco
Unified IP phones are connected directly.
Example: • This router should be the Cisco Unified SRST
Router(config-dhcp)# default-router 10.0.0.1 router because this is the default address that
is used to obtain SRST service in the event of
a WAN outage. As long as the
Cisco IP phones have a connection to the
Cisco Unified SRST router, the phones are
able to get the required network details.
Step 5 exit Exits DHCP pool configuration mode.
Example:
Router(config-dhcp)# exit
Defining the DHCP Relay Server

This task sets up DHCP relay on the LAN interface where the Cisco Unified IP phones are connected
and enables the Cisco IOS DHCP server feature to relay requests from DHCP clients (phones) to a DHCP
server. For further details about DHCP configuration, see the Cisco IOS DHCP Server document.
The Cisco IOS DHCP server feature is enabled on routers by default. If the DHCP server is not enabled
on your Cisco Unified SRST router, use the following steps to enable it.
SUMMARY STEPS
1. service dhcp
2. interface type number
3. ip helper-address ip-address
4. exit

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DETAILED STEPS

Step 1 service dhcp Enables the Cisco IOS DHCP Server feature on
the router.
Example:
Router(config)# service dhcp
Step 2 interface type number Enters interface configuration mode for the
specified interface. See Cisco IOS Interface and
Hardware Component Command Reference,
Example:
Router(config)# interface serial 0
Release 12.3T for more information.
Step 3 ip helper-address ip-address Specifies the helper address for any unrecognized
broadcast for TFTP server and Domain Name
System (DNS) requests. For each server, a
Example:
Router(config-if)# ip helper-address 10.0.22.1
separate ip helper-address command is required
if the servers are on different hosts. You can also
configure multiple TFTP server targets by using
the ip helper-address command for multiple
servers.
Step 4 exit Exits interface configuration mode.
Example:
Router(config-if)# exit
Specifying Keepalive Intervals

The keepalive interval is the period of time between keepalive messages sent by a network device. A
keepalive message is a message sent by one network device to inform another network device that the
virtual circuit between the two is still active.
Note If you plan to use the default time interval between messages, which is 30 seconds, you do not have to
perform this task.
SUMMARY STEPS
2. keepalive seconds
3. exit

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Where to Go Next
DETAILED STEPS

Example:
Step 2 keepalive seconds Sets the time interval, in seconds, between keepalive
messages that are sent to the router by Cisco Unified IP
Phones.
Example:
Router(config-cm-fallback)# keepalive 60 • seconds: Range is 10 to 65535. Default is 30.
Example:
Examples
The following example sets a keepalive interval of 45 seconds:
keepalive 45
Where to Go Next
The next step is setting up the phone and getting a dial tone. For instructions, see the
“Cisco Unified SIP SRST 4.1” section on page 127.
For additional information, see the “Additional References” section on page 30 in the “Cisco Unified
SRST Feature Overview” section on page 1 chapter.

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CHAPTER 5
Cisco Unified SIP SRST 4.1
This chapter describes the features and provides the configuration information for Cisco Unified
SIP SRST 4.1:
• Out-of-Dialog REFER(OOD-R)
• Digit Collection on SIP Phones
• Caller ID Display
• Disabling SIP Supplementary Services for Call Forward and Call Transfer
• Idle Prompt Status
Note With Cisco IOS Release 12.4(15)T, the number of SIP phones supported on each platform is now
equivalent to the number of SCCP phones supported. For example, 3845 now supports 720 phones
regardless of whether these are SIP or SCCP.
Contents
• Prerequisites for Cisco Unified SIP SRST 4.1, page 127
• Restrictions for Cisco Unified SIP SRST 4.1, page 128
• Information About Cisco Unified SIP SRST 4.1, page 128
• How to Configure Cisco Unified SIP SRST 4.1 Features, page 131
Prerequisites for Cisco Unified SIP SRST 4.1

• Cisco IOS Release 12.4(15)T or a later release.
• Cisco Unified IP Phones 7911G, 7941G, 7941GE, 7961G, 7961GE, 7970G, and 7971GE require
firmware load 8.2(1) or a later version.
• For the prerequisites for the Enhanced 911 Services for Cisco Unified SRST feature introduced in
Version 4.1, see Prerequisites for Enhanced 911 Services.

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Chapter 5 Cisco Unified SIP SRST 4.1
Restrictions for Cisco Unified SIP SRST 4.1
Restrictions for Cisco Unified SIP SRST 4.1

• Cisco Unified SRST does not support BLF speed-dial notification, call forward all synchronization,
dial plans, directory services, or music-on-hold (MOH).
• Prior to SIP phone load 8.0, SIP phones maintained dual registration with both
Cisco Unified Communications Manager and Cisco Unified SRST simultaneously. In SIP phone
load 8.0 and later versions, SIP phones use keepalive to maintain a connection with
Cisco Unified SRST during active registration with Cisco Unified Communications Manager.
Every 2 minutes, a SIP phone sends a keepalive message to Cisco Unified SRST.
Cisco Unified SRST responds to this keepalive with a 404 message. This process repeats until
fallback to Cisco Unified SRST occurs. After fallback, SIP phones send a keepalive message every
two minutes to Cisco Unified Communications Manager while the phones are registered with
Cisco Unified SRST. Cisco Unified SRST continues to support dual registration for SIP phone loads
older than 8.0.
Information About Cisco Unified SIP SRST 4.1

• Out-of-Dialog REFER, page 128
• Digit Collection on SIP Phones, page 129
• Caller ID Display, page 130
• Idle Prompt Status, page 130
Out-of-Dialog REFER
Out-of-dialog REFER (OOD-R) enables remote applications to establish calls by sending a REFER
message to Cisco Unified SRST without an initial INVITE. After the REFER is sent, the remainder of
the call setup is independent of the application and the media stream does not flow through the
application. The application using OOD-R triggers a call setup request that specifies the Referee address
in the Request-URI and the Refer-Target in the Refer-To header. The SIP messaging used to
communicate with Cisco Unified SRST is independent of the end-user device protocol, which can be
H.323, plain old telephone service (POTS), SCCP, or SIP. Click-to-dial is an example of an application
that can be created using OOD-R.
A click-to-dial application enables users to combine multiple steps into one click for a call setup. For
example, a user can click a web-based directory application from his or her PC to look up a telephone
number, off-hook the desktop phone, and dial the called number. The application initiates the call setup
without the user having to out-dial from his or her own phone. The directory application sends a REFER
message to Cisco Unified SRST, which sets up the call between both parties based on this REFER.
For more information about OOD-R, see Out-of-Dialog REFER from the Cisco Unified Communications
Manager Express System Administrator Guide.

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Digit Collection on SIP Phones

Digit strings dialed by phone users must be collected and matched against predefined patterns to place
calls to the destination corresponding to the user's input. Previously, SIP phones in a
Cisco Unified SRST system required users to press the DIAL soft key or # key, or wait for the
interdigit-timeout to trigger call processing. This could cause delays in processing the call.
Two new methods of collecting and matching digits are supported for SIP phones depending on the
model of the phone:
• KPML Digit Collection, page 129
• SIP Dial Plans, page 129
KPML Digit Collection

The Key Press Markup Language (KPML) uses SIP SUBSCRIBE and NOTIFY methods to report user
input digit by digit. Each digit dialed by the phone user generates its own signaling message to
Cisco Unified SRST, which performs pattern recognition by matching a destination pattern to a dial peer
as it collects the dialed digits. This process of relaying each digit immediately is similar to the process
used by SCCP phones. It eliminates the need for the user to press the Dial soft key or wait for the
interdigit timeout before the digits are sent to the Cisco Unified SRST for processing.
KPML is supported on Cisco Unified IP Phones 7911G, 7941G, 7941GE, 7961G, 7961GE, 7970G, and
7971GE. For configuration information, see the “Enabling KPML for SIP Phones” section on page 131.
SIP Dial Plans

A dial plan is a set of dial patterns that SIP phones use to determine when digit collection is complete
after a user goes off-hook and dials a destination number. Dial plans enable SIP phones to perform local
digit collection and recognize dial patterns as user input is collected. After a pattern is recognized, the
SIP phone sends an INVITE message to Cisco Unified SRST to initiate the call to the number matching
the user's input. All of the digits entered by the user are presented as a block to Cisco Unified SRST for
processing. Because digit collection is done by the phone, dial plans reduce signaling messages overhead
compared to KPML digit collection.
SIP dial plans eliminate the need for a user to press the Dial soft key or # key or to wait for the interdigit
timeout to trigger an outgoing INVITE. You configure a SIP dial plan and associate the dial plan with a
SIP phone. The dial plan is downloaded to the phone in the configuration file.
You can configure SIP dial plans and associate them with the following SIP phones:
• Cisco Unified IP Phone 7911G, 7941G, 7941GE, 7961G, 7961GE, 7970G, and 7971GE: These
phones use dial plans and support KPML. If both a dial plan and KPML are enabled, the dial plan
has priority.
If a matching dial plan is not found and KPML is disabled, the user must wait for the interdigit
timeout before the SIP NOTIFY message is sent to Cisco Unified SRST. Unlike other SIP phones,
these phones do not have a Dial soft key to indicate the end of dialing, except when on-hook dialing
is used.
• Cisco Unified IP Phone 7905, 7912, 7940, and 7960: These phones use dial plans and do not support
KPML. If you do not configure a SIP dial plan for these phones, or if the dialed digits do not match
a dial plan, the user must press the Dial soft key or wait for the interdigit timeout before digits are
sent to Cisco Unified SRST for processing.

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When you reset a phone, the phone requests its configuration files from the TFTP server, which builds
the appropriate configuration files depending on the type of phone.
• Cisco Unified IP Phone 7905 and 7912: The dial plan is a field in their configuration files.
• Cisco Unified IP Phone 7911G, 7940, 7941G, 7941GE, 7960, 7961G, 7961GE, 7970G, and
7971GE: The dial plan is a separate XML file that is pointed to from the normal configuration file.
The Cisco Unified SRST supports SIP dial plans if they are provisioned in
Cisco Unified Communications Manager. You cannot configure dial plans in Cisco Unified SRST.
Caller ID Display
The name and number of the caller is included in the Caller ID display on the
Cisco Unified IP Phone 7911G, 7941G, 7941GE, 7961G, 7961GE, 7970G, and 7971GE. Other SIP
phones display only the number of the caller. Also, the caller ID information is updated on the
destination phone when there is a change in the caller ID of the originating party such as with call
forwarding or call transfer. No new configuration is required to support these enhancements.
If a destination gateway does not support supplementary services, you can disable REFER messages for
call transfers and redirect responses for call forwarding from being sent by Cisco Unified SRST.
Disabling supplementary services is supported if all endpoints use SCCP or all endpoints use SIP. It is
not supported for a mix of SCCP and SIP endpoints.
Idle Prompt Status

A message displays on the status line of a SIP phone after the phone registers to Cisco Unified SRST to
indicate that Cisco Unified SRST is providing fallback support for the Cisco Unified Communications
Manager. This message informs the user that the phone is operating in fallback mode and that not all
features are available. The default message that displays “CM Fallback Service Operating” is taken from
the phone dictionary file. You can customize the message by using the system message command on the
Cisco Unified SRST router. Cisco Unified SRST updates the idle prompt message when a SIP phone
registers or when you modify the message through the configuration. The message displays until a phone
switches back to the Cisco Unified Communications Manager.
The idle prompt status message is supported for the Cisco Unified IP Phone 7911G, 7941G, 7941GE,
7961G, 7961GE, 7970G, and 7971GE with Cisco Unified SRST 4.1 and later versions. For versions
earlier than Cisco Unified SRST 4.1, the phones display the default message from the dictionary file.
Enhanced 911 Services

Enhanced 911 Services for Cisco Unified SRST enables 911 operators to:
• Immediately pinpoint the location of the 911 caller based on the calling number
• Callback the 911 caller if a disconnect occurs
Before this feature was introduced, Cisco Unified SRST supported only outbound calls to 911. With
basic 911 functionality, calls were simply routed to a Public Safety Answering Point (PSAP). The 911
operator at the PSAP would then have to verbally gather the emergency information and location from

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How to Configure Cisco Unified SIP SRST 4.1 Features
the caller, before dispatching a response team from the ambulance service, fire department, or police
department. Calls could not be routed to different PSAPs, based on the specific geographic areas that
they cover.
With Enhanced 911 Services, 911 calls are selectively routed to the closest PSAP based on the caller’s
location. In addition, the caller’s phone number and address automatically display on a terminal at the
PSAP. Therefore, the PSAP can quickly dispatch emergency help, even if the caller is unable to
communicate the location. Also, if the caller disconnects prematurely, the PSAP has the information it
needs to contact the 911 caller.
See Configuring Enhanced 911 Services from Cisco Unified Communications Manager Express System
Administrator Guide for more information.

• Enabling KPML for SIP Phones, page 131
Enabling KPML for SIP Phones

Perform the following steps to enable KPML digit collection on a SIP phone.
Restrictions
• This feature is supported only on Cisco Unified IP Phone 7911G, 7941G, 7941GE, 7961G, 7961GE,
7970G, and 7971GE.
• A dial plan assigned to a phone has priority over KPML.
SUMMARY STEPS
1. enable
4. digit collect kpml
5. end
6. show voice register dial-peer

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DETAILED STEPS

Example:
Router> enable
Example:
Example: • pool-tag: Unique sequence number of the SIP phone to
Router(config)# voice register pool 4 be configured. Range is version and
platform-dependent; type ? to display range. You can
modify the upper limit for this argument with the
max-pool command.
Step 4 digit collect kpml Enables KPML digit collection for the SIP phone.
Note This command is enabled by default for supported
Example: phones in Cisco Unified CME and
Router(config-register-pool)# digit collect Cisco Unified SRST.
kpml
Example:
Step 6 show voice register dial-peers Displays details of all dynamically created VoIP dial peers
associated with the Cisco Unified CME SIP register
including the defined digit collection method.
Example:
What to Do Next
After changing the KPML configuration in Cisco Unified SRST, you do not need to create new
configuration profiles and restart the phones. Enabling or disabling KPML is effective immediately in
Cisco Unified SRST.

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Perform the following steps to disable REFER messages for call transfers and redirect responses for call
forwarding from being sent to the destination by Cisco Unified SRST. You can disable these
supplementary features if the destination gateway does not support them.
SUMMARY STEPS
1. enable
or
dial-peer voice tag voip
4. no supplementary-service sip {moved-temporarily | refer}
5. end
DETAILED STEPS

Example:
Router> enable
Example:
Step 3 voice service voip Enters voice-service configuration mode to set global
or parameters for VoIP features.
dial-peer voice tag voip or
Enters dial peer configuration mode to set parameters for a
Example: specific dial peer.
or

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Step 4 no supplementary-service sip {moved-temporarily Disables SIP call forwarding or call transfer supplementary
| refer} services globally or for a dial peer.
• moved-temporarily: SIP redirect response for call
Example: forwarding.
Router(conf-voi-serv)# no supplementary-service
sip refer • refer: SIP REFER message for call transfers.
or • Sending REFER and redirect messages to the
Router(config-dial-peer)# no destination is the default behavior.
Note This command is supported for calls between SIP
phones and calls between SCCP phones. It is not
supported for a mixture of SCCP and SIP endpoints.
Example:
Router(config-voi-serv)# end
or
Configuring Idle Prompt Status for SIP Phones

Perform the following steps to customize the message that displays on SIP phones after the phones
failover to Cisco Unified SRST.
Note You do not need to create new configuration files with the create profile command and restart the phones
after changing the idle status message in Cisco Unified SRST. Modifying the status message takes effect
immediately in Cisco Unified SRST.
Prerequisites
Cisco Unified SRST 4.1 or a later version.
SUMMARY STEPS
1. enable
4. system message string
5. end
6. show voice register global

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Where to Go Next
DETAILED STEPS

Example:
Router> enable
Example:
global parameters for all supported SIP phones in a
Cisco Unified CME environment.
Example:
Step 4 system message string Defines a status message that displays on SIP phones
registered to Cisco Unified SRST.
Example: • string: Up to 32 alphanumeric characters. Default is
Router(config-register-global)# system message “CM Fallback Service Operating.”
fallback active
Example:
Step 6 show voice register global Displays all global configuration parameters associated
with SIP phones.
Example:
Where to Go Next
The next step is configuring Cisco Unified IP phones using SCCP. For instructions, see the “Setting Up
Cisco Unified IP Phones using SCCP” section on page 137.

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Where to Go Next

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CHAPTER 6
Setting Up Cisco Unified IP Phones using
SCCP
This chapter describes how to set up the displays and features that callers will see and use on Cisco
Unified IP Phones during Cisco Unified CM fallback.
Note Ciso Unified IP Phones discussed in this chapter are just examples. For a complete list of IP phones, see
Compatibility Information.
Contents
• Information About Setting Up Cisco Unified IP Phones, page 137
• How to Set Up Cisco Unified IP Phones, page 138
• How to Set Up Cisco IP Communicator for Cisco Unified SRST, page 154
Information About Setting Up Cisco Unified IP Phones

Cisco Unified IP Phone configuration is limited for Cisco Unified SRST because IP phones retain nearly
all Cisco Unified CM settings during Cisco Unified CM fallback. You can configure the date format,
time format, language, and system messages that appear on Cisco Unified IP Phones during
Cisco Unified Communications Manager fallback. All four of these settings have defaults, and the
available language options depend on the IP phones and Cisco Unified CM version in use. Also available
for configuration is a secondary dial tone, which can be generated when a phone user dials a predefined
PSTN access prefix and can be terminated when additional digits are dialed. Dual-line phone
configuration is required for dual-line phone operation during Cisco Unified CM fallback.

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Chapter 6 Setting Up Cisco Unified IP Phones using SCCP
How to Set Up Cisco Unified IP Phones

• Configuring Cisco Unified SRST to Support Phone Functions, page 138 (Required)
• Configuring Cisco Unified 8941 and 8945 SCCP IP Phones, page 140 (Required)
• Verifying That Cisco Unified SRST Is Enabled, page 141 (Optional)
• Configuring IP Phone Clock, Date, and Time Formats, page 142 (Optional)
• Configuring IP Phone Language Display, page 144 (Optional)
• Configuring Customized System Messages for Cisco Unified IP Phones, page 146 (Optional)
• Configuring a Secondary Dial Tone, page 147 (Optional)
• Configuring Dual-Line Phones, page 148 (Required Under Certain Conditions)
• Configuring Eight Calls per Button (Octo-Line), page 150 (Optional)
• Configuring the Maximum Number of Calls, page 152 (Optional)
• Troubleshooting, page 154 (Optional)
Configuring Cisco Unified SRST to Support Phone Functions
Tip When the Cisco Unified SRST is enabled, Cisco Unified IP Phones do not have to be reconfigured while
in Cisco Unified Communications Manager fallback mode because phones retain the same configuration
that was used with Cisco Unified Communications Manager.
To configure Cisco Unified SRST on the router to support the Cisco Unified IP Phone functions, use the
following commands beginning in global configuration mode.
SUMMARY STEPS
2. ip source-address ip-address [port port] [any-match | strict-match]
3. max-dn max-directory-numbers [dual-line] [preference preference-order]
5. limit-dn phone-type max-lines
6. exit

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DETAILED STEPS

Example:
[any-match | strict-match] phones through the specified IP addresses and provides
for strict IP address verification. The default port number
Example: is 2000.
Router(config-cm-fallback)# ip source-address
10.6.21.4 port 2002 strict-match
Step 3 max-dn max-directory-numbers [dual-line] Sets the maximum number of directory numbers (DNs)
[preference preference-order] or virtual voice ports that can be supported by the router
and activates the dual-line mode.
Example: • max-directory-numbers: Maximum number of
Router(config-cm-fallback)# max-dn 15 dual-line directory numbers (dns) or virtual voice ports
preference 1
platform-dependent. The default is 0. See
Compatibility Information for further details.
• dual-line (Optional). Allows IP phones in
Cisco Unified Communications Manager fallback
mode to have a virtual voice port with two channels.
• preference preference-order (Optional). Sets the
global preference for creating the VoIP dial peers for
all directory numbers that are associated with the
primary number. Range is from 0 to 10. Default is 0,
which is the highest preference.
The alias command also has a preference keyword
that sets alias command preference values. Setting
the alias command preference keyword allows the
default preference set with the max-dn command to
be overridden. See the “Configuring Call Rerouting”
section on page 182 for more information on using
the max-dn command with the alias command.
Note You must reboot the router to reduce the limit of
the directory numbers or virtual voice ports after
the maximum allowable number is configured.
Step 4 max-ephones max-phones Configures the maximum number of Cisco IP phones
that can be supported by the router. The default is 0. The
maximum number is platform dependent. See
Example:
Router(config-cm-fallback)# max-ephones 24
Note You must reboot the router to reduce the limit of
Cisco IP phones after the maximum allowable
number is configured.

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Step 5 limit-dn phone-type max-lines (Optional) Limits the directory number lines on
Cisco IP phones during Cisco Unified CM fallback.
Example: Note You must configure this command during initial
Router(config-cm-fallback)# limit-dn 7945 2 Cisco Unified SRST router configuration, before
any phone actually registers with the
Cisco Unified SRST router. However, you can
modify the number of lines at a later time.
For a list of available phones, see Cisco SRST

and SIP SRST Command Reference (All
Versions).
The setting for maximum lines is from 1 to 6. The default

number of maximum directory lines is set to 6. If there is
any active phone with the last line number greater than
this limit, warning information is displayed for phone
reset.
Example:
Configuring Cisco Unified 8941 and 8945 SCCP IP Phones

To configure Cisco Unified 8941 and 8945 SCCP IP Phones in SRST mode, perform the following
commands:
Note This section is required only in SRST version 8.6 and is not required for version 8.6 and higher.
SUMMARY STEPS
1. enable
3. ephone-type phone-type
4. device-id number
5. device-type phone-type
6. end

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DETAILED STEPS

Example:
Router> enable
Example:
Step 3 ephone-type phone-type Enters phone type to configure.
• 8941
Example: • 8945
Router(config)# ephone-type 8941
Step 4 device-id number Specifies the device ID for the phone type.
• 8941—586
Example: • 8945—585
Router(config-ephone-type)# device-id 586
Step 5 device-type phone-type Specifies the device type for the phone.
• 8941
Example: • 8945
Router(config-ephone-type)# device-type 8941
Example:
Router(config-ephone-type)# end
Verifying That Cisco Unified SRST Is Enabled

To verify that the Cisco Unified SRST feature is enabled, perform the following steps:
Step 1 Enter the show running-config command to verify the configuration.

Step 2 Enter the show call-manager-fallback all command to verify that the Cisco Unified SRST feature is
enabled.
Step 3 Use the Settings display on the Cisco IP phones in your network to verify that the default router IP
address on the phones matches the IP address of the Cisco Unified SRST router.

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Step 4 To temporarily block the TCP port 2000 Skinny Client Control Protocol (SCCP) connection for one of
the Cisco IP phones to force the Cisco IP phone to lose its connection to the
Cisco Unified Communications Manager and register with the Cisco Unified SRST router, perform the
following steps:
a. Use the appropriate IP access-list command to temporarily disconnect a Cisco Unified IP Phone
from the Cisco Unified Communications Manager.
During a WAN connection failure, when Cisco Unified SRST is enabled, Cisco Unified IP Phones
display a message informing you that they are operating in Cisco Unified Communications Manager
fallback mode. The Cisco IP Phone 7960 and Cisco IP Phone 7940 display a “CM Fallback Service
Operating” message, and the Cisco IP Phone 7910 displays a “CM Fallback Service” message when
operating in Cisco Unified Communications Manager fallback mode. When the Cisco
Unified Communications Manager is restored, the message goes away and full Cisco IP phone
functionality is restored.
b. Use the debug ephone register command to observe the registration process of the Cisco IP phone
on the Cisco Unified SRST router.
c. Use the show ephone command to display the Cisco IP phones that have registered to the
Cisco Unified SRST router.
d. Enter the no form of the appropriate access-list command to restore normal service for the phone.
Configuring IP Phone Clock, Date, and Time Formats

The Cisco Unified IP Phone 7970G and Cisco Unified IP Phone 7971G-GE IP phones obtain the correct
timezone from Cisco Unified Communications Manager. They also receive the Coordinated Universal
Time (UTC) time from the SRST router during SRST registration. When in SRST mode, the phones take
the timezone and the UTC time, and apply a timezone offset to produce the correct time display.
Cisco IP Phone 7960 IP phones and other similar SCCP phones such as the Cisco IP Phone 7940, get
their display clock information from the local time of the SRST router during SRST registration. If the
Cisco Unified SRST router is configured to use the Network Time Protocol (NTP) to automatically sync
the Cisco Unified SRST router time from an NTP time server, only UTC time is delivered to the router.
This is because the NTP server could be physically located anywhere in the world, in any timezone. As
it is important to display the correct local time, use the clock timezone command to adjust or offset the
Cisco Unified SRST router time.
The date and time formats that appear on the displays of all Cisco Unified IP Phones in Cisco Unified
CM fallback mode are selected using the date-format and time-format commands as configured below:
SUMMARY STEPS
1. clock timezone zone hours-offset [minutes-offset]

3. date-format {mm-dd-yy | dd-mm-yy | yy-dd-mm | yy-mm-dd}
4. time-format {12 | 24}
5. exit

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DETAILED STEPS

Step 1 clock timezone zone hours-offset Sets the time zone for display purposes.
[minutes-offset]
• zone: Name of the time zone to be displayed when
standard time is in effect. The length of the zone
Example: argument is limited to 7 characters.
Router(config)# clock timezone PST -8
• hours-offset: The number of hour difference from
Coordinated Universal Time (UTC).
• minutes-offset (Optional). Minutes difference from
UTC.
Example:
Step 3 date-format {mm-dd-yy | dd-mm-yy | yy-dd-mm | Sets the date format for IP phone display. The choices are
yy-mm-dd} mm-dd-yy, dd-mm-yy, yy-dd-mm, and yy-mm-dd, where
• dd: day
Example:
Router(config-cm-fallback)# date-format
• mm: month
yy-dd-mm • yy: year
The default is set to mm-dd-yy.
Step 4 time-format {12 | 24} Sets the time display format on all Cisco Unified IP Phones
registered with the router. The default is set to a 12-hour
clock.
Example:
Router(config-cm-fallback)# time-format 24
Example:
Example
The following example sets the time zone to Pacific Standard Time (PST), which is 8 hours behind UTC
and sets the time display format to a 24 hour clock:
Router(config)# clock timezone PST -8
Rounter(config)# call-manager-fallback
Rounter(config-cm-fallback)# time-format 24

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Configuring IP Phone Language Display

During Cisco Unified CM fallback, the language displays shown on
Cisco Unified IP Phones default to the ISO-3166 country code of US (United States). The
Cisco Unified IP Phone 7940 and Cisco Unified IP Phone 7960 can be configured for different languages
(character sets and spelling conventions) using the user-locale command.
Note This configuration option is available in Cisco SRST V2.1 and later versions running under
Cisco Unified CM V3.2 and later versions. Systems with software prior to
Cisco Unified SRST V2.1 and Cisco Unified CM V3.2 can use the default country, United States (US),
only.
SUMMARY STEPS
2. user-locale country-code
3. exit

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DETAILED STEPS

Example:
Step 2 user-locale country-code Selects a language by country for displays on the Cisco IP
Phone 7940 and Cisco IP Phone 7960.
Example: The following ISO-3166 codes are available to Cisco SRST
Router(config-cm-fallback)# user-locale ES and Cisco Unified SRST systems running under
Cisco Communications Manager V3.2 or later versions:
• DE: German.
• DK: Danish.
• ES: Spanish.
• FR: French.
• IT: Italian.
• JP: Japanese Katakana (available under
Cisco Unified Communications Manager V4.0 or later
versions).
• NL: Dutch.
• NO: Norwegian.
• PT: Portuguese.
• RU: Russian.
• SE: Swedish.
• US: United States English (default).
Example:
Examples
The following example offers a configuration for the Portugal user locale:
user-locale PT

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Configuring Customized System Messages for Cisco Unified IP Phones

Use the system message command to customize the system message displayed on all Cisco Unified IP
Phones during Cisco Unified CM fallback.
One of two keywords, primary and secondary, must be included in the command. The primary
keyword is for IP phones that can support static text messages during fallback. The default display
message for primary IP phones in fallback mode is “CM Fallback Service Operating.”
The secondary keyword is for Cisco Unified IP Phones that do not support static text messages and have
a limited display space. Secondary IP phones flash messages during fallback. The default display
message for secondary IP phones in fallback mode is “CM Fallback Service.”
Changes to the display message will occur immediately after configuration or at the end of each call.
Note The normal in-service static text message is controlled by Cisco Unified Communications Manager.
SUMMARY STEPS
2. system message {primary primary-string | secondary secondary-string}
3. exit
DETAILED STEPS

Example:
Step 2 system message {primary primary-string | Declares the text for the system display message on IP
secondary secondary-string} phones in fallback mode.
• primary primary-string: For Cisco Unified IP Phones
Example: that can support static text messages during fallback,
Router(config-cm-fallback)# system message such as the Cisco Unified IP Phone 7940 and Cisco
primary Custom Message
Unified IP Phone 7960 units. A string of approximately
27 to 30 characters is allowed.
• secondary secondary-string: For Cisco Unified IP
Phones that do not support static text messages, such as
the Cisco Unified IP Phone 7910. A string of
approximately 20 characters is allowed.
Example:

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Examples
The following example sets “SRST V3.0” as the system display message for all Cisco Unified IP Phones
on a router:
system message primary SRST V3.0
system message secondary SRST V3.0
exit
Configuring a Secondary Dial Tone

A secondary dial tone can be generated when a phone user dials a predefined PSTN access prefix and
can be terminated when additional digits are dialed. An example is when a secondary dial tone is heard
after the number 9 is dialed to reach an outside line.
SUMMARY STEPS
2. secondary-dialtone digit-string
3. exit
DETAILED STEPS

Example:
Step 2 secondary-dialtone digit-string Activates a secondary dial tone when a digit string is dialed.
Example:
Router(config-cm-fallback)# secondary-dialtone
9
Example:
Examples
The following example sets the number 8 to trigger a secondary dial tone:
secondary-dialtone 8

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Configuring Dual-Line Phones

Dual-line phone configuration is required for dual-line phone operation during Cisco Unified CM
fallback, see the “Enabling Consultative Call Transfer and Forward Using H.450.2 and H.450.3 with
Cisco SRST 3.0” section on page 200.
Dual-line IP phones are supported during Cisco Unified CM fallback using the max-dn command.
Dual-line IP phones have one voice port with two channels to handle two independent calls. This
capability enables call waiting, call transfer, and conference functions on a phone-line button.
In dual-line mode, each IP phone and its associated line button can support one or two calls. Selection
of one of two calls on the same line is made using the blue Navigation button located below the phone
display. When one of the dual-line channels is used on a specific phone, other phones that share the
ephone-dn will be unable to use the secondary channel. The secondary channel will be reserved for use
with the primary dual-line channel.
It is recommended that hunting be disabled to the second channel. For more information, see the
“Configuring Dial-Peer and Channel Hunting” section on page 196.
SUMMARY STEPS
3. exit

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DETAILED STEPS

Example:
Step 2 max-dn max-directory-numbers [dual-line] Sets the maximum number of directory numbers (DNs) or
[preference preference-order] virtual voice ports that can be supported by the router and
activates dual-line mode.
Router(config-cm-fallback)# max-dn 15 dual-line directory numbers (dns) or virtual voice ports
preference 1
platform-dependent. The default is 0. See
• dual-line (Optional). Allows IP phones in
• preference preference-order (Optional). Sets the
global preference for creating the VoIP dial peers for all
directory numbers that are associated with the primary
number. Range is from 0 to 10. Default is 0, which is
the highest preference.
The alias command also has a preference keyword that
sets alias command preference values. Setting the alias
command preference keyword allows the default
preference set with the max-dn command to be
overridden. See the “Configuring Call Rerouting”
section on page 182 for more information on using the
max-dn command with the alias command.
Example:
Examples
The following example sets the maximum number of DNs or virtual voice ports that can be supported
by a router to 10 and activates the dual-line mode for all IP phones in Cisco Unified CM fallback mode:
max-dn 10 dual-line
exit

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Configuring Eight Calls per Button (Octo-Line)

The octo-line feature supports up to eight active calls, both incoming and outgoing, on a single button.
Eight incoming calls to an octo-line directory number ring simultaneously. After an incoming call is
answered, the ringing stops and the remaining seven incoming calls hear a call waiting tone.
After an incoming call on an octo-line directory number is answered, the answering phone is in the
connected state. Other phones that share the directory number are in the remoteMultiline state. A
subsequent incoming call sends the call waiting tone to the phone connected to the call, and sends the
ringing tone to the other phones that are in the remoteMultiline state. All phones sharing the directory
number can pick up any of the incoming unanswered calls.
When multiple incoming calls ring on an octo-line directory number that is shared among multiple
phones, the ringing tone stops on the phone that answers the call, and the call waiting tone is heard for
other unanswered calls. The multiple instances of the ringing calls is displayed on other ephones sharing
the directory number. After a connected call on an octo-line directory number is put on-hold, any phone
that shares this directory number can pick up the held call. If a phone is in the process of transferring a
call or creating a conference, other phones that share the octo-line directory number cannot steal the call.
As new calls come in on an octo-line, the system searches for the next available idle line using the
huntstop chan tag command, where tag is a number from 1 to 8. An idle channel is selected from the
lowest number to the highest. When the highest number of allowed calls is received, the system stops
hunting for available channels. Use this command to limit the number of incoming calls on an octo-line
directory number and reserve channels for outgoing calls or features such as call transfer or conference
calls.
With the new feature, you can:
• Configure only dual-line mode
• Configure only octo-line mode
• Configure dual-line mode and octo-line mode
Prerequisites
• Cisco Unified SRST 7.0/4.3
• Cisco Unified CM 6.0
• Cisco IOS Release 12.4(15)XZ
Restrictions
Octo-line directory numbers are not supported by the Cisco Unified IP Phone 7902, 7920, or 7931, or
by analog phones connected to Cisco ATA or Cisco VG224.
SUMMARY STEPS
1. enable
4. max-dn max-no-of-directories [dual-line | octo-line] [number octo-line]
5. huntstop channel 1-8
6. end

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DETAILED STEPS

Example:
Router> enable
Example:
Example:
Step 4 max-dn max-no-of-directories [dual-line | Sets the maximum number of DNs or virtual voice ports
octo-line] [number octo-line] that can be supported by the router and activates dual-line
mode, octo-line mode, or both modes.
Example: • max-no-of-directories: Maximum number of directory
Router(config-cm-fallback)# max-dn 15 dual-line numbers (dns) or virtual voice ports supported by the
6 octo-line
router. The maximum number is platform-dependent.
The default is 0.
• dual-line: (Optional) Allows IP phones in
• octo-line: (Optional) Allows IP phones in
mode to have a virtual voice port with eight channels.
• number (Optional): Sets the number of directory
numbers for octo-mode.
Step 5 huntstop channel 1-8 Enables channel huntstop on an octo-line, which keeps a
call from hunting to the next channel of a directory number
if the last allowed channel is busy or does not answer.
Example:
Router(config-cm-fallback)# huntstop channel 4 • number: Number of channels available to accept
incoming calls. The remaining channels are reserved
for outgoing calls and features such as call transfer,
call waiting, and conferencing. The range is 1 to 8 and
the default is 8.
• The command is supported for octo-line directory
numbers only.
Example:
Router(config)# end

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Examples
In the following example, octo-line mode is enabled, there are 8 octo-line directory numbers, there are
a maximum of 23 directory numbers, and a maximum of 6 channels are available for incoming calls:
!
max-dn 23 octo-line 8
huntstop channel 6
Configuring the Maximum Number of Calls

To configure the maximum number of calls on a Cisco Unified SCCP IP phone in Cisco Unified SRST
9.0, perform the following steps.
Prerequisites
• Cisco Unified SRST 9.0 and later versions.
• Correct firmware, 9.2(1) or a later version, is installed.
SUMMARY STEPS
1. enable
4. max-dn max-no-of-directories [dual-line | octo-line]
5. timeouts busy seconds
6. end
DETAILED STEPS

Example:
Router> enable
Example:
Step 3 call-manager-fallback Enables Cisco Unified SRST support and enters
call-manager-fallback configuration mode.
Example:

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Step 4 max-dn max-no-of-directories [dual-line | Sets the maximum possible number of directory numbers or
octo-line] virtual voice ports that can be supported by a router and
enables dual-line mode, octo-line mode, or both modes.
Example: • max-no-of-directories—Maximum number of directory
Router(config-cm-fallback)# max-dn 10 octo-line numbers or virtual voice ports supported by the router.
The maximum possible number is platform-dependent.
The default is 0 directory numbers and 1 channel per
virtual port.
• dual-line—(Optional) Sets all Cisco Unified IP phones
connected to a Cisco Unified SRST router to one virtual
voice port with two channels.
• octo-line—(Optional) Sets all Cisco Unified IP phones
connected to a Cisco Unified SRST router to one virtual
voice port with eight channels.
Step 5 timeouts busy seconds Sets the timeout value for call transfers to busy destinations.
• seconds—Number of seconds after connection to a
Example: busy destination before a transferred call is
Router(config-cm-fallback)# timeouts busy 10 disconnected. Range is 0 to 30. Default: 10.
Step 6 end Exits configuration mode and enters privileged EXEC
mode.
Example:

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How to Set Up Cisco IP Communicator for Cisco Unified SRST
Troubleshooting
To troubleshoot your Cisco Unified SRST configuration, use the following commands:
• To set keepalive debugging for Cisco IP phones, use the debug ephone keepalive command.
• To set registration debugging for Cisco IP phones, use the debug ephone register command.
• To set state debugging for Cisco IP phones, use the debug ephone state command.
• To set detail debugging for Cisco IP phones, use the debug ephone detail command.
• To set error debugging for Cisco IP phones, use the debug ephone error command.
• To set call statistics debugging for Cisco IP phones, use the debug ephone statistics command.
• To provide voice-packet-level debugging and to display the contents of one voice packet in every
1024 voice packets, use the debug ephone pak command.
• To provide raw low-level protocol debugging display for all SCCP messages, use the debug ephone
raw command.
For further debugging, see Cisco IOS Debug Command Reference.
How to Set Up Cisco IP Communicator for Cisco Unified

SRST
Cisco IP Communicator is a software-based application that delivers enhanced telephony support on
personal computers. Cisco IP Communicator appears on a user’s computer monitor as a graphical,
display-based IP phone with a color screen, a keypad, feature buttons, and soft keys.
For information about operation, see the Cisco IP Communicator online help and user documentation.
Prerequisites
You should have the following before you begin this task:
• IP address of the Cisco Unified CM (Call Manager) TFTP server
• IP address of the Cisco Unified SRST TFTP server
• Headset with microphone for your PC (Optional; you can use PC internal speakers and microphone)
Step 1 Download the latest version of the Cisco IP Communicator software and install it on your PC. The
software is available for download at http://www.cisco.com/cisco/web/download/index.html.
a. Click Voice and Unified Communication.
b. Click IP Telephony.
c. Click IP Phones.
d. Click Cisco IP Communicator.
Step 2 (Optional) Attach a headset to your PC.
Step 3 Start the Cisco IP Communicator software application.

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Where to Go Next
Step 4 Define the IP address of the Cisco Unified CM as primary TFTP server
a. Open the Network > User Preferences window.
b. Enter the IP address of the Cisco Unified CM TFTP server.
Step 5 Define the IP address of the Cisco Unified SRST as secondary TFTP server.
a. Open the Network > User Preferences window.
b. Enter the IP address of the Cisco Unified SRST TFTP server.
Step 6 Ensure that Cisco IP Communicator has at least once registered to Cisco Unified CM. For more details,
see Install and Configure IP Communicator with CallManager.
Step 7 Wait for the Cisco IP Communicator to connect to the Cisco Unified SRST system (upon Cisco Unified
CM Failure) and register itself.
Step 8 Cisco IP Communicator should have retained the original buttons and numbers for Cisco IP
Communicator.
Verifying Cisco IP Communicator

Step 1 Use the show running-config command to display ephone-dn and ephone information associated with
this phone.
Step 2 After Cisco IP Communicator registers with Cisco Unified SRST, it displays the phone extensions and
soft keys in its configuration. Verify that these are correct.
Step 3 Make a local call from the phone and ask someone to call you. Verify that you have a two-way voice path.
Troubleshooting Cisco IP Communicator

Use the debug ephone detail command to diagnose problems with calls. For more information, see
Cisco IOS Debug Command Reference.
Where to Go Next
The next step is configuring Cisco Unified IP Phones using SIP. For more information, see the “” section
on page 157.
For additional information, see the “Additional References” section on page 30.

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Where to Go Next

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CHAPTER 7
Setting Up Cisco Unified IP Phones using SIP
Session Initiation Protocol (SIP) registrar functionality in Cisco IOS software is an essential part of
Cisco Unified SIP Survivable Remote Site Telephony (SRST). According to RFC 3261, a SIP registrar
is a server that accepts Register requests and is typically collocated with a proxy or redirect server. A
SIP registrar may also offer location services.
Contents
• Prerequisites for Configuring the SIP Registrar, page 157
• Restrictions for Configuring the SIP Registrar, page 157
• Information About Configuring the SIP Registrar, page 157
• How to Configure the SIP Registrar, page 158
Prerequisites for Configuring the SIP Registrar

Complete the prerequisites documented in the “Prerequisites for Configuring Cisco Unified SIP SRST”
section on page 9 section in “Cisco Unified SRST Feature Overview” section on page 1.
Restrictions for Configuring the SIP Registrar

See the restrictions documented in the “Restrictions for Configuring Cisco Unified SIP SRST” section
on page 9 section in “Cisco Unified SRST Feature Overview” section on page 1.
Information About Configuring the SIP Registrar

Cisco Unified SIP SRST provides backup to an external SIP call control (IP-PBX) by providing basic
registrar and call handling services. These services are used by a SIP IP phone in the event of a WAN
connection outage when the SIP phone is unable to communicate with its primary SIP proxy. The Cisco
Unified SIP SRST device also provides PSTN gateway access for placing and receiving PSTN calls.
Cisco Unified SIP SRST works for the following types of calls:

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Chapter 7 Setting Up Cisco Unified IP Phones using SIP
How to Configure the SIP Registrar
• Local SIP IP phone to local SIP phone, if the main proxy is unavailable.
• Additional services like class of restriction (COR) for local SIP IP phones to the outgoing PSTN.
For example, to block outgoing 1-900 numbers.

This section contains the following procedures:
• Configuring the SIP Registrar, page 158 (required)
• Configuring Backup Registrar Service to SIP Phones, page 160 (required)
• Configuring Backup Registrar Service to SIP Phones (Using Optional Commands), page 164
(optional)
• Verifying SIP Registrar Configuration, page 167 (optional)
• Verifying Proxy Dial-Peer Configuration, page 169 (optional)
Configuring the SIP Registrar

The local SIP gateway that becomes the SIP registrar acts as a backup SIP proxy and accepts SIP
Register messages from SIP phones. It becomes a location database of local SIP IP phones.
A registrar accepts SIP Register requests and dynamically builds VoIP dial peers, allowing the Cisco IOS
voice gateway software to route calls to SIP phones.
If a SIP Register request has a Contact header that includes a DNS address, the Contact header is
resolved before the contact is added to the SIP registrar database. This is done because during a WAN
failure (and the resulting Cisco Unified SIP SRST functionality), DNS servers may not be available.
SIP registrar functionality is enabled with the following configuration. By default, Cisco Unified SIP
SRST is not enabled and cannot accept SIP Register messages. The following configuration must be set
up to accept incoming SIP Register messages.
SUMMARY STEPS
1. enable
5. sip
6. registrar server [expires [max sec] [min sec]]

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7. end
DETAILED STEPS

Example:
Router> enable
Example:
Example:
Step 4 allow-connections sip to sip Allows connections from SIP to SIP endpoints.
Example:
Router(config-voi-srv)# allow-connections sip
to sip
Example:

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Step 6 registrar server [expires [max sec] [min sec]] Enables SIP registrar functionality. The keywords and
Example: • expires: (Optional) Sets the active time for an incoming
Router(conf-serv-sip)# registrar server expires registration.
max 600 min 60
• max sec: (Optional) Maximum expiration time for a
registration, in seconds. The range is from 600 to
86400. The default is 3600.
Note
Ensure that the registration expiration timeout is set
to a value smaller than the TCP connection aging
timeout to avoid disconnection from the TCP.

The default is 60.
Example:
What to Do Next
For incoming SIP Register messages to be successfully accepted, users must also set up a voice register
pool. See the “Configuring Backup Registrar Service to SIP Phones” section on page 160.
Configuring Backup Registrar Service to SIP Phones

Backup registrar service to SIP IP phones can be provided by configuring a voice register pool on SIP
gateways. The voice register pool configuration provides registration permission control and can also be
used to configure some dial-peer attributes that are applied to the dynamically created VoIP dial peers
when SIP phone registrations match the pool. The following call types are supported:
• SIP IP phone to or from:
– Local PSTN
– Local analog FXS phones
– Local SIP IP phone
The commands in the configuration below provide registration permission control and set up a basic
voice register pool. The pool gives users control over which registrations are accepted by a Cisco Unified
SIP SRST device and which can be rejected. Registrations that match this pool create VoIP SIP dial peers
with the dial-peer attributes set to these configurations. Although only the id command is mandatory,
this configuration example shows basic functionality.
For command-level information, see the appropriate command page in Cisco Unified SRST and Cisco
Unified SIP SRST Command Reference (All Versions).

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Prerequisites
• The SIP registrar must be configured before a voice register pool is set up. See the “Configuring the
SIP Registrar” section on page 158 for complete instructions.
Restrictions
• The id command identifies the individual SIP IP phone or sets of SIP IP phones that are to be
configured. Thus, the id command configured in Step 5 is required and must be configured before
any other voice register pool commands. When the mac address keyword and argument are used,
the IP phone must be in the same subnet as that of the router’s LAN interface, such that the phone’s
MAC address is visible in the router’s Address Resolution Protocol (ARP) cache. Once a MAC
address is configured for a specific voice register pool, remove the existing MAC address before
changing to a new MAC address.
• Proxy dial peers are autogenerated dial peers that route all calls from the PSTN to
Cisco Unified SIP SRST. When a SIP phone registers to Cisco Unified SIP SRST and the proxy
command is enabled, two dial peers are automatically created. The first dial peer routes to the proxy,
and the second (or fallback) dial peer routes to the SIP phone. The same functionality can also be
achieved with the appropriate creation of static dial peers (manually creating dial peers that point to
the proxy). Proxy dial peers can be monitored to one proxy IP address, only. That is, only one proxy
from a voice registration pool can be monitored at a time. If more than one proxy address needs to
be monitored, you must manually create and configure additional dial peers.
Note To monitor SIP proxies, the call fallback active command must be configured, as described in Step 3.
SUMMARY STEPS
1. enable
3. call fallback active
5. id {network address mask mask | ip address mask mask | mac address}
6. preference preference-order
8. voice-class codec tag
9. application application-name
10. end

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DETAILED STEPS

Example:
Router> enable
Example:
Step 3 call fallback active (Optional) Enables a call request to fall back to alternate
dial peers in case of network congestion.
Example: • This command is used if you want to monitor the proxy
Router(config)# call fallback active dial peer and fallback to the next preferred dial peer.
For full information on the call fallback active
command, see PSTN Fallback Feature.
phones.
Example: • Use this command to control which registrations are
Router(config)# voice register pool 12 accepted or rejected by a Cisco Unified SIP SRST
device.
Step 5 id {network address mask mask | ip address mask Explicitly identifies a locally available individual or set of
mask | mac address} SIP IP phones. The keywords and arguments are defined as
follows:
Example: • network address mask mask: The network address
Router(config-register-pool)# id network mask mask keyword/argument combination is used to
172.16.0.0 mask 255.255.0.0
subnet.
individual phone.
Step 6 preference preference-order Sets the preference order for the VoIP dial peers to be
created. Range is from 0 to 10. Default is 0, which is the
highest preference.
Example:
Router(config-register-pool)# preference 2 • The preference must be greater (lower priority) than the
preference configured with the preference keyword in
the proxy command.

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Step 7 proxy ip-address [preference value] [monitor Autogenerates additional VoIP dial peers to reach the main
probe {icmp-ping | rtr} [alternate-ip-address]] SIP proxy whenever a Cisco Unified SIP IP Phone registers
with a Cisco Unified SIP SRST gateway. The keywords and
Example: arguments are defined as follows:
Router(config-register-pool)# proxy • ip-address: IP address of the SIP proxy.
10.2.161.187 preference 1
• preference value: (Optional) Defines the preference of
the proxy dial peers that are created. The preference
must be less (higher priority) than the preference
configured with the preference command.
Range is from 0 to 10. The highest preference is 0.
There is no default.
• monitor probe: (Optional) Enables monitoring of
proxy dial peers.
• icmp-ping: Enables monitoring of proxy dial peers
using ICMP ping.
Note The dial peer on which the probe is configured will
be excluded from call routing only for outbound
calls. Inbound calls can arrive through this dial peer.
• rtr: Enables monitoring of proxy dial peers using RTR

probes.
• alternate-ip-address: (Optional) Enables monitoring of
alternate IP addresses other than the proxy address. For
example, to monitor a gateway front end to a SIP proxy.
Step 8 voice-class codec tag Sets the voice class codec parameters. The tag argument is
a codec group number between 1 and 10000.
Example:
Router(config-register-pool)# voice-class codec
15
Step 9 application application-name (Optional) Selects the session-level application on the VoIP
dial peer. Use the application-name argument to define a
specific interactive voice response (IVR) application.
Example:
Router(config-register-pool)# application
SIP.App
Example:
What to Do Next
There are several more voice register pool commands that add functionality, but that are not required.
See the “Configuring Backup Registrar Service to SIP Phones (Using Optional Commands)” section on
page 164 for these commands.

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Configuring Backup Registrar Service to SIP Phones (Using Optional

Commands)
The prior configurations set up a basic voice register pool. The configuration in this procedure adds
optional attributes to increase functionality.
Prerequisites
• Prerequisites as described in the “Configuring Backup Registrar Service to SIP Phones” section on
page 160.
• Configuration of the required commands as described in the “Configuring Backup Registrar Service
to SIP Phones” section on page 160.
• Before configuring the 'alias' command, translation rules must be set using the translate-outgoing
(voice register pool) command.
SUMMARY STEPS
1. enable
4. translation-profile outgoing profile-tag
5. alias tag pattern to target [preference value]
6. cor {incoming | outgoing} cor-list-name {cor-list-number starting-number [- ending-number] |
default}
7. incoming called-number [number]
8. number tag number-pattern {preference value} [huntstop]
9. dtmf-relay [cisco-rtp] [rtp-nte] [sip-notify]
10. end
DETAILED STEPS

Example:
Router> enable
Example:

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Step 4 translation-profile outgoing profile-tag Use this command to apply the translation profile to a
specific directory number or to all directory numbers on a
SIP phone.
Example:
Router(config-register-pool)# • Profile-tag: Translation profile name to handle
voice translation-rule 1 translation to outgoing calls.
rule 1 /1000/ /1006/
!
!
voice translation-profile 1
translate called 1
!
voice register pool xxx
translation-profile outgoing 1
Step 5 alias tag pattern to target [preference value] Allows Cisco Unified SIP IP Phones to handle inbound
PSTN calls to telephone numbers that are unavailable when
the main proxy is not available. The keywords and
Example:
Router(config-register-pool)# alias 1 94... to
91011 preference 8 • tag: Number from 1 to 5 and the distinguishing factor
when there are multiple alias commands.
• pattern: The prefix number; matches the incoming
telephone number and may include wildcards.
number.
• target: The target number; an alternate telephone
number to route incoming calls to match the number
pattern.
• preference value: (Optional) Assigns a dial-peer
preference value to the alias. The value argument is the
value of the associated dial peer, and the range is from
1 to 10. There is no default.

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Step 6 cor {incoming | outgoing} cor-list-name Configures a class of restriction (COR) on the VoIP dial
{cor-list-number starting-number [- peers associated with directory numbers. COR specifies
ending-number] | default}
which incoming dial peers can use which outgoing dial
peers to make a call. Each dial peer can be provisioned with
Example: an incoming and outgoing COR list. The keywords and
Router(config-register-pool)# cor incoming arguments are defined as follows:
call91 1 91011
• outgoing: COR list to be used by outgoing dial peers.
• cor-list-name: COR list name.
number of COR lists that can be created is four,
comprised of incoming or outgoing dial peers.
• starting-number: Start of a directory number range, if
an ending number is included. Can also be a standalone
number.
• (Optional) Indicator that a full range is configured.
• ending-number: (Optional) End of a directory number
range.
COR list.
Step 7 incoming called-number [number] Applies incoming called parameters to dynamically created
dial peers. The number argument is optional and indicates a
sequence of digits that represent a phone number prefix.
Example:
Router(config-register-pool)# incoming
called-number 308
Step 8 number tag number-pattern {preference value} Indicates the E.164 phone numbers that the registrar permits
[huntstop] to handle the Register message from the Cisco Unified SIP
IP Phone. The keywords and arguments are defined as
Example: follows:
Router(config-register-pool)# number 1 50.. • tag: Number from 1 to 10 and the distinguishing factor
preference 2
when there are multiple number commands.
• number-pattern: Phone numbers (including wildcards
and patterns) that are permitted by the registrar to
handle the Register message from the SIP IP phone.
• preference value: (Optional) Defines the number list
preference order.
• huntstop: (Optional) Stops hunting if the dial peer is
busy.

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Step 9 dtmf-relay [cisco-rtp] [rtp-nte] [sip-notify] Specifies how a SIP gateway relays dual tone
multifrequency (DTMF) tones between telephony
interfaces and an IP network. The keywords are defined as
Example:
follows:
rtp-nte • cisco-rtp: (Optional) Forwards DTMF tones by using
Real-Time Transport Protocol (RTP) with a Cisco
proprietary payload type.
• rtp-nte: (Optional) Forwards DTMF tones by using
RTP with the Named Telephone Event (NTE) payload
type.
• sip-notify: (Optional) Forwards DTMF tones using SIP
NOTIFY messages.
Example:
Examples
The following partial output from the show running-config command shows that voice register pool 12
is configured to accept all registrations from SIP IP phones with extension number 50xx from the
172.16.0.0/16 network. Autogenerated dial peers for registrations that match pool 12 have attributes
configured in this pool.
.
.
.
id network 172.16.0.0 mask 255.255.0.0
number 1 50.. preference 2
application SIP.app
preference 2
incoming called-number
cor incoming allowall default
translate-outgoing called 1
voice-class codec 1
.
.
.
Verifying SIP Registrar Configuration

To help you troubleshoot a SIP registrar and voice register pool, perform the following steps.
SUMMARY STEPS
1. debug voice register errors

2. debug voice register events
3. show sip-ua status registrar

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DETAILED STEPS

Step 1 debug voice register errors Use this command to debug errors that happen
during registration.
Example: If there are no voice register pools configured for a
Router# debug voice register errors particular registration request, the message “Contact
*Apr 22 11:52:54.523 PDT: VOICE_REG_POOL: Contact doesn’t match any pools” is displayed.
doesn't match any pools
request for (33015) from (10.2.152.39)
*Apr 22 11:52:54.539 PDT: VOICE_REG_POOL: Contact
doesn't match any pools.
request for (33017) from (10.2.152.39)
*Apr 22 11:53:04.559 PDT: VOICE_REG_POOL: Maximum
registration threshold for pool(3) hit
Step 2 debug voice register events Using the debug voice register events command
should suffice to display registration activity.
Registration activity includes matching of pools,
Example:
Router# debug voice register events
registration creation, and automatic creation of dial
Apr 22 10:50:21.731 PDT: VOICE_REG_POOL: Contact peers. For more details and error conditions, you can
matches pool 1 use the debug voice register errors command.
Apr 22 10:50:21.731 PDT: VOICE_REG_POOL: key(91011)
contact(192.168.0.2) add to contact table The phone number 91011 registered successfully,
Apr 22 10:50:21.731 PDT: VOICE_REG_POOL: key(91011) and type 1 is reported, which means there is a
exists in contact table pre-existing VoIP dial peer.
contact(192.168.0.2) exists in contact table, ref
updated
Apr 22 10:50:21.731 PDT: VOICE_REG_POOL: Created
dial-peer entry of type 1
Registration successful for 91011, registration id
is 257
Step 3 show sip-ua status registrar Use this command to display all the SIP endpoints
currently registered with the contact address.
Example:
======= =========== ============ =======
91021 192.168.0.3 227 192.168.0.3
91011 192.168.0.2 176 192.168.0.2
95021 10.2.161.50 419 10.2.161.50
95012 10.2.161.50 419 10.2.161.50
95011 10.2.161.50 420 10.2.161.50
95500 10.2.161.50 420 10.2.161.50
94011 10.2.161.40 128 10.2.161.40
94500 10.2.161.40 129 10.2.161.40

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Verifying Proxy Dial-Peer Configuration

To use the icmp-ping keyword with the proxy command to assist in troubleshooting proxy dial peers,
SUMMARY STEPS
4. end
5. show voice register dial-peers
6. show dial-peer voice

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DETAILED STEPS

Step 1 configure terminal Use this command to enter global configuration
mode.
Example:
Step 2 voice register pool tag Use this command to enter voice register pool
configuration mode.
Example:
Step 3 proxy ip-address [preference value] [monitor probe Set the proxy command to monitor with icmp-ping.
{icmp-ping | rtr} [alternate-ip-address]]
Example:
Router(config-register-pool)# proxy 10.2.161.187
preference 1 monitor probe icmp-ping
Example:
Step 5 show voice register dial-peers Use this command to verify dial-peer
configurations, and notice that icmp-ping
monitoring is set.
Example:
preference 5
voice-class codec 1

preference 1
voice-class codec 1
monitor probe icmp-ping 10.2.161.187

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Step 6 show dial-peer voice Use the show dial-peer voice command on dial peer
40036, and notice the monitor probe status.
Example:
Router# show dial-peer voice
VoiceOverIpPeer40036 Note Also highlighted is the output of the cor and
peer type = voice, information type = voice, incoming called-number commands.
description = `',
tag = 40036, destination-pattern = `91011',
answer-address = `', preference=1,
CLID Restriction = None
CLID Network Number = `'
CLID Second Number sent
source carrier-id = `', target carrier-id = `',
source trunk-group-label = `', target
trunk-group-label = `',
numbering Type = ùnknown'
group = 40036, Admin state is up, Operation state is
up,
incoming called-number = `', connections/maximum =
0/unlimited,
! Default output for incoming called-number command
DTMF Relay = disabled,
modem transport = system,
huntstop = disabled,
in bound application associated: 'DEFAULT'
out bound application associated: ''
dnis-map =
permission :both
incoming COR list:maximum capability
outgoing COR list:minimum requirement
Translation profile (Incoming):
Translation profile (Outgoing):
incoming call blocking:
translation-profile = `'
disconnect-cause = `no-service'
advertise 0x40 capacity_update_timer 25 addrFamily 4
oldAddrFamily 4
type = voip, session-target = ìpv4:10.2.161.187',
technology prefix:
settle-call = disabled
ip media DSCP = ef, ip signaling DSCP = af31,
ip video rsvp-none DSCP = af41,ip video rsvp-pass
DSCP = af41
ip video rsvp-fail DSCP = af41,
UDP checksum = disabled,
session-protocol = sipv2, session-transport =
system,
req-qos = best-effort, acc-qos = best-effort,
req-qos video = best-effort, acc-qos video =
best-effort,
req-qos audio def bandwidth = 64, req-qos audio max
bandwidth = 0,
req-qos video def bandwidth = 384, req-qos video max
bandwidth = 0,

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Where to Go Next

RTP dynamic payload type values: NTE = 101
Cisco: NSE=100, fax=96, fax-ack=97, dtmf=121,
fax-relay=122
S=123, ClearChan=125, PCM switch over
u-law=0,A-law=8
RTP comfort noise payload type = 19
fax rate = voice, payload size = 20 bytes
fax protocol = system
fax-relay ecm enable
fax NSF = 0xAD0051 (default)
codec = g729r8, payload size = 20 bytes,
Media Setting = flow-through (global)
Expect factor = 0, Icpif = 20,
Playout Mode is set to adaptive,
Initial 60 ms, Max 300 ms
Playout-delay Minimum mode is set to default, value
40 ms
Fax nominal 300 ms
Max Redirects = 1, signaling-type = cas,
VAD = enabled, Poor QOV Trap = disabled,
Source Interface = NONE
voice class sip url = system,
voice class sip rel1xx = system,
monitor probe method: icmp-ping ip address:
10.2.161.187,
Monitored destination reachable
voice class perm tag = `'
Time elapsed since last clearing of voice call
statistics never
Connect Time = 0, Charged Units = 0,
Successful Calls = 0, Failed Calls = 0, Incomplete
Calls = 0
Accepted Calls = 0, Refused Calls = 0,
Last Disconnect Cause is "",
Last Disconnect Text is "",
Last Setup Time = 0.
Where to Go Next
The next step is configuring incoming and outgoing calls for Cisco Unified SRST. For more information,
see the “Configuring Call Handling” section on page 175.

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Where to Go Next

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Where to Go Next

7-174
CHAPTER 8
Configuring Call Handling
This chapter describes how to configure Cisco Unified Survivable Remote Site Telephony (Cisco Unified
SRST) for incoming and outgoing calls for SCCP phones.
This chapter also describes support for standardized RFC 3261 features for SIP phones. Features include
call blocking and call forwarding.
Note Configuring Call Handling for SIP phones applies to versions 4.0 and 3.4 only.
Contents
• Prerequisites for Configuring SIP SRST Features Using Back-to-Back User Agent Mode, page 176
• Restrictions for Configuring SIP SRST Features Using Back-to-Back User Agent Mode, page 176
• Information About Configuring SCCP SRST Call Handling, page 176
• Information About Configuring SIP SRST Features Using Back-to-Back User Agent Mode,
page 177
• How to Configure Cisco Unified SCCP SRST, page 179
• How to Configure Cisco Unified SIP SRST, page 216
• How to Configure Optional Features, page 226
• Configuration Examples for Call Handling, page 228

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Chapter 8 Configuring Call Handling
Prerequisites for Configuring SIP SRST Features Using Back-to-Back User Agent Mode
Prerequisites for Configuring SIP SRST Features Using

Back-to-Back User Agent Mode
• Complete the prerequisites documented in the “Prerequisites for Configuring Cisco Unified SIP
SRST” section on page 9 section in the “Cisco Unified SRST Feature Overview” section on page 1.
• Configure the SIP registrar. The SIP registrar gives users control of accepting or rejecting
registrations. To configure acceptance of incoming SIP Register messages, see the “” section on
page 157.
Restrictions for Configuring SIP SRST Features Using

• See the restrictions documented in the “Restrictions for Configuring Cisco Unified SIP SRST”
section on page 9 section in the “Cisco Unified SRST Feature Overview” section on page 1.
Information About Configuring SCCP SRST Call Handling

Cisco Unified SRST offers a smaller set of call handling capabilities than Cisco Unified CM, and much
of the configuration for these feature involves enabling existing Cisco Unified CM or Cisco Unified IP
Phone settings.
• H.323 VoIP Call Preservation Enhancements for WAN Link Failures, page 176
• Toll Fraud Prevention, page 177
H.323 VoIP Call Preservation Enhancements for WAN Link Failures

H.323 VoIP call preservation enhancements for WAN link failures sustains connectivity for H.323
topologies where signaling is handled by an entity, such as Cisco Unified Communications Manager,
that is different from the other endpoint and brokers signaling between the two connected parties.
Call preservation is useful when a gateway and the other endpoint (typically a Cisco Unified IP phone)
are collocated at the same site and call agent is remote and therefore more likely to experience
connectivity failures.
For configuration information see “Configuring H.323 Gateways” chapter in Cisco IOS H.323
Configuration Guide, Release 12.4T.

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Information About Configuring SIP SRST Features Using Back-to-Back User Agent Mode
Toll Fraud Prevention

When a Cisco router platform is installed with a voice-capable Cisco IOS software image, appropriate
features must be enabled on the platform to prevent potential toll fraud exploitation by unauthorized
users. Deploy these features on all Cisco router Unified Communications applications that process voice
calls, such as Cisco Unified Communications Manager Express (CME), Cisco Survivable Remote Site
Telephony (SRST), Cisco Unified Border Element (UBE), Cisco IOS-based router and standalone
analog and digital PBX and public-switched telephone network (PSTN) gateways, and Cisco
contact-center VoiceXML gateways. For more information about Toll Fraud Prevention, see Toll Fraud
Prevention in Cisco Unified Communications Manager Express System Administration Guide.
Information About Configuring SIP SRST Features Using

A Cisco Unified SRST system can now support SIP phones with standard-based RFC 3261 feature
support locally and across SIP WAN networks. With Cisco Unified SIP SRST, SIP phones can place calls
across SIP networks with similar features, as SCCP phones do. For example, most SCCP phone features
such as caller ID, speed dial, and redial are supported now on SIP networks, which gives users the
opportunity to choose SCCP or SIP.
Cisco Unified SIP SRST also uses a back-to-back user agent (B2BUA), which is a separate call agent
that has more features than Cisco SIP SRST 3.0, which used a redirect server that only accepted and
forwarded calls. The main advantage of a B2BUA call agent is in call forwarding, because it forwards
calls on behalf of the phone. In addition, it maintains a presence as call middleman in the call path.
Cisco SIP SRST 3.4 supports the following call combinations:
• SIP phone to PSTN / router voice port
• SIP phone to SCCP phone
Cisco Unified SIP SRST and Cisco SIP Communications Manager Express
Feature Crossover
The voice regisiter dn, voice register global and voice register pool configuration mode commands are
accessible in both Cisco Unified SIP CME and Cisco Unified SIP SRST modes of operation. However,
not all of the commands within these modes are intended for use in SIP SRST mode. Table 8-1 provides
a summary guide to which commands are relevant to the CME or SRST modes of operation.
For more detailed information, refer to the command reference pages for each of the individual
commands.
Note Table 8-1 is not all-inclusive; additional commands may exist.

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Information About Configuring SIP SRST Features Using Back-to-Back User Agent Mode
Table 8-1 Version 3.4 New or Enhanced Commands for Cisco Unified SRST and Cisco Unified CME (Sorted
by Configuration Mode)
Configurable for
Voice Cisco Unified (SIP ) CME Applicable to
Dial Register and Cisco Unified SIP Cisco Unified (SIP) CME
Command Peer Mode SRST Only
after-hour exempt X dn X —
auto-answer — dn — X
call forward X dn X —
huntstop X dn X —
label — dn — X
name — dn — X
number X dn X —
preference X dn X —
application X global X —
authenticate — global — X
create — global — X
date-format — global — X
dst — global — X
external ring — global X —
file — global — X
hold-alert — global — X
load — global — X
logo — global — X
max-dn — global X —
max-pool — global X —
max-redirect — global — X
mode — global X —
mwi — global — X
reset — global — X
tftp-path — global — X
timezone — global — X
upgrade — global — X
url — global — X
voicemail — global — X
after-hour exempt X pool X —
application X pool X —
call-forward — pool X —
call-waiting — pool — X

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How to Configure Cisco Unified SCCP SRST
Table 8-1 Version 3.4 New or Enhanced Commands for Cisco Unified SRST and Cisco Unified CME (Sorted
by Configuration Mode) (continued)
Configurable for
Voice Cisco Unified (SIP ) CME Applicable to
Dial Register and Cisco Unified SIP Cisco Unified (SIP) CME
Command Peer Mode SRST Only
codec X pool X —
description — pool — X
dnd-control — pool — X
dtmf-relay — pool X —
id — pool X —
keep-conference — pool — X
max-pool — pool X —
number X pool X —
preference X pool X —
proxy X pool X —
reset — pool — X
speed-dial — pool — X
template — pool — X
translation-profile X pool X —
type — pool — X
username — pool — X
vad X pool X —
anonymous — template — X
caller-id — template — X
conference — template — X
dnd-control — template — X
forward — template — X
transfer — template — X

Setting up call handling involves the following set of tasks:
• Configuring Incoming Calls, page 180
• Configuring Outgoing Calls, page 199
• Configuring Call Blocking Based on Time of Day, Day of Week, or Date, page 220

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Configuring Incoming Calls

Incoming call configuration can include the following tasks:
• Call Forwarding and Rerouting
– Configuring Call Forwarding During a Busy Signal or No Answer, page 180 (Optional)
– Configuring Call Rerouting, page 182 (Optional)
– Configuring Call Pickup, page 185 (Optional)
– Configuring Transfer Digit Collection Method, page 188
• Phone Number Conversion and Translation
– Configuring Global Prefixes, page 189 (Optional)
– Enabling Digit Translation Rules, page 191 (Optional)
– Enabling Translation Profiles, page 192 (Optional)
– Verifying Translation Profiles, page 195 (Optional)
• Hunting and Ringing Timeout Behavior
– Configuring Dial-Peer and Channel Hunting, page 196 (Optional)
– Configuring Busy Timeout, page 197 (Optional)
– Configuring the Ringing Timeout Default, page 198 (Optional)
Configuring Call Forwarding During a Busy Signal or No Answer

Incoming calls that reach a busy signal or go unanswered during Cisco Unified CM fallback can be
configured to be forwarded to one or more E.164 numbers.
SUMMARY STEPS
2. call-forward busy directory-number
3. call-forward noan directory-number timeout seconds
4. exit

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DETAILED STEPS

Example:
Step 2 call-forward busy directory-number Configures call forwarding to another number when the
Cisco IP phone is busy.
Example: • directory-number: Selected directory number
Router(config-cm-fallback)# call-forward busy representing a fully qualified E.164 number. This
50.. number can contain “.” wildcard characters that
correspond to the right-justified digits in the directory
number extension.
Step 3 call-forward noan directory-number timeout Configures call forwarding to another number when no
seconds answer is received from the Cisco IP phone.
• directory-number: Selected directory number
Example: representing a fully qualified E.164 number or a local
Router(config-cm-fallback)# call-forward noan extension number. This number can contain “.”
5005 timeout 10
wildcard characters that correspond to the
right-justified digits in the directory number extension.
• timeout seconds: Sets the waiting time, in seconds,
before the call is forwarded to another phone. The
seconds range is from 3 to 60000.
Example:
Examples
The following example forwards calls to extension number 5005 when an incoming call reaches a busy
or unattended IP phone extension number. Incoming calls will ring for 15 seconds before being
forwarded to extension 5005.
call-forward noan 5005 timeout seconds 15
The following example transforms an extension number for call forwarding when the extension number
is busy or unattended. The call-forward busy command has an argument of 50.., which prepends the
digits 50 to the last two digits of the called extension. The resulting extension is the number to which
incoming calls are forwarded when the original extension number is busy or unattended. For instance,
an incoming call to the busy extension 6002 will be forwarded to extension 5002, and an incoming call
to the busy extension 3442 will be forwarded to extension 5042. Incoming calls will ring for 15 seconds
before being forwarded.
call-forward busy 50..
call-forward noan 50.. timeout seconds 15

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Configuring Call Rerouting
Note We recommend the alias command, which obsoletes the default-destination command, instead of the
default-destination command.
The alias command provides a mechanism for rerouting calls to telephone numbers that are unavailable
during fallback. Up to 50 sets of rerouting alias rules can be created for calls to telephone numbers that
are unavailable during Cisco Unified Communications Manager fallback. Sets of alias rules are created
using the alias command. An alias is activated when a telephone registers that has a phone number
matching a configured alternate-number alias. Under that condition, an incoming call is rerouted to the
alternate number. The alternate-number argument can be used in multiple alias commands, allowing you
to reroute multiple different numbers to the same target number.
The configured alternate-number must be a specific E.164 phone number or extension that belongs to
an IP phone registered on the Cisco Unified SRST router. When an IP phone registers with a number that
matches an alternate-number, an additional POTS dial peer is created. The destination pattern is set to
the initial configured number-pattern, and the POTS dial peer voice port is set to match the voice port
associated with the alternate-number.
If other IP phones register with specific phone numbers within the range of the initial number-pattern,
the call is routed back to the IP phone rather than to the alternate-number (according to normal dial-peer
longest-match, preference, and huntstop rules).
Call Forward Destination
The cfw keyword allows you to configure a call forward destination for calls that are busy or not
answered. Call forward no answer is defined as when the phone rings for a user configurable amount of
time, the call is not answered, and is forwarded to the configured destination. Call forward busy and call
forward no answer can be configured to a set string and override globally configured call forward
settings.
Note Globally configured settings are selected under call-manager-fallback and apply to all phones that
register for SRST service.
You can also create a specific call forwarding path for a particular number. The benefit of using the cfw
keyword is that during SRST, you can reroute calls from otherwise unreachable numbers onto phones
that are available. Basic hunt groups can be established with call-forwarding rules so that if the first
SRST phone is busy, you can forward the call to a second SRST phone.
The cfw keyword also allows you to alias a phone number to itself, permitting setting of per-phone
number forwarding. An example of aliasing a number to itself follows. If a phone registers with
extension 1001, a dial peer that routes calls to the phone is automatically created for 1001. If the
call-manager-fallback dial-peer preference (set with the max-dn command) for this initial dial peer is
set to 2, the dial peer uses 2 as its preference setting.
Then, use the alias command to alias the phone number to itself:
alias 1 1001 to 1001 preference 1 cfw 2001 timeout 20

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In this example, you have created a second dial peer for 1001 to route calls to 1001, but that has
preference 1 and call forwarding to 2001. Because the preference on the dial peer created by the alias
command is now a lower numeric value than the preference that the dial peer first created, all calls come
initially to the dial peer created by the alias command. In that way, they are subject to the forward as set
by the alias command, instead of any call forwarding that may have been set globally.
Huntstop on an Individual Alias
The alias huntstop keyword is relevant only if you have also set the global no huntstop command under
call-manager-fallback. Also, you may need to set the global no huntstop if you have multiple alias
commands with the same number-pattern and you want to enable hunting on busy between the aliases.
That is, one alias for number-pattern is tried, and then if that phone is busy, the second alias for
number-pattern is tried.
The alias huntstop keyword allows you to turn huntstop behavior back on for an individual alias, if
huntstop is turned off globally by the no huntstop command. Setting the huntstop keyword on an
individual alias stops hunting at the alias, making the alias the final member of the hunt sequence.
SUMMARY STEPS
2. alias tag number-pattern to alternate-number [preference preference-value] [cfw number timeout
timeout-value] [huntstop]
4. end
5. show dial-peer voice summary

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DETAILED STEPS

Example:
Step 2 alias tag number-pattern to alternate-number Creates a set rules for rerouting calls to sets of phones that
[preference preference-value] [cfw number are unavailable during Cisco Unified CM fallback.
timeout timeout-value] [huntstop]
• tag: Identifier for alias rule range. The range is from 1
to 50.
Example:
Router(config-cm-fallback)# alias 1 60.. to • number-pattern: Pattern to match the incoming
5001 preference 1 cfw 2000 timeout 10 telephone number. This pattern may include wildcards.
number.
• alternate-number: Alternate telephone number to
route incoming calls to match the number pattern. The
alternate number has to be a specific extension that
belongs to an IP phone that is actively registered on the
Cisco Unified SRST router. The alternate telephone
number can be used in multiple alias commands.
• preference preference-value (Optional). Assigns a
dial-peer preference value to the alias. The preference
value of the associated dial peer is from 0 to 10. Use
with the max-dn command.
• cfw number (Optional). The cfw keyword allows users
to set call forward busy and call forward no answer to
a set string and override globally configured call
forward settings.
• timeout timeout-value (Optional). Sets the ring
no-answer timeout duration for call forwarding, in
seconds. Range is from 3 to 60000.
• huntstop (Optional). Stops call hunting after trying
the alternate number.
Step 3 max-dn max-directory-numbers [dual-line] Sets the maximum possible number of directory numbers
[preference preference-order] or virtual voice ports that can be supported by a router and
sets the global preference for creating the VoIP dial peers
Example: for all directory numbers that are associated with the
Router(config-cm-fallback)# max-dn 10 primary number.
preference 2
• Using the max-dn command sets the preference for the
default dial peers created with the alias command.
• When configuring call rerouting, set the max-dn
preference to a higher numeric preference than the
preference that was set with the alias command.

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Example:
Step 5 show dial-peer voice summary Displays information for voice dial peers.
• If you suspect a problem with the dial peers, use this
Example: command to display the dial peers created by the alias
Router# show dial-peer voice summary command.
Examples
The following example sets the preference keyword in the alias command to a lower preference value
that the preference value created by the max-dn command. Setting the value lower allows the cfw
keyword to take effect. The incoming call to extension 1000 hunts to alias because it has a lower
preference, and no-answer/busy calls to 1000 are forwarded to 2000. All incoming calls to other
extensions in SRST mode are forwarded to 3000 after 10 seconds.
alias 1 1000 to 1000 preference 1 cfw 2000 timeout 10
max-dn 10 preference 2
Configuring Call Pickup

Configuring the pickup command enables the PickUp soft key on all SRST phones. You can then press
the PickUp key and answer any currently ringing IP phone that has a DID called number that matches
the configured telephone-number. This command does not enable the Group PickUp (GPickUp) soft key.
When a user presses the PickUp soft key, SRST searches through all the SRST phones to find a ringing
call that has a called number that matches the configured telephone-number. When a match is found, the
call is automatically forwarded to the extension number of the phone that requested the call pickup.
The SRST pickup command is designed to operate in a manner compatible with
Cisco Unified Communications Manager.
Note The default phone load on Cisco Unified Communications Manager, Release 4.0(1) for the Cisco 7905
and Cisco 7912 IP phones does not enable the PickUp soft key during fallback. To enable the PickUp
soft key on Cisco 7905 and Cisco 7912 IP phones, upgrade your default phone load to
Cisco Unified CM, Version 4.0(1) Sr2. Alternatively, you can upgrade the phone load to
cmterm-7905g-sccp.3-3-8.exe or cmterm-7912g-sccp.3-3-8.exe, respectively.
SUMMARY STEPS
2. no huntstop
3. alias tag number-pattern to alternate-number
4. pickup telephone-number
5. end

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DETAILED STEPS

Example:
Step 2 no huntstop Disables huntstop.
Example:
Router(config-cm-fallback)# no huntstop
Step 3 alias tag number-pattern to alternate-number Creates a set rules for rerouting calls to sets of phones that
are unavailable during Cisco Unified CM fallback.
Example: • tag: Identifier for alias rule range. The range is from 1
Router(config-cm-fallback)# alias 1 8005550100 to 50.
to 5001
• number-pattern: Pattern to match the incoming
telephone number. This pattern may include wildcards.
number.
• alternate-number: Alternate telephone number to
route incoming calls to match the number pattern. The
alternate number has to be a specific extension that
belongs to an IP phone that is actively registered on the
Cisco Unified SRST router. The alternate telephone
number can be used in multiple alias commands.
Step 4 pickup telephone-number Enables the PickUp soft key on all Cisco Unified IP
Phones, allowing an external Direct Inward Dialing (DID)
call coming into one extension to be picked up from
Example:
Router(config-cm-fallback)# pickup 8005550100
another extension during SRST. The telephone-number
argument is the telephone number to match an incoming
called number.
Example:
Examples
The pickup command is best used with the alias command. The following partial output from the show
running-config command shows the pickup command and the alias command configured to provide
call routing for a pilot number of a hunt group:
no huntstop
alias 1 8005550100 to 5001
alias 2 8005550100 to 5002
alias 3 8005550100 to 5003
alias 4 8005550100 to 5004
pickup 8005550100

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When a DID incoming call to 800 555-0100 is received, the alias command routes the call at random to
one of the four extensions (5001 to 5004). Because the pickup command is configured, if the DID call
rings on extension 5002, the call can be answered from any of the other extensions (5001, 5003, 5004)
by pressing the PickUp soft key.
The pickup command works by finding a match based on the incoming DID called number. In this
example, a call from extension 5004 to extension 5001 (an internal call) does not activate the pickup
command because the called number (5001) does not match the configured pickup number (800
555-0100). Thus, the pickup command distinguishes between internal and external calls if multiple calls
are ringing simultaneously.
Configuring Consultative Transfer

Before Cisco Unified SRST 4.3, the consultative transfer feature played a dial tone and collected dialed
digits until the digits matched the pattern for consultative transfer, blind transfer, or PSTN transfer
blocking. The after-hours blocking criteria was applied after the consultative transfer digit collection and
pattern matching.
The new feature modifies the transfer digit-collection process to make it consistent with Cisco Unified
Communications Manager. This feature is supported only if the transfer-system full-consult command
(default) is specified in call-manager-fallback configuration mode and an idle line or channel is available
for seizing, digit collection, and dialing.
Two lines are required for a consultative transfer. When the transferor party is an octo-line directory
number, Cisco Unified SRST selects the next available idle channel on that directory number. If the
maximum number of channels of the directory number are in use, another idle line on the transferor
phone is considered. If the auto-line command is configured on the phone, the specified autoline (if idle)
takes precedence over other nonauto lines. If no idle line is available on the transferor phone, a blind
transfer is initiated instead of the consultative transfer.
During the consultative transfer, the transferor line to the transferee party is locked on the transferor
phone to prevent it from being stolen by other phones sharing the same directory number. When the user
presses the Transfer soft key for a consultative transfer, the Transfer soft key does not display while
digits are being dialed and collected on this seized consultative transfer call leg. The method for
consultative transfer pattern matching, blind transfer, PSTN transfer blocking, or after-hour blocking
criteria remain the same although the manipulation after the matching is different. When the criteria for
the blind transfer is met, Cisco Unified SMST terminates the consultative transfer call leg, informs the
Cisco IOS software to transfer the call, and then terminates the original call bubble. The PARK FAC code
is handled in the same way as by a new call which requires that a ten-second timer is applied by the Cisco
IOS software.
Note The enhancement, by default, collects the transfer digits from the new call leg. If required, you can
configure the system to collect the transfer digits from the original call leg. See the “Configuring
Transfer Digit Collection Method” section on page 188.
The error handling for transfer failure because of transfer blocking or interdigit timer expiration remains.
It includes displaying an error message on the prompt line and logging it if “debug ephone error” is
enabled, playing a fast-busy or busy tone, and terminating the consultative transfer call leg.
No new configuration is required to support these enhancements.

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Conference Calls
No configuration steps are required for these conference call enhancements.
Single-Line Directory Number

If the initiating party for the conference call is a single-line directory number and the phone has multiple
directory numbers configured, the system selects another directory number’s idle channel for creating
the conference. If there are multiple directory numbers (dual-line or single-line directory numbers) on
the phone and each has calls on hold, the system prompts the user to select a line for the conference call.
Dual-Line Directory Number

If the initiating party for the conference call is a dual-line directory number, the system selects another
idle channel from the dual-line directory number. If the selected channel has a call on hold, the
conference operation will automatically select the hold channel and create the conference.
Octo-Line Directory Number

If the initiating party for the conference call is an octo-line directory number, the system selects an idle
channel from the initiating party directory number and the user must establish a new call to complete the
conference. If there is no idle channel on the same directory number, other idle directory numbers or
channels on the same phone are not selected. If there are existing calls on hold on the other channels of
the same directory number or other directory numbers, the user will not have the option to select them
to join the conference. If there is no idle channel on the same directory number, the conference will abort
with a No Line Available message.
Configuring Transfer Digit Collection Method

By default, transfer digits are collected from the new call leg. To change the transfer digit collection
method, perform the following steps.
Prerequisites for Cisco Unified SRST 4.3
• Cisco Unified SRST 4.3

• Cisco Unified CM 6.0
• Cisco IOS Release 12.4(15)XZ
Restrictions for Cisco Unified SRST 4.3
• The Cisco 3200 Series Mobile Access Router does not support SRST.
SUMMARY STEPS
1. enable
4. transfer-digit-collect {new-call | orig-call}
5. end

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DETAILED STEPS

Example:
Router> enable
Example:
Example:
Step 4 transfer-digit-collect {new-call | orig-call} Selects the digit-collection method used for consultative
call transfers.
Example: • new-call: Digits are collected from the new call leg.
Router(config-cm-fallback)#
transfer-digit-collect orig-call
• orig-call: Digits are collected from the original
call-leg. This was the default behavior in versions
before Cisco Unified SRST 4.3.
Example:
Router(config)# end
Examples
The following example shows the transfer-digit-collect method set to the legacy value of orig-call:
!
transfer-digit collect orig-call
!
Configuring Global Prefixes

The dialplan-pattern command creates a dial-plan pattern that specifies a global prefix for the
expansion of abbreviated extension numbers into fully qualified E.164 numbers.
The extension-pattern keyword allows additional manipulation of abbreviated extension-number prefix
digits. When this keyword and its argument are used, the leading digits of an extension pattern are
stripped and replaced by the corresponding leading digits of the dial-plan pattern. This command can be
used to avoid Direct Inward Dialing (DID) numbers like 408 555-0101 resulting in 4-digit extensions
such as 0101.
Global prefixes are set with the dialplan-pattern command. Up to five dial-plan patterns can be created.
The no-reg keyword provides dialing flexibility and prevents the E.164 numbers in the dial peer from
registering to the gatekeeper. You have the option not to register numbers to the gatekeeper so that those
numbers can be used for other telephony services.

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SUMMARY STEPS
2. dialplan-pattern tag pattern extension-length length [extension-pattern extension-pattern]
[no-reg]
3. exit
DETAILED STEPS

Example:
Step 2 dialplan-pattern tag pattern extension-length Creates a global prefix that can be used to expand the
length [extension-pattern extension-pattern] abbreviated extension numbers into fully qualified E.164
[no-reg]
numbers
• tag: Dial-plan string tag used before a 10-digit
Example: telephone number. The tag number is from 1 to 5.
Router(config-cm-fallback)# dialplan-pattern 1
4085550100 extension-length 3 extension-pattern • pattern: Dial-plan pattern, such as the area code, the
4.. prefix, and the first one or two digits of the extension
number, plus wildcard markers or dots (.) for the
Note This example maps all extension numbers 4xx remainder of the extension number digits.
to the PSTN number 40855501xx, so that
extension 412 corresponds to 4085550112. • extension-length: Sets the number of extension digits.
• length: The number of extension digits. The range is
from 1 to 32.
• extension-pattern: (Optional) Sets an extension
number’s leading digit pattern when it is different from
the E.164 telephone number’s leading digits defined in
the pattern argument.
• extension-pattern: (Optional) The extension number’s
leading digit pattern. Consists of one or more digits
and wildcard markers or dots (.). For example, 5..
would include extension 500 to 599; 5... would include
5000 to 5999.
• no-reg: (Optional) Prevents the E.164 numbers in the
dial peer from registering with the gatekeeper.
Example:

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Examples
The following example shows how to create dial-plan pattern 1 for extension numbers 101 to 199 with
the telephone prefix starting with 4085550. If the following example is set, the router will recognize that
4085550144 matches dial-plan pattern 1. It will use the extension-length keyword to extract the last
three digits of the number 144 and present this as the caller ID for the incoming call.
dialplan-pattern 1 40855501.. extension-length 3 no-reg
In the following example, the leading prefix digit for the 3-digit extension numbers is transformed from
0 to 4, so that the extension-number range becomes 400 to 499:
dialplan-pattern 1 40855500.. extension-length 3 extension-pattern 4..
In the following example, the dialplan-pattern command creates dial-plan pattern 2 for extensions 801
to 899 with the telephone prefix starting with 4085559. As each number in the extension pattern is
declared with the number command, two POTS dial peers are created. In the example, they are 801 (an
internal office number) and 4085559001 (an external number).
dialplan-pattern 2 40855590.. extension-length 3 extension-pattern 8..
Enabling Digit Translation Rules

Digit translation rules can be enabled during Cisco Unified CM fallback. Translation rules are a
number-manipulation mechanism that performs operations such as automatically adding telephone area
codes and prefix codes to dialed numbers.
Note Digit translation rules have many applications and variations. For further information about them, see
Cisco IOS Voice Configuration Library.
If you are running Cisco SRST 3.2 and later or Cisco Unified SRST 4.0 and later, use the configuration
described in the “Enabling Translation Profiles” section on page 192 instead of using the translate
command as described below. Translation Profiles are new to Cisco SRST 3.2 and provide added
capabilities.
Translation rules can be used as follows:

• To manipulate the answer number indication (ANI) (calling number) or dialed number identification
service (DNIS) (called number) digits for a voice call.
• To convert a telephone number into a different number before the call is matched to an inbound dial
peer or before the call is forwarded by the outbound dial peer.
To view the translation rules configured for your system, use the show translation-rule command.
SUMMARY STEPS
2. translate {called | calling} translation-rule-tag
3. exit

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DETAILED STEPS

Example:
Step 2 translate {called | calling} Applies a translation rule to modify the phone number
translation-rule-tag dialed or received by any Cisco Unified IP Phone user
while Cisco Unified CM fallback is active.
Example: • called: Applies the translation rule to an outbound call
Router(config-cm-fallback)# translate called 20 number.
• calling: Applies the translation rule to an inbound call
number.
• translation-rule-tag: The reference number of the
translation rule from 1 to 2147483647.
Example:
Examples
The following example applies translation rule 10 to the calls coming into extension 1111. All inbound
calls to 1111 will go to 2222 during Cisco Unified CM fallback.
translation-rule 10
rule 1 1111 2222 abbreviated
exit
translate calling 10
The following is a sample configuration of digit translation rule 20, where the priority of the translation
rule is 1 (the range is from 1 to 15) and the abbreviated representation of a complete number (1234) is
replaced with the number 2345:
translation-rule 20
rule 1 1234 2345 abbreviated
exit
Enabling Translation Profiles

Cisco SRST 3.2 and later and Cisco Unified SRST 4.0 and later support translation profiles. Translation
profiles are the suggested way to allow you to group translation rules and provide instructions on how
to apply the translation rules to the following:
• Called numbers
• Calling numbers
• Redirected called numbers

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In the configuration below, the voice translation-rule and the rule command allow you to set and define
how a number is to be manipulated. The translate command in voice translation-profile mode defines
the type of number you are going to manipulate, such as a called, calling, or a redirecting number. Once
you have defined your translation profiles, you can then apply the translation profiles in various places,
such as dial peers and voice ports. For SRST, you apply your profiles in call-manager fallback mode.
Cisco IP phones support one incoming and one outgoing translation profile when in SRST mode.
Note For Cisco SRST 3.2 and later versions and Cisco Unified SRST 4.0 and later versions, use the voice
translation-rule and translation-profile commands shown below instead of the translation rule
configuration described in the “Enabling Digit Translation Rules” section on page 191. Voice translation
rules are a separate feature from translation rules. See the voice translation-rule command in Cisco IOS
Voice Command Reference for more information and the VoIP Gateway Trunk and Carrier Based
Routing Enhancements documentation for more general information on translation rules and profiles.
SUMMARY STEPS
1. voice translation-rule number

2. rule precedence/match-pattern/ /replace-pattern/
3. exit
4. voice translation-profile name
5. translate {called | calling | redirect-called} voice-translation-rule-tag
6. exit
8. translation-profile {incoming | outgoing} name
9. exit

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DETAILED STEPS

Step 1 voice translation-rule number Defines a translation rule for voice calls and enters voice
translation-rule configuration mode.
Example: • number: Number that identifies the translation rule.
Router(config)# voice translation-rule 1 Range is from 1 to 2147483647.
Step 2 rule precedence/match-pattern/ Defines a translation rule.
/replace-pattern/
• precedence: Priority of the translation rule. Range is
from 1 to 15.
Example:
Router(cfg-translation-rule)# rule 1/^9/ //
• match-pattern: Stream editor (SED) expression used to
match incoming call information. The slash (/) is a
delimiter in the pattern.
• replace-pattern: SED expression used to replace the
match pattern in the call information. The slash (/) is a
delimiter in the pattern.
Step 3 exit Exits voice translation-rule configuration mode.
Example:
Router(cfg-translation-rule)# exit
Step 4 voice translation-profile name Defines a translation profile for voice calls.
• name: Name of the translation profile. Maximum
Example: length of the voice translation profile name is 31
Router(config)# voice translation-profile name1 alphanumeric characters.
Step 5 translate {called | calling | redirect-called} Associates a voice translation rule with a voice translation
translation-rule-number profile.
• called: Associates the translation rule with called
Example: numbers.
Router(cfg-translation-profile)# translate
called 1 • calling: Associates the translation rule with calling
numbers.
• redirect-called: Associates the translation rule with
redirected called numbers.
• translation-rule-number: The reference number of the
translation rule from 1 to 2147483647.
Step 6 exit Exits translation-profile configuration mode.
Example:
Router(cfg-translation-profile)# exit
Example:

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Step 8 translation-profile {incoming | outgoing} name Assigns a translation profile for incoming or outgoing call
legs on a Cisco IP phone.
Example: • incoming: Applies the translation profile to incoming
Router(config-cm-fallback)# translation-profile calls.
outgoing name1
• outgoing: Applies the translation profile to outgoing
calls.
• name: The name of the translation profile.
Example:
Examples
The following example shows the configuration where a translation profile called name1 is created with
two voice translation rules. Rule1 consists of associated calling numbers, and rule2 consists of redirected
called numbers. The Cisco Unified IP Phones in SRST mode are configured with name1.
voice translation-profile name1
translate calling 1
translate called redirect-called 2
translation-profile incoming name1
Verifying Translation Profiles

To verify translation profiles, perform the following steps.
SUMMARY STEPS
1. show voice translation-rule number

2. test voice translation-rule number input-test-string [type match-type [plan match-type]]

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DETAILED STEPS

Step 1 show voice translation-rule number Use this command to verify the translation rules that
you have defined for your translation profiles.
Example:
Router# show voice translation-rule 6
Translation-rule tag: 6
Rule 1:
Match pattern: 65088801..
Replace pattern: 6508880101
Match type: none Replace type: none
Match plan: none Replace plan: none
Step 2 test voice translation-rule number input-test-string Use this command to test your translation profiles.
[type match-type [plan match-type]] See the test voice translation-rule command in
Cisco IOS Voice Command Reference for more
Example: information.
Router(config)# voice translation-rule 5
Router(cfg-translation-rule)# rule 1 /201/ /102/
Router(cfg-translation-rule)# end
Router# test voice translation-rule 5 2015550101
Matched with rule 5
Original number:2015550101 Translated
number:1025550101
Original number type: none Translated number
type: none
Original number plan: none Translated number
plan: none
Configuring Dial-Peer and Channel Hunting

Dial-peer hunting, the search through a group of dial peers for an available phone line, is disabled during
Cisco Unified CM fallback by default. To enable dial-peer hunting, use the no huntstop command. For
more information about dial-peer hunting, see Cisco IOS Voice Configuration Library.
If you have a dual-line phone configuration, see the “Configuring Dual-Line Phones” section on
page 148. Keep incoming calls from hunting to the second channel if the first channel is busy or does
not answer by using the channel keyword in the huntstop command.
Channel huntstop also prevents situations in which a call can ring for 30 seconds on the first channel of
a line with no person available to answer and then ring for another 30 seconds on the second channel
before rolling over to another line.
SUMMARY STEPS
2. huntstop [channel]
3. exit

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DETAILED STEPS

Example:
Step 2 huntstop [channel] Sets the huntstop attribute for the dial peers associated with
the Cisco Unified IP Phone dial peers created during
Communications Manager fallback.
Example:
Router(config-cm-fallback)# huntstop channel • For dual-line configurations, the channel keyword
keeps incoming calls from hunting to the second
channel if the first channel is busy or does not answer.
Example:
Examples
The following example disables dial-peer hunting during Cisco Unified CM fallback and hunting to the
secondary channels in dual-line phone configurations:
no huntstop channel
Configuring Busy Timeout

This task sets the timeout value for call transfers to busy destinations. The busy timeout value is the
amount of time that can elapse after a transferred call reaches a busy signal before the call is
disconnected.
SUMMARY STEPS
2. timeouts busy seconds
3. exit

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DETAILED STEPS

Example:
Step 2 timeouts busy seconds Sets the amount of time after which calls are disconnected
when they are transferred to busy destinations.
Example: • seconds: Number of seconds. Range is from 0 to 30.
Router(config-cm-fallback)# timeouts busy 20 Default is 10.
Note This command sets the busy timeout only for calls
that are transferred to busy destinations and does
not affect the timeout for calls that directly dial
busy destinations.
Example:
Examples
The following example sets a timeout of 20 seconds for calls that are transferred to busy destinations:
timeouts busy 20
Configuring the Ringing Timeout Default

The ringing timeout default is the length of time for which a phone can ring with no answer before
returning a disconnect code to the caller. This timeout prevents hung calls received over interfaces such
as Foreign Exchange Office (FXO) that do not have forward-disconnect supervision. It is used only for
extensions that do not have no-answer call forwarding enabled.
SUMMARY STEPS
2. timeouts ringing seconds
3. exit

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DETAILED STEPS

Example:
Step 2 timeouts ringing seconds Sets the ringing timeout default, in seconds. The range is
from 5 to 60000. There is no default value.
Example:
Router(config-cm-fallback)# timeouts ringing 30
Example:
Examples
The following example sets the ringing timeout default to 30 seconds:
timeouts ringing 30
Configuring Outgoing Calls

Outgoing call configuration can include the following tasks:
• Configuring Call Transfer
– Configuring Local and Remote Call Transfer, page 199 (Optional)
– Enabling Consultative Call Transfer and Forward Using H.450.2 and H.450.3 with Cisco SRST
3.0, page 200 (Optional)
– Enabling Analog Transfer Using Hookflash and the H.450.2 Standard with Cisco SRST 3.0 or
Earlier, page 204 (Optional)
• Configuring Trunk Access Codes, page 208 (Required Under Certain Conditions)
• Configuring Interdigit Timeout Values, page 209 (Optional)
• Configuring Class of Restriction, page 210 (Optional)
• Call Blocking (Toll Bar) Based on Time of Day and Day of Week or Date, page 214 (Optional)
Configuring Local and Remote Call Transfer

You must configure Cisco Unified SRST to allow Cisco Unified IP Phones to transfer telephone calls
from outside the local IP network to another Cisco Unified IP Phone. By default, all Cisco Unified IP
Phone directory numbers or virtual voice ports are allowed as transfer targets. A maximum of 32 transfer
patterns can be entered.
Call transfer configuration is performed using the transfer-pattern command.

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SUMMARY STEPS
3. exit
DETAILED STEPS

Example:
Step 2 transfer-pattern transfer-pattern Enables the transfer of a call from a non-IP phone number
to another Cisco Unified IP Phone on the same IP network
using the specified transfer pattern.
Example:
Router(config-cm-fallback)# transfer-pattern • transfer-pattern: String of digits for permitted call
52540.. transfers. Wildcards are permitted.
Example:
Examples
In the following example, the transfer-pattern command permits transfers from a non-IP phone number
to any Cisco Unified IP Phone on the same IP network with a number in the range from 5550100 to
5550199:
transfer-pattern 55501..
Enabling Consultative Call Transfer and Forward Using H.450.2 and H.450.3 with Cisco SRST
3.0
Consultative call transfer using H.450.2 adds support for initiating call transfers and call forwarding on
a call leg using the ITU-T H.450.2 and ITU-T H.450.3 standards. Call transfers and call forwarding using
H.450.2 and H.450.3 can be blind or consultative. A blind call transfer or blind call forward is one in
which the transferring or forwarding phone connects the caller to a destination line before a ringing tone
begins. A consultative transfer is one in which the transferring or forwarding party either connects the
caller to a ringing phone (ringback heard) or speaks with the third party before connecting the caller to
the third party.
Note For Cisco SRST 3.1 and later versions and Cisco Unified SRST 4.0 and later versions, call transfer and
call forward using H.450.2 is supported automatically with the default session application.

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Prerequisites
• Call transfer with consultation is available only when a second line or call instance is supported by
the IP phone. Please see the dual-line keyword in the max-dn command.
• All voice gateway routers in the VoIP network must support the H.450 standard.
• All voice gateway routers in the VoIP network must be running the following software:
– Cisco IOS Release 12.3(2)T or a later release
– Cisco SRST 3.0
Restrictions
H.450.12 Supplementary Services Capabilities exchange among routers is not implemented.
SUMMARY STEPS
2. call-forward pattern pattern (call forward only)
3. transfer-system {blind | full-blind | full-consult | local-consult} (call transfer only)
4. transfer-pattern transfer-pattern (call transfer only)
5. exit
7. h323
8. h450 h450-2 timeout {T1 | T2 | T3 | T4} milliseconds
9. end

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DETAILED STEPS

Example:
Step 2 call-forward pattern pattern Specifies the H.450.3 standard for call forwarding.
• pattern: Digits to match for call forwarding using the
Example: H.450.3 standard. If an incoming calling-party number
Router(config-cm-fallback)# call-forward matches the pattern, it can be forwarded using the
pattern 4... H.450.3 standard. A pattern of .T forwards all calling
parties using the H.450.3 standard.
Step 3 transfer-system {blind | full-blind | Not supported if the transfer-to destination is on the Cisco
full-consult | local-consult} ATA, Cisco VG224, or an SCCP-controlled FXS port.
Defines the call-transfer method for all lines served by the
Example: Cisco Unified SRST router.
Router(config-cm-fallback)# transfer-system
full-consult • blind: Calls are transferred without consultation with a
single phone line using the Cisco proprietary method.
Note We do not recommend the blind keyword. Use

either the full-blind or full-consult keyword
instead.
• full-blind: Calls are transferred without consultation

using H.450.2 standard methods.
• full-consult: Calls are transferred with consultation
using a second phone line if available. The calls fall
back to full-blind if the second line is unavailable.
• local-consult: Calls are transferred with local
consultation using a second phone line if available. The
calls fall back to blind for nonlocal consultation or
nonlocal transfer target.
Step 4 transfer-pattern transfer-pattern Allows transfer of telephone calls by Cisco Unified IP
Phones to specified phone number patterns.
Example: • transfer-pattern: String of digits for permitted call
Router(config-cm-fallback)# transfer-pattern transfers. Wildcards are allowed.
52540..
Timesaver Before exiting call-manager-fallback
Example: configuration mode, configure any other
Router(config-cm-fallback)# exit parameters that you need to set for the entire
Cisco Unified SRST phone network.

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Step 6 voice service voip (Optional) Enters voice service configuration mode.
Example:
Step 7 h323 (Optional) Enters H.323 voice service configuration mode.
Example:
Router(conf-voi-serv)# h323
Step 8 h450 h450-2 timeout {T1 | T2 | T3 | T4} (Optional) Sets timeouts for supplementary service timers,
milliseconds in milliseconds. This command is used primarily when the
default settings for these timers do not match your network
Example: delay parameters. See the ITU-T H.450.2 specification for
Router(conf-serv-h323)# h450 h450-2 timeout T1 more information on these timers.
750
• T1: Timeout value to wait to identify a response.
Default is 2000.
• T2: Timeout value to wait for call setup.
Default is 5000.
• T3: Timeout value to wait to initiate a response. Default
is 5000.
• T4: Timeout value to wait for setup of a response.
Default is 5000.
• milliseconds: Number of milliseconds.
Range is from 500 to 60000.
Step 9 end (Optional) Returns to privileged EXEC mode.
Example:
Router(conf-serv-h323)# end
Examples
The following example specifies transfer with consultation using the H.450.2 standard for all IP phones
serviced by the Cisco Unified SRST router:
destination-pattern 9.T
port 1/0/0

destination-pattern 4…
session-target ipv4:10.1.1.1
transfer-pattern 4…
The following example enables call forwarding using the H.450.3 standard:
port 1/0/0

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!
!
call-forward pattern 4
Enabling Analog Transfer Using Hookflash and the H.450.2 Standard with Cisco SRST 3.0 or
Earlier
Analog call transfer using hookflash and the H.450.2 standard allows analog phones to transfer calls with
consultation by using the hookflash to initiate the transfer. Hookflash refers to the short on-hook period
usually generated by a telephone-like device during a call to indicate that the telephone is attempting to
perform a dial-tone recall from a PBX. Hookflash is often used to perform call transfer. For example, a
hookflash occurs when a caller quickly taps once on the button in the cradle of an analog phone’s
handset.
This feature requires installation of a Tool Command Language (Tcl) script. The script
app-h450-transfer.tcl must be downloaded from the Cisco Software Center at
http://www.cisco.com/cgi-bin/tablebuild.pl/ip-iostsp and copied to a TFTP server that is available to the
Cisco Unified SRST router or copied to the flash memory on the Cisco Unified SRST router. To apply
this script globally to all dial peers, use the call application global command in global configuration
mode. The Tcl script has parameters to which you can pass values using attribute-value (AV) pairs in the
call application voice command. The parameter that applies to this feature is as follows:
• delay-time: Speeds up or delays the setting up of the consultation call during a call transfer from an
analog phone using a delay timer. When all digits have been collected, the delay timer is started. The
call setup to the receiving party does not begin until the delay timer expires. If the transferring party
goes on-hook before the delay timer expires, the transfer is considered a blind transfer rather than a
consultative transfer. If the transferring party goes on-hook after the delay timer expires, either
while the destination phone is ringing or after the destination party answers, the transfer is
considered a consultative transfer.
In addition to the Tcl script, a ReadMe file describes the script and the configurable AV pairs. Read this
file whenever you download a new version of the script because it may contain additional script-specific
information, such as configuration parameters and user interface descriptions.
Note For Cisco SRST 3.1 and later versions and Cisco Unified SRST 4.0 and later versions, call transfer using
H.450.2 is supported automatically with the default session application.
Prerequisites
• The H.450 Tcl script named app-h450-transfer.tcl must be downloaded from the Cisco Software
Center. The following versions of the script are available:
– app-h450-transfer.2.0.0.2.tcl for Cisco IOS Release 12.2(11)YT1 and later releases
– app-h450-transfer.2.0.0.1.tcl for Cisco IOS Release 12.2(11)YT
• All voice gateway routers in the VoIP network must support H.450 and be running the following
software:
– Cisco IOS Release 12.2(11)YT or a later release
– Cisco SRST V3.0 or a lower version

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– Tcl IVR 2.0

– H.450 Tcl script (app-h450-transfer.tcl)
Note You can continue to use the app-h450-transfer.2.0.0.1.tcl script if you install Cisco IOS
Release 12.2(11)YT1 or later, but you cannot use the app-h450-transfer.2.0.0.2.tcl script with a release
of Cisco IOS software that is earlier than Cisco IOS Release 12.2(11)YT1.
Restrictions
• When a consultative transfer is made by an analog FXS phone using hookflash, the consultation call
itself cannot be further transferred (that is, it cannot become a recursive or chained transfer) until
after the initial transfer operation is completed and the transferee and transfer-to parties are
connected. After the initial call transfer operation is completed and the transferee and transfer-to
parties are now the only parties in the call, the transfer-to party may further transfer the call.
• Call transfer with consultation is not supported for Cisco ATA-186, Cisco ATA-188, and Cisco IP
Conference Station 7935. Transfer attempts from these devices are executed as blind transfers.
SUMMARY STEPS
1. call application voice application-name location

2. call application voice application-name language number language
3. call application voice application-name set-location language category location
4. call application voice application-name delay-time seconds
5. dial-peer voice number pots
7. exit
8. dial-peer voice number voip
10. exit

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DETAILED STEPS

Step 1 call application voice application-name Loads the Tcl script and specifies its application name.
location
• application-name: User-defined name for the IVR
application. This name does not have to match the
Example: script filename.
Router(config)# call application voice
transfer_app flash:app-h450-transfer.tcl • location: Script directory and filename in URL format.
For example, flash memory (flash:filename), a TFTP
Step 2 call application voice application-name (Optional) Sets the language for dynamic prompts used by
language number language the application.
• application-name: IVR application name that was
Example: assigned in Step 1.
transfer_app language 1 en • number: Number that identifies the language used by
the audio files for the IVR application.
• language: Two-character code that specifies the
language of the prompts. Valid entries are en
(English:default), sp (Spanish), ch (Chinese), or aa
(all).
Step 3 call application voice application-name Defines the location and category of the audio files that are
set-location language category location used by the application for dynamic prompts.
• application-name: Name of the Tcl IVR application.
Example:
• language: Two-character code to specify the language
transfer_app set-location en 0 flash:/prompts of the prompts. Valid entries are en (English: default),
sp (Spanish), ch (Chinese), or aa (all).
• category: Category group (0 to 4) for the audio files
from this location. The value 0 means all categories.
• location: URL of the directory that contains the
language audio files used by the application, without
filenames. Flash memory (flash) or a directory on a
server (TFTP, HTTP, or RTSP) are all valid.
Prompts are required for call transfer from analog FXS
phones. No prompts are needed for call transfer from IP
phones.

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Step 4 call application voice application-name (Optional) Sets the delay time for consultation call setup for
delay-time seconds an analog phone that is making a call transfer using the
H.450 application. This command passes a value to the Tcl
Example: script by using an attribute-value (AV) pair.
Router(config)# call application voice • seconds: Number of seconds to delay call setup. Range
transfer_app delay-time 1
is from 1 to 10. Default is 2.
A delay of more than 2 seconds is generally noticeable to
users.
For more information about AV pairs and the Tcl script for
H.450 call transfer and forwarding, see the ReadMe file that
accompanies the script.
Step 5 dial-peer voice number pots Enters dial-peer configuration mode to configure a POTS
dial peer.
Example:
Router(config)# dial-peer voice 25 pots
Step 6 application application-name Loads the application named in Step 1 onto the dial peer.
Example:
Router(config-dial-peer)# application
transfer_app
Step 7 exit Exits dial-peer configuration mode.
Timesaver Before exiting dial-peer configuration mode,
Example: configure any other dial-peer parameters that
Router(config-dial-peer)# exit you need to set for this dial peer.
Step 8 dial-peer voice number voip Enters dial-peer configuration mode to configure a VoIP
dial peer.
Example:
Step 9 application application-name Loads the application named in Step 1 onto the dial peer.
Example:
Router(config-dial-peer)# application
transfer_app
Timesaver Before exiting dial-peer configuration mode,
Example: configure any other dial-peer parameters that
Router(config-dial-peer)# exit you need to set for this dial peer.

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Examples
The following example enables the H.450 Tcl script for analog transfer using hookflash and sets a delay
time of 1 second:
call application voice transfer_app flash:app-h450-transfer.tcl
call application voice transfer_app language 1 en
call application voice transfer_app set-location en 0 flash:/prompts
call application voice transfer_app delay-time 1
!
port 1/0/0
application transfer_app
!
destination-pattern 4…
application transfer_app
Configuring Trunk Access Codes
Note Configure trunk access codes only if your normal network dial-plan configuration prevents you from
configuring permanent POTS voice dial peers to provide trunk access for use during fallback. If you
already have local PSTN ports configured with the appropriate access codes provided by dial peers (for
example, dial 9 to select an FXO PSTN line), this configuration is not needed.
Trunk access codes provide IP phones with access to the PSTN during Cisco Unified CM fallback by
creating POTS voice dial peers that are active during Cisco Unified CM fallback only. These temporary
dial peers, which can be matched to voice ports (BRI, E&M, FXO, and PRI), allow Cisco Unified IP
Phones access to trunk lines during Cisco Unified CM mode. When Cisco Unified SRST is active, all
PSTN interfaces of the same type are treated as equivalent, and any port may be selected to place the
outgoing PSTN call.
Trunk access codes are created using the access-code command.
SUMMARY STEPS
2. access-code {{fxo | e&m} dial-string | {bri | pri} dial-string [direct-inward-dial]}
3. exit

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DETAILED STEPS

Example:
Step 2 access-code {{fxo | e&m} dial-string | {bri | Configures trunk access codes for each type of line so that
pri} dial-string [direct-inward-dial]} the Cisco Unified IP Phones can access the trunk lines only
in Cisco Unified Communications Manager fallback mode
Example: when the Cisco Unified SRST is enabled.
Router(config-cm-fallback)# access-code e&m 8 • fxo: Enables a Foreign Exchange Office (FXO)
interface.
• e&m: Enables an analog Ear and Mouth (E&M)
interface.
• dial-string: String of characters that sets up dial access
codes for each specified line type by creating dial
peers. The dial-string argument is used to set up
temporary dial peers for each specified line type.
• bri: Enables a BRI interface.
• pri: Enables a PRI interface.
• direct-inward-dial: (Optional) Enables Direct Inward
Dialing (DID) on the POTS dial peer.
Example:
Examples
The following example creates access code number 8 for BRI and enables DID on the POTS dial peer:
access-code bri 8 direct-inward-dial
Configuring Interdigit Timeout Values

Configuring interdigit timeout values involves specifying how long, in seconds, all Cisco Unified IP
Phones attached to a Cisco Unified SRST router are to wait after an initial digit or a subsequent digit is
dialed. The timeouts interdigit timer is enabled when a caller enters a digit and is restarted each time
the caller enters subsequent digits until the destination address is identified. If the configured timeout
value is exceeded before the destination address is identified, a tone sounds and the call is terminated.

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SUMMARY STEPS
2. timeouts interdigit seconds
3. exit
DETAILED STEPS

Example:
Step 2 timeouts interdigit seconds (Optional) Configures the interdigit timeout value for all
Cisco IP phones that are attached to the router.
Example: • seconds: Interdigit timeout duration, in seconds, for all
Router(config-cm-fallback)# timeouts interdigit Cisco Unified IP Phones. Valid entries are integers
5 from 2 to 120.
Example:
Examples
The following example sets the interdigit timeout value to 5 seconds for all Cisco Unified IP Phones. In
this example, 5 seconds are the elapsed time after which an incompletely dialed number times out. For
example, a caller who dials nine digits (408555010) instead of the required ten digits (4085550100) will
hear a busy tone after the second timeout elapses.
timeouts interdigit 5
Configuring Class of Restriction

The class of restriction (COR) functionality provides the ability to deny certain call attempts on the basis
of the incoming and outgoing class of restrictions provisioned on the dial peers. This functionality
provides flexibility in network design, allows users to block calls (for example, calls to 900 numbers),
and applies different restrictions to call attempts from different originators. The cor command sets the
dial-peer COR parameter for dial peers associated with the directory numbers created during
Cisco Unified CM fallback.
You can have up to 20 COR lists for each incoming and outgoing call. A default COR is assigned to
directory numbers that do not match any COR list numbers or number ranges. An assigned COR is
invoked for the dial peers and created for each directory number automatically during Communications
Manager fallback registration.
If a COR is applied on an incoming dial peer (for incoming calls) and it is a superset of or is equal to the
COR applied to the outgoing dial peer (for outgoing calls), the call will go through. Voice ports
determine whether a call is considered incoming or outgoing. If you hook up a phone to an FXS port on

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a Cisco Unified SRST router and try to make a call from that phone, the call will be considered an
incoming call to the router and voice port. If you make a call to the FXS phone, the call will be
considered outgoing.
By default, an incoming call leg has the highest COR priority; the outgoing call leg has the lowest
priority. If there is no COR configuration for incoming calls on a dial peer, you can make a call from a
phone attached to the dial peer, so that the call will go out of any dial peer regardless of the COR
configuration on that dial peer. Table 8-2 describes call functionality based on how your COR lists are
configured.
Table 8-2 Combinations of COR List and Results
COR List on COR List on

Incoming Dial Peer Outgoing Dial Peer Result
No COR No COR Call will succeed.
No COR COR list applied for Call will succeed. By default, the incoming dial peer
outgoing calls has the highest COR priority when no COR is applied.
If you apply no COR for an incoming call leg to a dial
peer, the dial peer can make a call out of any other dial
peer regardless of the COR configuration on the
outgoing dial peer.
COR list applied for No COR Call will succeed. By default, the outgoing dial peer
incoming calls has the lowest priority. Because there are some COR
configurations for incoming calls on the incoming or
originating dial peer, it is a superset of the outgoing
call’s COR configuration for the outgoing or
terminating dial peer.
COR list applied for COR list applied for Call will succeed. The COR list for incoming calls on
incoming calls outgoing calls the incoming dial peer is a superset of the COR list for
(superset of COR list (subsets of COR list outgoing calls on the outgoing dial peer.
applied for outgoing applied for incoming
calls on the outgoing calls on the incoming
dial peer) dial peer)
COR list applied for COR list applied for Call will not succeed. The COR list for incoming calls
incoming calls outgoing calls on the incoming dial peer is not a superset of the COR
(subset of COR list (supersets of COR list list for outgoing calls on the outgoing dial peer.
applied for outgoing applied for incoming
calls on the outgoing calls on the incoming
dial peer) dial peer)
SUMMARY STEPS
2. cor {incoming | outgoing} cor-list-name {cor-list-number starting-number - ending-number |
default}
3. exit

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DETAILED STEPS

Example:
Step 2 cor {incoming | outgoing} cor-list-name Configures a COR on dial peers associated with directory
[cor-list-number starting-number - numbers.
ending-number | default]
Example: • outgoing: COR list to be used by outgoing dial peers.

Router(config-cm-fallback)# cor outgoing • cor-list-name: COR list name.
LockforPhoneC 1 5010 – 5020
number of COR lists that can be created is 20,
comprised of incoming or outgoing dial peers. The
first six COR lists are applied to a range of directory
numbers. The directory numbers that do not have a
COR configuration are assigned to the default COR
list.
• starting-number - ending-number: Directory number
range; for example, 2000–2025.
COR list.
Example:
Examples
The following example shows how to set a dial-peer COR parameter for outgoing calls to the
Cisco Unified IP Phone dial peers and directory numbers created during fallback:
cor outgoing LockforPhoneC 1 5010 - 5020
The following example shows how to set the dial-peer COR parameter for incoming calls to the Cisco IP
phone dial peers and directory numbers in the default COR list:
cor incoming LockforPhoneC default
The following example shows how sub- and super-COR sets are created. First, a custom dial-peer COR
is created with names declared under it:
name 911
name 1800
name 1900
name local_call

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In the following configuration example, COR lists are created and applied to the dial peer:
dial-peer cor list call911
member 911

member 1800

member 1900
dial-peer cor list calllocal

member local_call
dial-peer cor list engineering

member 911
member local_call
dial-peer cor list manager

member 911
member 1800
member 1900
member local_call
dial-peer cor list hr

member 911
member 1800
member local_call
In the example below, five dial peers are configured for destination numbers 734…., 1800…….,
1900……., 316…., and 911. A COR list is applied to each of the dial peers.
destination pattern 734....
cor outgoing calllocal

destination pattern 1800.......
cor outgoing call1800

destination pattern 1900.......
port 1/0/0

destination pattern 316....
port 1/1/0
! No COR is applied.

destination pattern 911
port 1/0/1
Finally, the COR list is applied to the individual phone numbers.

max-conferences 8
cor incoming engineering 1 1001 - 1001
cor incoming hr 2 1002 - 1002
cor incoming manager 3 1003 - 1008

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The sample configuration allows for the following:

• Extension 1001 to call 734... numbers, 911, and 316....
• Extension 1002 to call 734..., 1800 numbers, 911, and 316....
• Extension 1003 to 1008 to call all of the possible Cisco Unified SRST router numbers
• All extensions to call 316....
Call Blocking (Toll Bar) Based on Time of Day and Day of Week or Date
Call blocking to prevent unauthorized use of phones is implemented by matching a pattern of specified
digits during a specified time of day and day of week or date. Up to 32 patterns of digits can be specified.
Call blocking is supported on IP phones only and not on analog foreign exchange station (FXS) phones.
When a user attempts to place a call to digits that match a pattern that is specified for call blocking during
a time period that is defined for call blocking, a fast busy signal is played for approximately 10 seconds.
The call is then terminated, and the line is placed back in on-hook status.
In SRST (call-manager-fallback configuration) mode, there is no phone- or pin-based exemption to
after-hours call blocking.
SUMMARY STEPS
5. exit

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DETAILED STEPS

Example:
pattern 1 91900
Example:
pattern command.
sat.
jan 1 0:00 0:00
The value 24:00 is not valid. If 00:00 is entered as an
Example:
Examples

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How to Configure Cisco Unified SIP SRST
and 2, which block calls to external numbers that begin with “1” and “011,” are blocked on Monday
through Friday before 7 a.m. and after 7 p.m., on Saturday before 7 a.m. and after 1 p.m., and all day
Sunday. Pattern 3 blocks calls to 900 numbers 7 days a week, 24 hours a day.
after-hours block day mon 19:00 07:00
after-hours block day tue 19:00 07:00
after-hours block day wed 19:00 07:00
after-hours block day thu 19:00 07:00
after-hours block day fri 19:00 07:00
after-hours block day sat 13:00 12:00
after-hours block day sun 12:00 07:00

• Configuring SIP Phone Features, page 216 (optional)
• Configuring SIP-to-SIP Call Forwarding, page 218 (required)
• Configuring Call Blocking Based on Time of Day, Day of Week, or Date, page 220 (required)
• SIP Call Hold and Resume, page 224 (no configuration necessary)
• Examples, page 224
Configuring SIP Phone Features

After a voice register pool has been set, this procedure adds optional features to increase functionality.
Some features can be made per pool or globally.
In voice register pool configuration, you can now configure several new options per pool (a pool can be
one phone or a group of phones). There is also a new voice register global configuration mode for
Cisco Unified SIP SRST. In voice register global mode, you can globally assign characteristics to
phones.
SUMMARY STEPS
1. enable
3. voice register global tag
6. external ring {bellcore-dr1 | bellcore-dr2 | bellcore-dr3 | bellcore-dr4 | bellcore-dr5}
7. exit
9. no vad

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10. codec codec-type [bytes]

11. end
DETAILED STEPS

Example:
Router> enable
Example:
Step 3 voice register global tag Enters voice register global configuration mode to set
global parameters for all supported Cisco SIP IP phones in
a Cisco Unified SIP SRST environment.
Example:
Router(config)# voice register global 12
Step 4 max-pool max-voice-register-pools Sets the maximum number of SIP voice register pools that
are supported in a Cisco Unified SIP SRST environment.
The max-voice-register-pools argument represents the
Example:
maximum number of SIP voice register pools supported by
the Cisco Unified SIP SRST router. The upper limit of voice
register pools is version- and platform-dependent; see
Cisco IOS command-line interface (CLI) help. Default is 0.
Step 5 application application-name Selects the session-level application for all dial peers
associated with SIP phones. Use the application-name
argument to define a specific interactive voice response
Example:
Router(config-register-global)# application
(IVR) application.
global_app
Step 6 external-ring {bellcore-dr1 | bellcore-dr2 | Specifies the type of ring sound used on Cisco SIP or
bellcore-dr3 | bellcore-dr4 | bellcore-dr5} Cisco SCCP IP phones for external calls. Each bellcore-dr
1-5 keyword supports standard distinctive ringing patterns
Example: as defined in the standard GR-506-CORE, LSSGR:
Router(config-register-global)# external-ring Signaling for Analog Interfaces.
bellcore-dr1
Example:
phones.
Example: • Use this command to control which phone registrations
Router(config)# voice register pool 20 are to be accepted or rejected by a Cisco Unified SIP
SRST device.

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Step 9 no vad Disables voice activity detection (VAD) on the VoIP dial
peer.
Example: • VAD is enabled by default. Because there is no comfort
Router(config-register-pool)# no vad noise during periods of silence, the call may seem to be
disconnected. You may prefer to set no vad on the SIP
phone pool.
Step 10 codec codec-type [bytes] Specifies the codec supported by a single SIP phone or a
VoIP dial peer in a Cisco Unified SIP SRST environment.
The codec-type argument specifies the preferred codec and
Example:
Router(config-register-pool)# codec g729r8
can be one of the following:
• g711alaw: G.711 a–law 64,000 bps.
• g711ulaw: G.711 mu–law 64,000 bps.
• g729r8: G.729 8000 bps (default).
The bytes argument is optional and specifies the number of
bytes in the voice payload of each frame
Example:
Configuring SIP-to-SIP Call Forwarding

SIP-to-SIP call forwarding (call routing) is available. Call forwarding is provided either by the phone or
by using a back-to-back user agent (B2BUA), which allows call forwarding on any dial peer. Calls into
a SIP device may be forwarded to other SIP or SCCP devices (including Cisco Unity, third-party
voice-mail systems, or an auto attendant or IVR system such as IPCC and IPCC Express). In addition,
SCCP IP phones may be forwarded to SIP phones.
Cisco Unity or other voice messaging systems connected by a SIP trunk or SIP user agent are able to
pass a message-waiting indicator (MWI) when a message is left. The SIP phone then displays the MWI
when indicated by the voice messaging system.
Note SIP-to-H.323 call forwarding is not supported.
To configure SIP-to-SIP call forwarding, you must first allow connections between specific types of
endpoints in a Cisco IP-to-IP gateway. The allow-connections command grants this capability. Once the
SIP-to-SIP connections are allowed, you can configure call forwarding under an individual SIP phone
pool. Any of the following commands can be used to configure call forwarding, according to your needs:
• Under voice register pool
– call-forward b2bua all directory-number
– call-forward b2bua busy directory-number
– call-forward b2bua mailbox directory-number
– call-forward b2bua noan directory-number [timeout seconds]

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In a typical Cisco Unified SIP SRST setup, the call-forward b2bua mailbox command is not used;
however, it is likely to be used in a Cisco Unified SIP Communications Manager Express (CME)
environment. Detailed procedures for configuring the call-forward b2bua mailbox command are found
in the Cisco Unified Communications Manager (CallManager) documentation on Cisco.com.
SUMMARY STEPS
1. enable
4. call-forward b2bua all directory-number
6. call-forward b2bua mailbox directory-number
8. end
DETAILED STEPS

Example:
Router> enable
Example:
• Use this command to control which phone registrations
Example: are accepted or rejected by a Cisco Unified SIP SRST
Step 4 call-forward b2bua all directory-number Enables call forwarding for a SIP back-to-back user agent
(B2BUA) so that all incoming calls are forwarded to
another non-SIP station extension (that is, SIP trunk, H.323
Example:
Router(config-register-pool)# call-forward
trunk, SCCP device or analog/digital trunk).
b2bua all 5005 • directory-number: Telephone number to which calls are
Maximum length of the telephone number is 32.
Step 5 call-forward b2bua busy directory-number Enables call forwarding for a SIP B2BUA so that incoming
calls to a busy extension are forwarded to another extension.
Example: • directory-number: Telephone number to which calls are
Router(config-register-pool)# call-forward forwarded. Represents a fully qualified E.164 number.
b2bua busy 5006 Maximum length of the telephone number is 32.

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Step 6 call-forward b2bua mailbox directory-number Controls the specific voice-mail box selected in a
voice-mail system at the end of a call forwarding exchange.
Example: • directory-number: Telephone number to which calls are
Router(config-register-pool)# call-forward forwarded when the forwarded destination is busy or
b2bua mailbox 5007 does not answer. Represents a fully qualified E.164
number.
Step 7 call-forward b2bua noan directory-number Enables call forwarding for a SIP B2BUA so that incoming
timeout seconds calls to an extension that does not answer after a configured
amount of time are forwarded to another extension.
Example: This command is used if a phone is registered with a
Router(config-register-pool)# call-forward Cisco Unified SIP SRST router, but the phone is not
b2bua noan 5010 timeout 10
reachable because there is no IP connectivity (there is no
response to Invite requests).
• directory-number: Telephone number to which calls are
• timeout seconds: Duration, in seconds, that a call can
ring with no answer before the call is forwarded to
another extension. Range is 3 to 60000.
The default value is 20.
Example:
Configuring Call Blocking Based on Time of Day, Day of Week, or Date

This section applies to both SCCP and SIP SRST. Call blocking prevents the unauthorized use of phones
and is implemented by matching a pattern of up to 32 digits during a specified time of day, day of week,
or date. Cisco Unified SIP SRST provides SIP endpoints the same time-based call blocking mechanism
that is currently provided for SCCP phones. The call blocking feature supports all incoming calls,
including incoming SIP and analog FXS calls.
Note Pin-based exemptions and the “Login” toll-bar override are not supported in Cisco Unified SIP SRST.
The commands used for SIP phone call blocking are the same commands that are used for SCCP phones
on your Cisco Unified SRST system. The Cisco SRST session application accesses the current
after-hours configuration under call-manager-fallback mode and applies it to calls originated by
Cisco SIP phones that are registered to the Cisco SRST router. The commands used in
call-manager-fallback mode that set block criteria (time/date/block pattern) are the following:
• after-hours block pattern pattern-tag pattern [7-24]
• after-hours day day start-time stop-time
• after-hours date month date start-time stop-time

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When a user attempts to place a call to digits that match a pattern that has been specified for call blocking
during a time period that has been defined for call blocking, the call is immediately terminated and the
caller hears a fast busy.
In SRST (call-manager-fallback configuration mode), there is no phone- or pin-based exemption to
after-hours call blocking. However, in Cisco Unified SIP SRST (voice register pool mode), individual IP
phones can be exempted from all call blocking using the after-hours exempt command.
SUMMARY STEPS
1. enable
7. exit
9. after-hour exempt
10. end
DETAILED STEPS

Example:
Router> enable
Example:
Example:
pattern 1 91900

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Example:
pattern command.
sat.
jan 1 00:00 00:00
Example:

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Step 9 after-hour exempt Specifies that for a particular voice register pool, none of its
outgoing calls are blocked even though call blocking is
enabled.
Example:
Router(config-register-pool)# after-hour exempt
Example:
Examples
and 2, which block calls to external numbers that begin with 1 and 011, are blocked on Monday through
Friday before 7 a.m. and after 7 p.m. Pattern 3 blocks calls to 900 numbers 7 days a week, 24 hours a day.
after-hours day mon 19:00 07:00
after-hours day tue 19:00 07:00
after-hours day wed 19:00 07:00
after-hours day thu 19:00 07:00
after-hours day fri 19:00 07:00
The following example exempts a Cisco SIP phone pool from the configured blocking criteria:
after-hour exempt
Verification
To verify the feature’s configuration, enter one of the following commands:
• show voice register dial-peer: Displays all the dial peers created dynamically by phones that have
registered. This command also displays configurations for after hours blocking and call forwarding.
• show voice register pool <tag>: Displays information regarding a specific pool.
• debug ccsip message: Debugs basic B2BUA calls.
For more information about these commands, see Cisco Unified SRST and Cisco Unified SIP SRST

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SIP Call Hold and Resume

Cisco Unified SRST supports the ability for SIP phones to place calls on hold and to resume from calls
placed on hold. This also includes support for a consultative hold where A calls B, B places A on hold,
B calls C, and B disconnects from C and then resumes with A. Support for call hold is signaled by SIP
phones using “re-INVITE c=0.0.0.0” and also by the receive-only mechanism.
Examples
Router# show running-config
Current configuration : 1462 bytes

configuration mode exclusive manual
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
service internal
!
boot-start-marker
boot-end-marker
!
logging buffered 8000000 debugging
!
no aaa new-model
!
resource policy
!
clock timezone edt -5
clock summer-time edt recurring
ip subnet-zero
!
!
!
ip cef
!
!
!
voice-card 0
no dspfarm
!
!
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
sip
!
!
!
max-dn 10
max-pool 10
!
! Define call forwarding under a voice register pool

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id mac 0012.7F57.60AA
number 1 1000
codec g711ulaw
!
id mac 0012.7F3B.9025
number 1 2800
codec g711ulaw
!
id mac 0012.7F57.628F
number 1 2801
codec g711ulaw
!
!
!
interface GigabitEthernet0/0
ip address 10.0.2.99 255.255.255.0
duplex auto
speed auto
!
interface GigabitEthernet0/1
no ip address
shutdown
duplex auto
speed auto
!
ip classless
ip route 0.0.0.0 0.0.0.0 GigabitEthernet0/0
!
ip http server
!
!
!
control-plane
!
!
!
destination-pattern 24..
codec g711ulaw
!
! Define call blocking under call-manager-fallback mode
after-hours date Dec 25 12:01 20:00

!
!
line con 0
exec-timeout 0 0
line aux 0
line vty 0 4
login
!
scheduler allocate 20000 1000

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How to Configure Optional Features
ntp server 10.0.2.10

!
end

This section describe the following optional additional call features:
• Three-party G.711 ad hoc conferencing—Cisco Unified Survivable Remote Site Telephony (SRST)
support for simultaneous three-party conferences.
• eXtensible Markup Language (XML) application program interface (API)— This interface supplies
data from Cisco Unified SRST to management software.
The following sections describe how to configure these optional features:
• Enabling Three-Party G.711 Ad Hoc Conferencing, page 226
• Defining XML API Schema, page 228
Enabling Three-Party G.711 Ad Hoc Conferencing

Enabling three-party G.711 ad hoc conferencing involves configuring the maximum number of
simultaneous three-party conferences supported by the Cisco Unified SRST router. For conferencing to
be available, an IP phone must have a minimum of two lines connected to one or more buttons. See the
“Configuring a Secondary Dial Tone” section on page 147.
SUMMARY STEPS
2. max-conferences max-conference-numbers
3. exit

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DETAILED STEPS

Example:
Step 2 max-conferences max-conference-numbers Sets the maximum number of simultaneous three-party
conferences supported by the router. The maximum number
possible is platform dependent:
Example:
Router(config-cm-fallback)# max-conferences 16 • Cisco 1751 router:8
• Cisco 1760 router:8
• Cisco 2600 series routers:8
• Cisco 2600-XM series routers:8
• Cisco 2801 router:8
• Cisco 2811, Cisco 2821, and Cisco 2851 routers:16
• Cisco 3640 and Cisco 3640A routers:8
• Cisco 3660 router: 16
• Cisco 3800 series router: 24
Example:
Examples
The following example configures up to eight simultaneous three-way conferences on a router:
max-conferences 8

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Configuration Examples for Call Handling
Defining XML API Schema

The Cisco IOS commands in this section allow you to specify parameters associated with the XML API.
For more information, see XML Provisioning Guide for Cisco CME/SRST. See the “Enabling
Consultative Call Transfer and Forward Using H.450.2 and H.450.3 with Cisco SRST 3.0” section on
page 200 for configuration instructions.
SUMMARY STEPS
2. xmlschema schema-url
3. exit
DETAILED STEPS

Example:
Step 2 xmlschema schema-url Specifies the URL for an XML API schema to be used with
this Cisco Unified SRST system.
Example: • schema-url: Local or remote URL as defined in
Router(config-cm-fallback)# xmlschema RFC 2396.
http://server2.example.com/
schema/schema1.xsd
Example:
Configuration Examples for Call Handling

• Example: Monitoring the Status of Key Expansion Modules, page 228
• Example: Configuring Voice Hunt Groups in Cisco Unified SIP SRST, page 229
Example: Monitoring the Status of Key Expansion Modules

Show commands are used to monitor the status and other details of Key Expansion Modules (KEMs).
The following example demonstrates how the show voice register all command displays KEM details
with all the Cisco Unified CME configurations and registration information:
show voice register all
VOICE REGISTER GLOBAL
=====================
CONFIG [Version=9.1]

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Where to Go Next
========================
............
Pool Tag 5
Config:
Mac address is B4A4.E328.4698
Type is 9971 addon 1 CKEM
Number list 1 : DN 2
Number list 2 : DN 3
Proxy Ip address is 0.0.0.0
DTMF Relay is disabled
Call Waiting is enabled
DnD is disabled
Video is enabled
Camera is enabled
Busy trigger per button value is 0
keep-conference is enabled
registration expires timer max is 200 and min is 60
kpml signal is enabled
Lpcor Type is none
The following example demonstrates how the show voice register pool type command displays all the
phones configured with add-on KEMs in Cisco Unified CME:
Router# show voice register pool type CKEM
Pool ID IP Address Ln DN Number State
==== =============== =============== == === ==================== ============
4 B4A4.E328.4698 9.45.31.111 1 4 5589$ REGISTERED
Example: Configuring Voice Hunt Groups in Cisco Unified SIP SRST

The following example shows how to configure longest-idle hunt group 20 with pilot number 4701, final
number 5000, and 6 numbers in the list. After a call is redirected six times (makes 6 hops), it is redirected
to the final number 5000.
Router(config-voice-hunt-group)# pilot 4701
Router(config-voice-hunt-group)# list 4001, 4002, 4023, 4028, 4045, 4062
Router(config-voice-hunt-group)# final 5000
Where to Go Next
If you need to configure security, see the “Configuring Secure SRST for SCCP and SIP” section on
page 231, or if you need to configure voicemail, see the “Integrating Voicemail with Cisco Unified
SRST” section on page 321. If you need to configure video parameters, see the “Setting Video
Parameters” section on page 345. If you do not need any of those features, go to the “Monitoring and
Maintaining Cisco Unified SRST” section on page 359.
SRST Feature Overview” section on page 1chapter.

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Where to Go Next

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CHAPTER 9
Configuring Secure SRST for SCCP and SIP
The Secure SRST adds security functionality to the Unified SRST.
Contents
This chapter describes new Secure SRST security features such as authentication, integrity, and media
encryption.
• Prerequisites for Configuring Secure SRST, page 231
• Restrictions for Configuring Secure SRST, page 232
• Information About Configuring Secure SRST, page 234
• How to Configure Secure Unified SRST, page 245
• Additional References, page 299
• Command Reference, page 301
• Feature Information for Secure SCCP and SIP SRST, page 302
Prerequisites for Configuring Secure SRST

General
• Secure Cisco Unified IP phones supported in secure SCCP and SIP SRST must have the
Certification Authority (CA) or third-party certificates installed, and encryption enabled. For more
information on CA server authentication, see Autoenrolling and Authenticating the Secure Cisco
Unified SRST Router to the CA Server, page 248.
• The SRST router must have a certificate; a certificate can be generated by a third party or by the
Cisco IOS certificate authority (CA). The Cisco IOS CA can run on the same gateway as Cisco
Unified SRST. Over the TLS channel (port 2445), automated certificate exchange happens between
the Unified SRST router and the Cisco Unified Communications Manager. However, the phone
certificate exchange to Unified SRST through Unified Communications Manager has to be
downloaded manually on the Unified SRST router.
• Certificate trust lists (CTLs) on Cisco Unified Communications Manager must be enabled.

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Chapter 9 Configuring Secure SRST for SCCP and SIP
Restrictions for Configuring Secure SRST
• It is mandatory to configure the command supplementary-service media-renegotiate under voice

service voip configuration mode to enable the supplementary features supported on Unified Secure
SRST.
Public Key Infrastructure on Secure SRST

• Set the clock, either manually or by using Network Time Protocol (NTP). Setting the clock ensures
synchronicity with Cisco Unified Communications Manager.
• Enable the IP HTTP server (Cisco IOS processor) with the ip http server command, if not already
enabled. For more information on public key infrastructure (PKI) deployment, see the Cisco IOS
Certificate Server feature.
• If the certificate server is part of your startup configuration, you may see the following messages
during the boot procedure:
% Failed to find Certificate Server's trustpoint at startup
% Failed to find Certificate Server's cert.
These messages are informational messages and indicate a temporary inability to configure the
certificate server because the startup configuration has not been fully parsed yet. The messages are
useful for debugging, in case the startup configuration is corrupted.
You can verify the status of the certificate server after the boot procedure using the show crypto pki
server command.
Supported Cisco Unified IP Phones, Platforms, and Memory Requirements

• For a list of supported Cisco Unified IP Phones, routers, network modules, and codecs for secure
SRST, see the Cisco Unified Survivable Remote Site Telephony Compatibility Information feature.
• For the most up-to-date information about the maximum number of Cisco Unified IP Phones, the
maximum number of directory numbers (DNs) or virtual voice ports, and memory requirements, see
the Cisco Unified SRST 12.3 Supported Firmware, Platforms, Memory, and Voice Products feature.

General
• Cryptographic software features (“k9”) are under export controls. This product contains
cryptographic features and is subject to United States and local country laws governing import,
export, transfer, and use. Delivery of Cisco cryptographic products does not imply third-party
authority to import, export, distribute or use encryption. Importers, exporters, distributors and, users
are responsible for compliance with U.S. and local country laws. By using this product you agree to
comply with applicable laws and regulations. If you are unable to comply with U.S. and local laws,
return this product immediately.
A summary of U.S. laws governing Cisco cryptographic products may be found at the following
URL:
http://www.cisco.com/wwl/export/crypto/tool/
If you require further assistance, please contact us by sending email to [email protected].
• When a Secure Real-Time Transport Protocol (SRTP) encrypted call is made between Cisco Unified
IP Phone endpoints or from a Cisco Unified IP Phone to a gateway endpoint, a lock icon is displayed
on the IP phones. The lock indicates security only for the IP leg of the call. Security of the PSTN
leg is not implied.

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SCCP SRST
• Secure SCCP SRST is supported only within the scope of a single router.
• Cisco 4000 Series Integrated Services Routers support Secure SCCP SRST only on Unified SRST
12.3 and later releases. For Secure SCCP support on Unified SRST 12.3 Release:
– Secure Cisco Jabber is not supported.
– SRTP passthrough is not supported.
– SDP Passthrough is not supported.
– Video Calling is not supported.
– Transcoding is not supported.
– Hardware Conferencing is not supported (Only Software Conferencing is supported).
– Secure Multicast MOH is not supported (Multicast MOH stays active, but non-secure).
– Live MOH is not supported.
– Secure H.323 is not supported.
– Hot Standby Routing Protocol (HSRP) is not supported.
– T.38 Fax Relay and Modem Relay is not supported for Unified Secure SRST.
• For call support on Voice Gateway introduced as part of Unified SRST 12.3 Release:
– Speed Dial is not supported.
– For a pure SCCP shared line, Hold and Remote Resume is not supported from an analog phone.
– Full Blind Transfer mode (Configured with the CLI command transfer-system full-blind) is
not supported.
– Consider a call between two Analog Voice Gateways (VG A and VG B) registered on Unified
Secure SRST as SCCP endpoints. If a call is already put on hold from the VG B endpoint (could
be an SCCP phone too), then VG A (has to be an Analog Voice Gateway) cannot put the same
call on hold (double hold). For more information, see CSCvi15203.
– For three-way software conference related behavior and limitations, see Three-way Software
Conferencing for Secure SCCP, Unified SRST Release 12.3, page 236.
SIP SRST
• Cisco 4000 Series Integrated Services Router supports Secure SIP SRST only on Unified SRST 12.1
and later releases.
• SRTP passthrough is not supported.
• SDP Passthrough is not supported.
• Video Calling is not supported.
• Transcoding is not supported.
• Hardware Conferencing is not supported (Only BIB Conferencing is supported).
• It is mandatory to configure security-policy secure under voice register global configuration
mode. Non-Secure endpoints cannot register when security-policy secure is configured. As such,
mixed deployments of secure and non-secure endpoints is not possible.

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Information About Configuring Secure SRST

• Benefits of Secure SRST, page 234
• Secure SIP SRST Support on Cisco 4000 Series Integrated Services Router, page 234
• Secure SCCP SRST on Cisco 4000 Series Integrated Services Router, page 235
• Cisco IP Phones Clear-Text Fallback During Non-Secure SRST, page 238
• Signaling Security on Unified SRST - TLS, page 238
• Media Security on Unified SRST - SRTP, page 242
• Establishment of Secure Cisco Unified SRST to the Cisco Unified IP Phone, page 243
• Secure SRST Authentication and Encryption, page 244
Benefits of Secure SRST

Secure Cisco Unified IP phones that are located at remote sites and that are attached to gateway routers
can communicate securely with Cisco Unified Communications Manager using the WAN. But if the
WAN link or Cisco Unified Communications Manager goes down, all communication through the
remote phones becomes non-secure. To overcome this situation, gateway routers can now function in
secure SRST mode, which activates when the WAN link or Cisco Unified Communications Manager
goes down. When the WAN link or Cisco Unified Communications Manager is restored, Cisco
Unified Communications Manager resumes secure call-handling capabilities.
Secure SRST provides new Cisco Unified SRST security features such as authentication, integrity, and
media encryption. Authentication provides assurance to one party that another party is whom it claims
to be. Integrity provides assurance that the given data has not been altered between the entities.
Encryption implies confidentiality; that is, that no one can read the data except the intended recipient.
These security features allow privacy for Cisco Unified SRST voice calls and protect against voice
security violations and identity theft.
SRST security is achieved when:
• End devices are authenticated using certificates.
• Signaling is authenticated and encrypted using Transport Layer Security (TLS) for TCP.
• A secure media path is encrypted using Secure Real-Time Transport Protocol (SRTP).
• Certificates are generated and distributed by a CA.
Secure SIP SRST Support on Cisco 4000 Series Integrated Services Router
For Unified SRST 12.1 and later releases, Secure SIP SRST support is introduced on the Cisco 4000
Series Integrated Services Router. As a part of the Secure SIP SRST feature on Unified SRST Release
12.1, support is provided for calls with the Transport Layer Security protocols (TLS) versions up to 1.2.
Also, TLS 1.2 exclusivity is supported as part of Unified SRST Release 12.1.
The Cisco IP Phone 7800 Series and Cisco IP Phone 8800 Series is supported on the Unified Secure SIP
SRST Release 12.1 configured on Cisco 4000 Series Integrated Services Routers.
For Secure SIP SRST to be supported on Cisco 4000 Series Integrated Services Routers, you need to
enable the following technology package licenses on the router:
• security

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• uck9
Note For Unified SRST 12.2 and previous releases, only SIP phones are supported on the Cisco 4000 Series
Integrated Services Router for Secure SIP SRST. For Unified SRST 12.3 and later releases, a mixed
deployment of SIP and SCCP phones are supported on the Cisco 4000 Series Integrated Services
Routers.
Secure Music On Hold for Unified Secure SRST (SIP)

From Unified SRST Release 12.1, support is introduced for Secure Music On Hold (MOH), as part of
the Secure SIP SRST solution on Cisco 4000 Series Integrated Services Router. For a Secure SIP call
that is put on hold, playback of Flash-based G.729 and G.711 codec format MOH files are supported.
Live MOH and transcoded MOH are not supported as part of Secure MOH feature support.
Note If the CLI command srtp pass-thru is configured under the dial peer voice configuration mode, Secure
MOH does not work.
Secure SCCP SRST on Cisco 4000 Series Integrated Services Router

For Unified SRST 12.3 and later releases, Secure SCCP SRST support is introduced on the Cisco 4000
Series Integrated Services Router. As a part of the Secure SCCP SRST feature on Unified SRST Release
12.3, support is provided for calls with the Transport Layer Security protocols (TLS) versions up to 1.2.
Also, TLS 1.2 exclusivity is supported as part of Unified SRST Release 12.3. For more information on
the TLS protocol support introduced for Secure SCCP in Unified SRST Release 12.3, see SRST Routers
and the TLS Protocol, page 239.
For Secure SCCP SRST to be supported on Cisco 4000 Series Integrated Services Routers, you need to
enable the following technology package licenses on the router:
• security
• uck9
The Cisco Unified IP Phone 6961 and Cisco Unified IP Phone 7962G is supported on the Unified Secure
SCCP SRST Release 12.3 configured on Cisco 4000 Series Integrated Services Routers. Also, analog
phones are supported for analog Voice Gateways as part of Unified Secure SCCP SRST Release 12.3.
For more information on support introduced on Voice Gateways, see Secure SCCP SRST for Analog
Voice Gateways, page 235.
Secure SCCP SRST for Analog Voice Gateways

For Unified SRST 12.3 and later releases on a Cisco 4000 Series Integrated Services Router, Secure
SCCP support is introduced for the following Voice Gateways:
• Cisco VG202 Analog Voice Gateway
• Cisco VG202XM Analog Voice Gateway
• Cisco VG204XM Analog Voice Gateway

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As a part of the Secure SCCP SRST feature on Unified SRST Release 12.3, Transport Layer Security
protocols (TLS) versions up to 1.2, and TLS 1.2 exclusivity is supported for Cisco VG202XM Analog
Voice Gateway, Cisco VG204XM Analog Voice Gateway, Cisco VG310 Analog Voice Gateway, and
Cisco VG320 Analog Voice Gateway.
For more information on configuring the Voice Gateways, see Supplementary Services Features for FXS
Ports on Cisco IOS Voice Gateways Configuration Guide.
Note Cisco VG202 Analog Voice Gateway, Cisco VG204 Analog Voice Gateway, and Cisco VG224 Analog
Voice Gateway only support Transport Layer Security protocols (TLS) version 1.0.
Feature Access Support for Analog Phones on Voice Gateway
For a user in basic call mode on analog phones on a voice gateway, you need to:
• Press hookflash for the first dial tone to dial an extension number to connect to a second call.
• When the second call is established, press hookflash for feature tone and #4 to transfer the call.
• When the second call is established, press hookflash for feature tone and #3 to initiate a three-way
conference.
• During a three-party conference, press hookflash to drop the last conferee in Unified
Communications Manager. For Unified Secure SRST, press hookflash to get feature tone and dial
#2 to drop the last active party in the conference.
• When the second call is established, press hookflash for feature tone and #5 to toggle back to the
previous call party.
Secure Music On Hold for Secure Unified SRST (SCCP)

From Unified SRST Release 12.3, support is introduced for Secure Music On Hold (MOH), as part of
the Secure SCCP SRST functionality on Cisco 4000 Series Integrated Services Router. For a Secure
SCCP call that is put on hold, playback of Flash-based G.729 and G.711 codec format MOH files are
supported. Live MOH and transcoded MOH are not supported as part of Secure MOH feature support.
Also, Multicast MOH is supported as non-secure on fallback from Cisco Unified Communications
Manager to Unified Secure SRST.
Three-way Software Conferencing for Secure SCCP, Unified SRST Release 12.3
From Unified SRST Release 12.3, three-way software conferencing is supported for Secure SCCP
endpoints on Cisco 4000 Series Integrated Services Routers. The audio codec supported as part of the
three-way software conferencing for Unified SRST 12.3 Release is G.711. The support is introduced for
Secure SCCP phones and Secure SCCP endpoints registered on Cisco Analog Voice Gateways.
Three-way software conferencing is supported for a pure SCCP deployment (only involving SCCP
endpoints), and a mixed deployment of secure SCCP and SIP phones. The SCCP phones such as Cisco
Unified IP Phone 7962, Cisco Unified IP Phone 6961, and Cisco Unified IP Phone 7975 are supported
as part of this deployment. For the mixed deployment, the Cisco IP Phone 7800 Series and Cisco IP
Phone 8800 Series SIP phones are supported. Three-way Software Conference is supported on TDM
trunks, for SIP and SCCP endpoints on Unified Secure SRST.

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You can set a limit for the maximum number of conferences that are supported. Configure the CLI
command max-conferences under call-manager-fallback configuration mode to set the maximum
number of conferences supported. If you do not set the maximum number of supported conferences using
the command max-conferences, the limit is set to the default value of 8.
Router(config-cm-fallback)#max-conferences ?
<1-16> Maximum conferences to support
For a three-way software conference supported on Secure Unified SRST:

• When a secure SCCP endpoint initiates the conference or the SCCP endpoint is a conference host,
the conference is created. The three-way software conference is hosted on a Unified Secure SRST
router.
• When a secure SIP endpoint initiates the conference, the three-way software conference is hosted
on the SIP phone.
• When the conference host puts the call on hold, the other participants in the three-way software
conference will hear Music On Hold until the call is resumed by the host. Multicast MOH is played
for an SCCP endpoint, whereas Unicast MOH is played for SIP endpoints.
• When the three-way software conference host is an Analog Voice Gateway endpoint, the host cannot
place the conference on hold. The three-way software conference can be put on hold only by SCCP
or SIP endpoints.
• When any of the conference participants (apart from the host) put the call on hold, the other
participants in the three-way software conference can continue to talk.
• For a three-way software conference on Unified SRST for Secure SCCP endpoints, the conference
participants can transfer the call. The conference host cannot transfer the conference call. During an
alert transfer, the other two participants can continue to talk without media interruption.
• Conference Cascading is not supported for a three-way software conference on Unified Secure
SRST.
• Consider a three-way software conference hosted by an Analog Voice Gateway endpoint, with SCCP
A and SCCP B as the second and third conference participants, respectively. In a scenario where
SCCP B places the call on hold and the conference host tries to commit the conference using
hookflash (followed by FAC), the call with SCCP B is terminated and conference attempt fails.
• Consider a scenario where an Analog Phone (AP 1) registered to the Analog Voice Gateway places
a call to SCCP Phone (SCCP 1) registered to Secure SCCP SRST. After placing SCCP 1 on hold,
AP 1 places a call to the third participant, SCCP Phone (SCCP 2), that is registered to the same
Secure SRST. Three-way Software Conferencing is established. When SCCP 2 tries to perform an
alert transfer to a phone (SIP 3/ SCCP 3) and it goes unanswered, the three-way conference is lost
and it becomes a one-to-one call between AP 1 and SCCP 1. Any further attempt by AP 1 to establish
a three-way software conference with another phone (SCCP 4) is not supported in this scenario.
Note If the failed alert transfer is by SCCP 1, then any further attempt to establish a three-way
software conference with another phone will be supported.
Feature Support for Secure SRST (SCCP), Unified SRST Release 12.3
The Secure SCCP SRST on Cisco 4000 Series Integrated Services Routers and the Analog Voice
Gateways introduced as part of Unified SRST Release 12.3, offers the following basic and
supplementary call processing support. For a list of restrictions for Unified SRST 12.3 and later releases
on Cisco Integrated Services Router Generation 2, see Restrictions for Configuring Secure SCCP SRST,

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page 270.
• Call Forward (Busy, No-answer, All)
• Call Hold or Resume
• Redial
• Secure MOH (Flash Based)
• Speed Dial (Only for Secure SCCP phones on Cisco 4000 Series Integrated Services Router)
• Secure Three-party Software Conference
• SIP trunks (Secure and Non-secure)
• TDM trunks
• Call Transfer (Alert, Consult, and Blind)
• Shared Line (Only for a pure SCCP-to-SCCP shared line. Mixed shared line is not supported.)
• Caller ID
• Call Waiting
• Media Inactivity
The following features are supported for Analog Voice Gateways for Fax and Modem calls on analog
FXS ports:
• Fax Passthrough
• Modem Passthrough
Cisco IP Phones Clear-Text Fallback During Non-Secure SRST

• Cisco Unified SRST versions before 12.3(14)T are not capable of supporting secure connections or
have security enabled. If an SRST router is not capable of SRST as a fallback mode—that is, it is
not capable of completing a TLS handshake with Cisco Unified Communications Manager—its
certificate is not added to the configuration file of the Cisco IP phone. The absence of a Cisco
Unified SRST router certificate causes the Cisco Unified IP phone to use nonsecure (clear-text)
communication when in Cisco Unified SRST fallback mode. The capability to detect and fallback
in clear-text mode is built into Cisco Unified IP phone firmware. See Media and Signaling
Authentication and Encryption Feature for Cisco IOS MGCP Gateways for more information on
clear-text mode.
Signaling Security on Unified SRST - TLS

• SRST Routers and the TLS Protocol, page 239
• Certificates Operation on Secure SRST, page 240
• Certificates Transport from CUCM to Secure SRST, page 242
SRST Routers and the TLS Protocol

Transport Layer Security (TLS) Version 1.0 provides secure TCP channels between Cisco Unified IP
phones, secure Cisco Unified SRST Routers, and Cisco Unified Communications Manager. The TLS
process begins with the Cisco Unified IP Phone establishing a TLS connection when registering with

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Cisco Unified Communications Manager. Assuming that Cisco Unified Communications Manager is
configured to fall back to Cisco Unified SRST, the TLS connection between the Cisco Unified IP Phones
and the secure Cisco Unified SRST Router is also established. If the WAN link or Cisco
Unified Communications Manager fails, call control reverts to the Cisco Unified SRST router.
From Unified Secure SIP SRST Release 12.1, support is introduced for SIP-to-SIP calls with Transport
Layer Security up to TLS Version,1.2. For configuring TLS 1.2 exclusivity functionality, you need to
configure the command transport tcp tls v1.2 under sip-ua configuration mode. When you configure
TLS 1.2 exclusivity on the Secure SIP SRST, any registration attempt by phones using lower versions of
TLS (1.0, 1.1) are rejected.
Before Unified SRST Release 12.3, support is available only for TLS 1.0 version with Unified Secure
SCCP SRST. For Unified Secure SCCP SRST Release 12.3 and later releases, support is introduced for
Transport Layer Security up to TLS version 1.2. To configure a specific TLS version or TLS 1.2
exclusivity for Unified Secure SCCP SRST, you need to configure transport-tcp-tls under
call-manager-fallback. When transport-tcp-tls is configured without specifying a version, the default
behavior of the CLI command is enabled. In the default form, all the TLS versions (except TLS 1.0) are
supported for this CLI command.
For Secure SIP and Secure SCCP endpoints that do not support TLS version 1.2, you need to configure
TLS 1.0 for the endpoints to register to Unified Secure SRST 12.3 (Cisco IOS XE Fuji Release 16.9.1).
This also means that endpoints which support 1.2 should also use the 1.0 suites.
For TLS 1.0 support on Cisco IOS XE Fuji Release 16.9.1 for SCCP endpoints, you need to specifically
configure:
• transport-tcp-tls v1.0 under call-manager-fallback configuration mode
For TLS 1.0 support on Cisco IOS XE Fuji Release 16.9.1 for pure SIP and mixed deployment scenarios,
you need to specifically configure:
• transport-tcp-tls v1.0 under sip-ua configuration mode
From Cisco IOS XE Fuji Release 16.9.1 Release, the security certificate exchange between Unified
Secure SRST Release 12.3 and Unified Communications Manager does not support TLS version 1.0.
Note Unified Communications Manager Release 11.5.1SU3 is the minimum version required to support
security certificate exchange with Unified Secure SRST Release 12.3 (Cisco IOS XE Fuji Release
16.9.1).
For more information on the transport-tcp-tls command, see Cisco Unified SRST Command Reference
(All Versions).
Note SCCP phones and the Analog Voice Gateways VG202, VG204, and VG224 support only TLS version
1.0. For Unified Secure SRST 12.3 Release and later, TLS versions 1.1 and 1.2 are supported only for
Cisco Analog Voice Gateways VG202XM, VG204XM, VG310, and VG320.
You can configure transport-tcp-tls under call-manager-fallback for Unified Secure SCCP SRST as
follows:
Router(config-cm-fallback)#transport-tcp-tls ?
v1.0 Enable TLS Version 1.0

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Note When you configure TLS 1.2 exclusivity on the Secure SCCP SRST, any new connection attempt by
phones using lower TLS versions (1.0, 1.1) are rejected. Also, the existing TLS connections will be in
tact, until the connection is reset.
For Unified Secure SCCP SRST Release 12.3 and later releases, Analog Voice Gateways can register
their SCCP endpoints with Transport Layer Security versions up to 1.2 (TLS 1.0, 1.1, and 1.2). For
support of a specific TLS version on the analog voice gateways for Unified SRST Release 12.3 and later,
you need to configure stcapp security tls-version under stcapp:
enable
configure terminal
stcapp security tls-version ?
exit
--
VG(config)#stcapp security tls-version ?
Certificates Operation on Secure SRST

• Cisco Unified SRST Routers and PKI, page 240
• Cisco IOS Credentials Server on Secure SRST Routers, page 241
• Generating a Certificate for the Credentials Server, page 242
Cisco Unified SRST Routers and PKI
The transfer of certificates between a Cisco Unified SRST router and Cisco Unified Communications
Manager is mandatory for secure SRST functionality. Public key infrastructure (PKI) commands are
used to generate, import, and export the certificates for secure Cisco Unified SRST. Table 9-1 shows the
secure SRST-supported Cisco Unified IP Phones and the appropriate certificate for each phone. The
“Additional References” section on page 299 contains information and configurations about generating,
importing, and exporting certificates that use PKI commands.
Note Certificate text can vary depending on your configuration. You may also need CAP-RTP-00X or
CAP-SJC-00X for older phones that support manufacturing installed certificate (MIC).
Note Cisco supports Cisco IP Phones 7900 series phone memory reclamation phones that use MIC or locally
significant certificate (LSC) certificates.
Table 9-1 Supported Cisco Unified IP Phones and Certificates

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Cisco Unified IP Phone 7940 Cisco Unified IP Phone 7960 Cisco Unified IP Phone 7970
The phone receives locally significant The phone receives locally The phone contains a manufacturing
certificate (LSC) from Certificate significant certificate (LSC) from installed certificate (MIC) used for
Authority Proxy Function (CAPF) in Certificate Authority Proxy Function device authentication. If the Cisco
Distinguished Encoding Rules (DER) (CAPF) in Distinguished Encoding 7970 implements MIC, two public
format. Rules (DER) format. certificate files are needed:
• 59fe77ccd.0 • 59fe77ccd.0 • CiscoCA.pem (Cisco Root CA,
used to authenticate the
The filename may change based The filename may change based
certificate.)
on the CAPF certificate subject on the CAPF certificate subject
name and the CAPF certificate name and the CAPF certificate Note The name of the
issuer. issuer. manufacturing certificate can
If Cisco If Cisco vary depending on your
Unified Communications Unified Communications configuration.
Manager is using a third-party Manager is using a third-party
• a69d2e04.0, in Privacy Enhanced
certificate provider, there can be certificate provider, there can be
Mail (PEM) format
multiple .0 files (from two to multiple .0 files (from two to
ten). Each .0 certificate file must ten). Each .0 certificate file must If Cisco Unified Communications
be imported individually during be imported individually during Manager is using a third-party
the configuration. the configuration. certificate provider, there can be
multiple .0 files (from two to ten).
Manual enrollment supported only. Manual enrollment supported only.
Each .0 certificate file must be
imported individually during the
configuration.
Manual enrollment supported only.
Cisco IOS Credentials Server on Secure SRST Routers
Secure SRST introduces a credentials server that runs on a secure SRST router. When the client,
Cisco Unified Communications Manager, requests a certificate through the TLS channel, the credentials
server provides the SRST router certificate to Cisco Unified Communications Manager.
Cisco Unified Communications Manager inserts the SRST router certificate in the Cisco Unified IP
Phone configuration file and downloads the configuration files to the phones. The secure Cisco Unified
IP Phone uses the certificate to authenticate the SRST router during fallback operations. The credentials
service runs on default TCP port 2445.
Three Cisco IOS commands configure the credentials server in call-manager-fallback mode:
• credentials
• ip source-address (credentials)
• trustpoint (credentials)
Two Cisco IOS commands provide credential server debugging and verification capabilities:
• debug credentials
• show credentials

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Generating a Certificate for the Credentials Server
In configuring the credentials server on the Unified Secure SRST, a certificate is required to complete
the "trustpoint <trustpoint name>" configuration entry.
To generate the certificate for Credentials Server, perform the following procedures:
• Autoenrolling and Authenticating the Secure Cisco Unified SRST Router to the CA Server,
page 248
• Enabling Credentials Service on the Secure Cisco Unified SRST Router, page 255
• Configuring SRST Fallback on Cisco Unified Communications Manager, page 266
Once the certificate is generated, fill in the name of the certificate (or the name of the trustpoint in IOS)
in the "trustpoint" entry.
This certificate for the Credentials Server on the Secure SRST will be seamlessly exported to the Cisco
Unified CM when requested in “Adding an SRST Reference to Cisco Unified Communications
Manager” section on page 265.
Certificates Transport from CUCM to Secure SRST

For more information about Certificates Transport from CUCM to Secure SRST, see “Importing Phone
Certificate Files in PEM Format to the Secure SRST Router” section on page 257.
Media Security on Unified SRST - SRTP

Media encryption, which uses Secure Real-Time Protocol (SRTP), ensures that only the intended
recipient can interpret the media streams between supported devices. Support includes audio streams
only.
If the devices support SRTP, the system uses a SRTP connection. If at least one device does not support
SRTP, the system uses an RTP connection. SRTP-to-RTP fallback may occur for transfers from a secure
device to a non-secure device for music-on-hold (MOH), and so on.
Note Secure SRST handles media encryption keys differently for different devices and protocols. All
phones that are running SCCP get their media encryption keys from SRST, which secures the
media encryption key downloads to phones with TLS encrypted signaling channels. Phones that
are running SIP generate and store their own media encryption keys. Media encryption keys that
are derived by SRST securely get sent through encrypted signaling paths to gateways over
IPSec-protected links for H.323.
Warning Before you configure SRTP or signaling encryption for gateways and trunks, Cisco strongly
recommends that you configure IPSec because Cisco H.323 gateways, and H.323/H.245/H.225
trunks rely on IPSec configuration to ensure that security-related information does not get sent
in the clear. Cisco Unified SRST does not verify that you configured IPSec correctly. If you do
not configure IPSec correctly, security-related information may get exposed.

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Establishment of Secure Cisco Unified SRST to the Cisco Unified IP Phone

Figure 9-1 shows the interworking of the credentials server on the SRST router, Cisco Unified
Communications Manager, and the Cisco Unified IP Phone. Table 9-2 describes the establishment of
secure SRST to the Cisco Unified IP Phone.
Figure 9-1 Interworking of Credentials Server on SRST Router, Cisco Unified Communications Manager,
and Cisco Unified IP Phone
Credentials server
1. Cisco Unified Communications Manager running on secure
Cisco Unified requests the Cisco Unified SRST certificate Cisco Unified
Communications from the credentials server. SRST router
Manager/client
WAN
155100
2. The credentials server responds
with the certificate.
3. Cisco Unified Communications Manager inserts

the certificate in the phone configuration file.
IP
Cisco IP phone
Table 9-2 Establishing Secure SRST
Mode Process Description or Detail

Regular Mode The Cisco Unified IP Phone configures DHCP and —
gets the TFTP server address.
The Cisco Unified IP Phone retrieves a CTL file The CTL file contains the certificates that the phone
from the TFTP server. should trust.
The Cisco IP Phone opens a Transport Layer Cisco Unified Communications Manager exports
Security (TLS) protocol channel and registers to secure Cisco Unified SRST router information and
Cisco Unified Communications Manager. the Cisco Unified SRST router certificate to the Cisco
Unified IP phone. The phone places the certificate
into its configuration. Once the phone has the Cisco
Unified SRST certificate, the Cisco Unified SRST
router is considered secure. See Figure 9-1.
If the Cisco Unified IP Phone is configured as The connection to the SRST router happens
“authenticated” or “encrypted” and Cisco automatically, assuming there is not a secondary
Unified Communications Manager is configured Cisco Unified Communications Manager and Cisco
in mixed mode, the phone looks for an SRST Unified SRST is configured as the backup device. See
certificate in its configuration file. If it finds an Figure 9-1.
SRST certificate, it opens a standby TLS Cisco Unified Communications Manager should be
connection to the default port. The default port is configured in mixed mode, which is its secure mode.
the Cisco Unified IP Phone TCP port plus 443;
that is, port 2443 on a Cisco Unified SRST router.

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In case of WAN failure, the Cisco Unified IP Phone starts Cisco Unified SRST registration.
SRST Mode The Cisco Unified IP Phone registers with the —
SRST router at the default port for secure
communications.
Secure SRST Authentication and Encryption

Figure 9-2 illustrates the process of secure SRST authentication and encryption, and Table 9-3 describes
the process.
Figure 9-2 Secure Cisco Unified SRST Authentication and Encryption
Cisco IOS router CA

Cisco Unified or third-party CA
CAPF TFTP
Communications Manager
4
Cisco Unified
SRST cert
2 4 5 3 1
7940/7960 Cisco Unified
LSC SRST cert
7970 SEPMACxxxx.cnf.xml
MIC Credentials
service
6 TLS handshake
IP
V
155101
IP phone 6b 6a
Cisco Unified
LSC/MIC Cisco Unified SRST
SRST cert
Table 9-3 Overview of the Process of Secure SRST Authentication and Encryption
Process Steps Description or Detail

1. The CA server, whether it is a Cisco IOS router CA or a third-party CA, issues a
device certificate to the SRST gateway, enabling credentials service. Optionally, the
certificate can be self-generated by the SRST router using a Cisco IOS CA server.
The CA router is the ultimate trustpoint for the Certificate Authority Proxy Function
(CAPF). For more information on CAPF, see Cisco Communications Manager
Security Guide.
2. The CAPF is a process where supported devices can request a locally significant
certificate (LSC). The CAPF utility generates a key pair and certificate that is specific
for CAPF, copies this certificate to all Cisco Unified Communications Manager
servers in the cluster, and provides the LSC to the Cisco Unified IP Phone.
An LSC is required for Cisco Unified IP Phones that do not have a manufacturing
installed certificate (MIC). The Cisco 7970 is equipped with a MIC and therefore does
not need to go through the CAPF process.
3. Cisco Unified Communications Manager requests the SRST certificate from
credentials server, and the credentials server responds with the certificate.

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How to Configure Secure Unified SRST
Table 9-3 Overview of the Process of Secure SRST Authentication and Encryption (continued)
Process Steps Description or Detail

4. For each device, Cisco Unified CM uses the TFTP process and inserts the certificate
into the SEPMACxxxx.cnf.xml configuration file of the Cisco Unified IP Phone.
5. Cisco Unified CM provides the PEM format files that contain phone certificate
information to the Cisco Unified SRST router. Providing the PEM files to the Cisco
Unified SRST router is done manually. See “Cisco IOS Credentials Server on Secure
SRST Routers” section on page 241 for more information.
When the Cisco Unified SRST router has the PEM files, the Cisco Unified SRST
Router can authenticate the IP phone and validate the issuer of the IP phones
certificate during the TLS handshake.
6. The TLS handshake occurs, certificates are exchanged, and mutual authentication and
registration occurs between the Cisco Unified IP Phone and the Cisco Unified SRST
Router.
a. The Cisco Unified SRST Router sends its certificate, and the phone validates the
certificate to the certificate that it received from Cisco Unified CM in Step 4.
b. The Cisco Unified IP Phone provides the Cisco Unified SRST Router the LSC or
MIC, and the router validates the LSC or MIC using the PEM format files that it was
provided in Step 5.
Note The media is encrypted automatically after the phone and router certificates are exchanged and the TLS
connection is established with the SRST router.

The following configuration sections ensure that the secure Cisco Unified SRST Router and the Cisco
Unified IP Phones can request mutual authentication during the TLS handshake. The TLS handshake
occurs when the phone registers with the Cisco Unified SRST Router, either before or after the WAN
link fails.
• Preparing the Cisco Unified SRST Router for Secure Communication, page 246
• Configuring Cisco Unified Communications Manager to the Secure Cisco Unified SRST Router,
page 265
• Enabling SRST Mode on the Secure Cisco Unified SRST Router, page 268
• Configuring Secure SCCP SRST, page 270
• Configuring Secure SIP Call Signaling and SRTP Media with Cisco SRST, page 284g

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Preparing the Cisco Unified SRST Router for Secure Communication

The following tasks prepare the Cisco Unified SRST Router to process secure communications.
• Configuring a Certificate Authority Server on a Cisco IOS Certificate Server, page 246 (optional)
• Autoenrolling and Authenticating the Secure Cisco Unified SRST Router to the CA Server,
page 248 (required)
• Disabling Automatic Certificate Enrollment, page 252 (required)
• Verifying Certificate Enrollment, page 253 (optional)
• Enabling Credentials Service on the Secure Cisco Unified SRST Router, page 255 (required)
• Troubleshooting Credential Settings, page 257
• Importing Phone Certificate Files in PEM Format to the Secure SRST Router, page 257
Configuring a Certificate Authority Server on a Cisco IOS Certificate Server

For Cisco Unified SRST Routers to provide secure communications, there must be a CA server that
issues the device certificate in the network. The CA server can be a third-party CA or one generated from
a Cisco IOS certificate server.
The Cisco IOS certificate server provides a certificate generation option to users who do not have a
third-party CA in their network. The Cisco IOS certificate server can run on the SRST router or on a
different Cisco IOS router.
If you do not have a third-party CA, full instructions on enabling and configuring a CA server can be
found in the Cisco IOS Certificate Server documentation. A sample configuration is provided below.
SUMMARY STEPS
1. crypto pki server cs-label

2. database level {minimal | names | complete}
3. database url root-url
4. issuer-name DN-string
5. grant auto
6. no shutdown

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DETAILED STEPS

Step 1 crypto pki server cs-label Enables the certificate server and enters certificate server
configuration mode.
Example: Note If you manually generated an RSA key pair, the
Router (config)# crypto pki server srstcaserver cs-label argument must match the name of the key
pair.
For more information on the certificate server, see the

Cisco IOS Certificate Server documentation.
Step 2 database level {minimal | names | complete} Controls what type of data is stored in the certificate
enrollment database.
Example: • minimal: Enough information is stored only to
Router (cs-server)# database level complete continue issuing new certificates without conflict; this
is the default.
• names: In addition to the information given in the
minimal level, the serial number and subject name of
each certificate are stored.
• complete: In addition to the information given in the
minimal and names levels, each issued certificate is
written to the database.
Note The complete keyword produces a large amount of
information; if it is issued, you should also specify
an external TFTP server on which to store the data
using the database url command.
Step 3 database url root-url Specifies the location where all database entries for the
certificate server will be written. After you create a
certificate server using the crypto pki server command, use
Example:
Router (cs-server)# database url nvram
this command to specify a combined list of all the
certificates that have been issued. The root-url argument
specifies the location where database entries are written.
• The default location for the database entries to be
written is flash; however, NVRAM is recommended for
this task.
Step 4 issuer-name DN-string Sets the CA issuer name to the specified distinguished name
(DN-string). The default value is as follows:
Example: issuer-name CN=cs-label.
Router (cs-server)# issuer-name CN=srstcaserver

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Step 5 grant auto Allows an automatic certificate to be issued to any
requestor.
Example: • This command is used only during enrollment and will
Router (cs-server)# grant auto be removed in the “Disabling Automatic Certificate
Enrollment” section on page 252.
Step 6 no shutdown Enables the Cisco IOS certificate server.
• You should issue this command only after you have
Example: completely configured your certificate server.
Router (cs-server)# no shutdown
Examples
The following example reflects one way of generating a CA:

Router(config)# crypto pki server srstcaserver
Router(cs-server)# database level complete
Router(cs-server)# database url nvram
Router(cs-server)# issuer-name CN=srstcaserver
Router(cs-server)# grant auto
% This will cause all certificate requests to be automatically granted.

Are you sure you want to do this? [yes/no]: y
Router(cs-server)# no shutdown
% Once you start the server, you can no longer change some of
% the configuration.
Are you sure you want to do this? [yes/no]: y
% Generating 1024 bit RSA keys ...[OK]
% Certificate Server enabled.
Autoenrolling and Authenticating the Secure Cisco Unified SRST Router to the CA Server
The secure Cisco Unified SRST Router needs to define a trustpoint; that is, it must obtain a device
certificate from the CA server. The procedure is called certificate enrollment. Once enrolled, the secure
Cisco Unified SRST Router can be recognized by Cisco Unified Communications Manager as a secure
SRST router.
There are three options to enroll the secure Cisco Unified SRST Router to a CA server: autoenrollment,
cut and paste, and TFTP. When the CA server is a Cisco IOS certificate server, autoenrollment can be
used. Otherwise, manual enrollment is required. Manual enrollment refers to cut and paste or TFTP.
Use the enrollment url command for autoenrollment and the crypto pki authenticate command to
authenticate the SRST router. Full instructions for the commands can be found in the Certification
Authority Interoperability Commands documentation. An example of autoenrollment is available in the
Certificate Enrollment Enhancements feature. A sample configuration is provided in the “Examples”
section on page 251.
SUMMARY STEPS
1. crypto pki trustpoint name

2. rsakeypair keypair-label
3. enrollment url url
4. revocation-check method1

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5. exit
6. crypto pki authenticate name
7. crypto pki enroll name

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DETAILED STEPS

Step 1 crypto pki trustpoint name Declares the CA that your router should use and enters
ca-trustpoint configuration mode.
Example: • The name provided will be the same as the trustpoint
Router(config)# crypto pki trustpoint srstca name that will be declared in the “Enabling Credentials
Service on the Secure Cisco Unified SRST Router”
Step 2 rsakeypair keypair-label To specify a named Rivest, Shamir, and Adelman (RSA)
key pair for this trustpoint, use the rsakeypair command in
trustpoint configuration mode.
• For TLS 1.2 version, the RSA key length is set to 2048
Example: bits.
Router(config-trustp)# rsakeypair srstcakey
2048
Step 3 enrollment url url Specifies the enrollment parameters of your CA.
• url url: Specifies the URL of the CA to which your
Example: router should send certificate requests.
Router(ca-trustpoint)# enrollment url
http://10.1.1.22
• If you are using Cisco proprietary SCEP for enrollment,
url must be in the form http://CA_name, where
CA_name is the host Domain Name System (DNS)
name or IP address of the Cisco IOS CA.
• If you used the procedure documented in the
“Configuring a Certificate Authority Server on a Cisco
IOS Certificate Server” section on page 246, the URL
is the IP address of the certificate server router
configured in Step 1. If a third-party CA was used, the
IP address is to an external CA.
Step 4 revocation-check method1 Checks the revocation status of a certificate. The argument
method1 is the method used by the router to check the
revocation status of the certificate. For this task, the only
Example:
Router(ca-trustpoint)# revocation-check none
available method is none. The keyword none means that a
revocation check will not be performed and the certificate
will always be accepted.
• Using the none keyword is mandatory for this task.
Step 5 exit Exits ca-trustpoint configuration mode and returns to global
configuration mode.
Example:
Router(ca-trustpoint)# exit

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Step 6 crypto pki authenticate name Authenticates the CA (by getting the certificate from the
CA).
Example: • Takes the name of the CA as the argument.
Router(config)# crypto pki authenticate srstca
Step 7 crypto pki enroll name Obtains the SRST router certificate from the CA.
• Takes the name of the CA as the argument.
Example:
Router(config)# crypto pki enroll srstca
Examples
The following example autoenrolls and authenticates the Cisco Unified SRST router:
Router(config)# crypto pki trustpoint srstca
Router(ca-trustpoint)# enrollment url http://10.1.1.22
Router(config)# crypto pki authenticate srstca
Certificate has the following attributes:

Fingerprint MD5: 4C894B7D 71DBA53F 50C65FD7 75DDBFCA
Fingerprint SHA1: 5C3B6B9E EFA40927 9DF6A826 58DA618A BF39F291
% Do you accept this certificate? [yes/no]: y
Trustpoint CA certificate accepted.
Router(config)# crypto pki enroll srstca

%
% Start certificate enrollment ..
% Create a challenge password. You will need to verbally provide this
password to the CA Administrator in order to revoke your certificate.
For security reasons your password will not be saved in the configuration.
Please make a note of it.
Password:
Re-enter password:
% The fully-qualified domain name in the certificate will be: router.cisco.com

% The subject name in the certificate will be: router.cisco.com
% Include the router serial number in the subject name? [yes/no]: y
% The serial number in the certificate will be: D0B9E79C
% Include an IP address in the subject name? [no]: n
Request certificate from CA? [yes/no]: y
% Certificate request sent to Certificate Authority
% The certificate request fingerprint will be displayed.
% The 'show crypto pki certificate' command will also show the fingerprint.
Sep 29 00:41:55.427: CRYPTO_PKI: Certificate Request Fingerprint MD5: D154FB75

2524A24D 3D1F5C2B 46A7B9E4
Sep 29 00:41:55.427: CRYPTO_PKI: Certificate Request Fingerprint SHA1: 0573FBB2
98CD1AD0 F37D591A C595252D A17523C1
Sep 29 00:41:57.339: %PKI-6-CERTRET: Certificate received from Certificate Authority

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Disabling Automatic Certificate Enrollment

The command grant auto allows certificates to be issued and was activated in the optional task
documented in the “Configuring a Certificate Authority Server on a Cisco IOS Certificate Server”
Note You should disable the grant auto command so that certificates cannot be continually granted.
SUMMARY STEPS
1. crypto pki server cs-label

2. shutdown
3. no grant auto
4. no shutdown
DETAILED STEPS

Step 1 crypto pki server cs-label Enables the certificate server and enters certificate server
configuration mode.
Example: Note If you manually generated an RSA key pair, the
Router (config)# crypto pki server srstcaserver cs-label argument must match the name of the key
pair.
Step 2 shutdown Disables the Cisco IOS certificate server.
Example:
Router (cs-server)# shutdown
Step 3 no grant auto Disables automatic certificates to be issued to any
requestor.
Example: • This command was for use during enrollment only and
Router (cs-server)# no grant auto thus needs to be removed in this task.
Step 4 no shutdown Enables the Cisco IOS certificate server.
• You should issue this command only after you have
Example: completely configured your certificate server.
Router (cs-server)# no shutdown
What to Do Next
For manual enrollment instructions, see the Manual Certificate Enrollment (TFTP and Cut-and-Paste)
feature.

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Verifying Certificate Enrollment

If you used the Cisco IOS certificate server as your CA, use the show running-config command to verify
certificate enrollment or the show crypto pki server command to verify the status of the CA server.
SUMMARY STEPS
1. show running-config
2. show crypto pki server

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DETAILED STEPS

Step 1 show running-config Use the show running-config command to verify
the creation of the CA server (01) and device (02)
certificates. This example shows the enrolled
Example:
certificates.
.
.
.
! SRST router device certificate.
crypto pki certificate chain srstca
certificate 02
308201AD 30820116 A0030201 02020102 300D0609
2A864886 F70D0101 04050030
17311530 13060355 0403130C 73727374 63617365
72766572 301E170D 30343034
31323139 35323233 5A170D30 35303431 32313935
3232335A 30343132 300F0603
55040513 08443042 39453739 43301F06 092A8648
86F70D01 09021612 6A61736F
32363931 2E636973 636F2E63 6F6D305C 300D0609
2A864886 F70D0101 01050003
4B003048 024100D7 0CC354FB 5F7C1AE7 7A25C3F2
056E0485 22896D36 6CA70C19
C98F9BAE AE9D1F9B D4BB7A67 F3251174 193BB1A3
12946123 E5C1CCD7 A23E6155
FA2ED743 3FB8B902 03010001 A330302E 300B0603
551D0F04 04030205 A0301F06
03551D23 04183016 8014F829 CE97AD60 18D05467
FC293963 C2470691 F9BD300D
06092A86 4886F70D 01010405 00038181 007EB48E
CAE9E1B3 D1E7A185 D7F0D565
CB84B17B 1151BD78 B3E39763 59EC650E 49371F6D
99CBD267 EB8ADF9D 9E43A5F2
FB2B18A0 34AF6564 11239473 41478AFC A86E6DA1
AC518E0B 8657CEBB ED2BDE8E
B586FE67 00C358D4 EFDD8D44 3F423141 C2D331D3
1EE43B6E 6CB29EE7 0B8C2752
C3AF4A66 BD007348 D013000A EA3C206D CF
quit
certificate ca 01
30820207 30820170 A0030201 02020101 300D0609
2A864886 F70D0101 04050030
17311530 13060355 0403130C 73727374 63617365
72766572 301E170D 30343034
31323139 34353136 5A170D30 37303431 32313934
3531365A 30173115 30130603
55040313 0C737273 74636173 65727665 7230819F
300D0609 2A864886 F70D0101
01050003 818D0030 81890281 8100C3AF EE1E4BB1
9922A8DA 2BB9DC8E 5B1BD332
1051C9FE 32A971B3 3C336635 74691954 98E765B1
059E24B6 32154E99 105CA989
9619993F CC72C525 7357EBAC E6335A32 2AAF9391
99325BFD 9B8355EB C10F8963

9-254

9D8FC222 EE8AC831 71ACD3A7 4E918A8F D5775159
76FBF499 5AD0849D CAA41417
DD866902 21E5DD03 C37D4B28 0FAB0203 010001A3
63306130 0F060355 1D130101
FF040530 030101FF 300E0603 551D0F01 01FF0404
03020186 301D0603 551D0E04
160414F8 29CE97AD 6018D054 67FC2939 63C24706
91F9BD30 1F060355 1D230418
30168014 F829CE97 AD6018D0 5467FC29 3963C247
0691F9BD 300D0609 2A864886
F70D0101 04050003 8181007A F71B25F9 73D74552
25DFD03A D8D1338F 6792C805
47A81019 795B5AAE 035400BB F859DABF 21892B5B
E71A8283 08950414 8633A8B2
C98565A6 C09CA641 88661402 ACC424FD 36F23360
ABFF4C55 BB23C66A C80A3A57
5EE85FF8 C1B1A540 E818CE6D 58131726 BB060974
4E1A2F4B E6195522 122457F3
DEDBAAD7 3780136E B112A6
quit
Step 2 show crypto pki server Use the show crypto pki server command to verify
the status of the CA server after a boot procedure.
Example:
Router# show crypto pki server
Certificate Server srstcaserver:
Status: enabled
Server's configuration is locked (enter "shut" to
unlock it)
Issuer name: CN=srstcaserver
CA cert fingerprint: AC9919F5 CAFE0560 92B3478A
CFF5EC00
Granting mode is: auto
Last certificate issued serial number: 0x2
CA certificate expiration timer: 13:46:57 PST Dec 1
2007
CRL NextUpdate timer: 14:54:57 PST Jan 19 2005
Current storage dir: nvram
Database Level: Complete - all issued certs written
as <serialnum>.cer
Enabling Credentials Service on the Secure Cisco Unified SRST Router

Once the Cisco Unified SRST Router has its own certificate, you need to provide Cisco
Unified Communications Manager the certificate. Enabling credentials service allows Cisco
Unified Communications Manager to retrieve the secure SRST device certificate and place it in the
configuration file of the Cisco Unified IP Phone.
Activate credentials service on all Cisco Unified SRST Routers.
Note A security best practice is to protect the credentials service port using Control Plane Policing. Control
Plane Policing protects the gateway and maintains packet forwarding and protocol states despite a heavy
traffic load. For more information on control planes, see the Control Plane Policing documentation. In
addition, a sample configuration is given in the “Control Plane Policing: Example” section on page 283.

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SUMMARY STEPS
1. credentials
2. ip source-address ip-address [port port]
3. trustpoint trustpoint-name
4. exit
DETAILED STEPS

Step 1 credentials Provides the Cisco Unified SRST Router certificate to
Cisco Unified Communications Manager and enters
credentials configuration mode.
Example:
Router(config)# credentials
Step 2 ip source-address ip-address [port port] Enables the Cisco Unified SRST Router to receive
messages from Cisco Unified Communications Manager
through the specified IP address and port.
Example:
Router(config-credentials)# ip source-address • ip-address: The IP address is the pre-existing router IP
10.1.1.22 port 2445 address, typically one of the addresses of the Ethernet
port of the router.
• port port: (Optional) The port to which the gateway
router connects to receive messages from
Cisco Unified Communications Manager. The port
number is from 2000 to 9999. The default port number
is 2445.
Step 3 trustpoint trustpoint-name Specifies the name of the trustpoint that is to be associated
with the Cisco Unified SRST Router certificate. The
trustpoint-name argument is the name of the trustpoint and
Example:
Router(config-credentials)# trustpoint srstca
corresponds to the SRST device certificate.
• The trustpoint name should be the same as the one
declared in the “Autoenrolling and Authenticating the
Secure Cisco Unified SRST Router to the CA Server”
Step 4 exit Exits credentials configuration mode.
Example:
Router(config-credentials)# exit
Examples
Router(config)# credentials
Router(config-credentials)# ip source-address 10.1.1.22 port 2445
Router(config-credentials)# trustpoint srstca
Router(config-credentials)# exit

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Troubleshooting Credential Settings

The following steps display credential settings or set debugging on the credential settings of the
Cisco Unified SRST Router.
SUMMARY STEPS
1. show credentials
2. debug credentials
DETAILED STEPS

Step 1 show credentials Use the show credentials command to display the
credential settings on the Cisco Unified SRST
Router that are supplied to Cisco
Example:
Router# show credentials
Unified Communications Manager for use during
secure Cisco Unified SRST fallback.
Credentials IP: 10.1.1.22
Credentials PORT: 2445
Trustpoint: srstca
Step 2 debug credentials Use the debug credentials command to set
debugging on the credential settings of the Cisco
Unified SRST Router.
Example:
Router# debug credentials
Credentials server debugging is enabled

Router#
Sep 29 01:01:50.903: Credentials service: Start TLS
Handshake 1 10.1.1.13 2187
Sep 29 01:01:50.903: Credentials service: TLS
Handshake returns OPSSLReadWouldBlockErr
Handshake completes.
Related Commands
Use the following commands to show if a certificate cannot be found (you are missing a certificate that
you are trying to authenticate) or to show that a particular certificate has matched (so you know what
certificate the router used to authenticate a phone):
• debug crypto pki messages
• debug crypto pki transactions
Importing Phone Certificate Files in PEM Format to the Secure SRST Router
This task completes the tasks required for Cisco IP Unified Phones to authenticate secure SRST.

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Cisco Unified Communications Manager 4.X.X and Earlier Versions

For systems running Cisco Unified Communications Manager 4.X.X and earlier versions, the secure
Cisco Unified SRST Router must retrieve phone certificates so that it can authenticate Cisco Unified IP
phones during the TLS handshake. Different certificates are used for different Cisco Unified IP Phones.
Table 9-1 lists the certificates needed for each type of phone.
Certificates must be imported manually from Cisco Unified Communications Manager to the
Cisco Unified SRST Router. The number of certificates depends on the Cisco Unified Communications
Manager configuration. Manual enrollment refers to cut and paste or TFTP. For manual enrollment
instructions, see the Manual Certificate Enrollment (TFTP and Cut-and-Paste) feature. Repeat the
enrollment procedure for each phone or PEM file.
For Cisco Unified Communications Manager 4.X.X and earlier versions, certificates are found by going
to the menu bar in Cisco Unified Communications Manager, choose Program Files > Cisco >
Certificates.
Open the .0 files with Windows WordPad or Notepad, and copy and paste the contents to the SRST router
console. Then, repeat the procedure with the .pem file. Copy all the contents that appear between
“-----BEGIN CERTIFICATE-----” and “-----END CERTIFICATE-----”.
For certification operation on Cisco Unified Communications Operating System Administration Guide,
Release 6.1(1), see
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/cucos/6_1_1/cucos/iptpch6.html.
Cisco Unified Communications Manager 5.0 and Later Versions

Systems running Cisco Unified CM 5.0 and later versions require four certificates (CAPF, CiscoCA,
CiscoManufactureCA, and CiscoRootCA2048) in addition to the requirements listed in Table 9-1, which
must be copied and pasted to Cisco Unified SRST Routers.
Note CiscoRootCA is also called CiscoRoot2048CA.
Prerequisites
You must have certificates available when the last configuration command (crypto pki authenticate)
issues the following prompt:
Enter the base 64 encoded CA certificate.
End with a blank line or the word "quit" on a line by itself
For Cisco Unified CM 5.0 and later versions, perform the following steps:
Step 1 Login to Cisco Unified Communications Manager.

Step 2 Go to Security > Certificate Management > Download Certificate/CTL.
Step 3 Select Download Trust Cert and click Next.
Step 4 Select CAPF-trust and click Next.
Step 5 Select CiscoCA and click Next.
Step 6 Click Continue.
Step 7 Click the file name.

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Step 8 Copy all the contents that appear between “-----BEGIN CERTIFICATE-----” and “-----END
CERTIFICATE-----” to a location where you can retrieve it later.
Step 9 Repeat Steps 5 to 8 for CiscoManufactureCA, CiscoRootCA2048, and CAPF.
Cisco Unified Communications Manager 6.0 and Later Versions

From Cisco Unified Communications Operating System Administration, download all certificates listed
under CAPF-trust, including Cisco_Manufacturing_CA, Cisco_Root_CA_2048, CAP-RTP-001,
CAP-RTP-002, CAPF, and CAPF-xxx. Also download any CAPF-xxx certificates that are listed under
CallManager-trust and not under CAPF-trust.
For instructions on downloading certificates, see the “Security” chapter in the appropriate version of
Cisco Unified Communications Operating System Administration Guide.
Authenticating the Imported Certificates on the Cisco Unified SRST Router

To authenticate certificates on the Cisco Unified SRST router, perform these steps.
Restrictions
HTTP automatic enrollment from Cisco Unified Communications Manager through a virtual web server
is not supported.
SUMMARY STEPS
1. crypto pki trustpoint name

2. revocation-check none
3. enrollment terminal
4. exit
5. crypto pki authenticate name

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DETAILED STEPS

Step 1 crypto pki trustpoint name Declares the CA that your router should use and enters
ca-trustpoint configuration mode.
Example: • name: Enter the name of each certificate individually
Router (config)# crypto pki trustpoint CAPF (for example, CAPF, CiscoCA, CiscoManufactureCA,
and CiscoRootCA2048).
Step 2 revocation-check none Checks the revocation status of a certificate using the
selected method.
Example: • Using the none keyword is mandatory for this task.
Router(ca-trustpoint)# revocation-check none The keyword none means that a revocation check is
not performed and the certificate is always accepted.
Step 3 enrollment terminal Specifies manual cut-and-paste certificate enrollment.
Example:
Router(ca-trustpoint)# enrollment terminal
Step 4 exit Exits ca-trustpoint configuration mode and returns to
global configuration.
Example:
Step 5 crypto pki authenticate name Authenticates the CA (by getting the certificate from the
CA).
Example: • Enter the same name argument used in the crypto pki
Router(config)# crypto pki authenticate CAPF trustpoint command in Step 1.
What to Do Next
Update the certificates in Cisco Unified CM. See the “Configuring a Secure Survivable Remote Site
Telephony (SRST) Reference” chapter in the appropriate version of Cisco Unified Communications
Manager Security Guide.
Examples
This section provides the following:
• Cisco Unified Communications Manager 4.X.X and Earlier Versions: Example, page 261
• Cisco Unified Communications Manager 5.0 and Later Versions Example, page 263

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Cisco Unified Communications Manager 4.X.X and Earlier Versions: Example
The following example shows three certificates (Cisco 7970, 7960, PEM) imported to the Cisco Unified
SRST Router:
Router(config)# crypto pki trustpoint 7970
Router(config)# crypto pki authenticate 7970
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quit
Fingerprint MD5: F7E150EA 5E6E3AC5 615FC696 66415C9F
Fingerprint SHA1: 1BE2B503 DC72EE28 0C0F6B18 798236D8 D3B18BE6
% Certificate successfully imported
Router(config)# crypto pki trustpoint 7960

Router(config)# crypto pki authenticate 7960
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quit
Fingerprint MD5: 4B9636DF 0F3BA6B7 5F54BE72 24762DBC

9-261
Fingerprint SHA1: A9917775 F86BB37A 5C130ED2 3E528BB8 286E8C2D

Router(config)# crypto pki trustpoint PEM

Router(config)# crypto pki authenticate PEM

MIIDqDCCApCgAwIBAgIQdhL5YBU9b59OQiAgMrcjVjANBgkqhkiG9w0BAQUFADAu
MRYwFAYDVQQKEw1DaXNjbyBTeXN0ZW1zMRQwEgYDVQQDEwtDQVAtUlRQLTAwMTAe
Fw0wMzAyMDYyMzI3MTNaFw0yMzAyMDYyMzM2MzRaMC4xFjAUBgNVBAoTDUNpc2Nv
IFN5c3RlbXMxFDASBgNVBAMTC0NBUC1SVFAtMDAxMIIBIDANBgkqhkiG9w0BAQEF
AAOCAQ0AMIIBCAKCAQEArFW77Rjem4cJ/7yPLVCauDohwZZ/3qf0sJaWlLeAzBlq
Rj2lFlSij0ddkDtfEEo9VKmBOJsvx6xJlWJiuBwUMDhTRbsuJz+npkaGBXPOXJmN
Vd54qlpc/hQDfWlbrIFkCcYhHws7vwnPsLuy1Kw2L2cP0UXxYghSsx8H4vGqdPFQ
NnYy7aKJ43SvDFt4zn37n8jrvlRuz0x3mdbcBEdHbA825Yo7a8sk12tshMJ/YdMm
vny0pmDNZXmeHjqEgVO3UFUn6GVCO+K1y1dUU1qpYJNYtqLkqj7wgccGjsHdHr3a
U+bw1uLgSGsQnxMWeMaWo8+6hMxwlANPweufgZMaywIBA6OBwzCBwDALBgNVHQ8E
BAMCAYYwDwYDVR0TAQH/BAUwAwEB/zAdBgNVHQ4EFgQU6Rexgscfz6ypG270qSac
cK4FoJowbwYDVR0fBGgwZjBkoGKgYIYtaHR0cDovL2NhcC1ydHAtMDAxL0NlcnRF
bnJvbGwvQ0FQLVJUUC0wMDEuY3Jshi9maWxlOi8vXFxjYXAtcnRwLTAwMVxDZXJ0
RW5yb2xsXENBUC1SVFAtMDAxLmNybDAQBgkrBgEEAYI3FQEEAwIBADANBgkqhkiG
9w0BAQUFAAOCAQEAq2T96/YMMtw2Dw4QX+F1+g1XSrUCrNyjx7vtFaRDHyB+kobw
dwkpohfkzfTyYpJELzV1r+kMRoyuZ7oIqqccEroMDnnmeApc+BRGbDJqS1Zzk4OA
c6Ea7fm53nQRlcSPmUVLjDBzKYDNbnEjizptaIC5fgB/S9S6C1q0YpTZFn5tjUjy
WXzeYSXPrcxb0UH7IQJ1ogpONAAUKLoPaZU7tVDSH3hD4+VjmLyysaLUhksGFrrN
phzZrsVVilK17qpqCPllKLGAS4fSbkruq3r/6S/SpXS6/gAoljBKixP7ZW2PxgCU
1aU9cURLPO95NDOFN3jBk3Sips7cVidcogowPQ==
quit
Fingerprint MD5: 233C8E33 8632EA4E 76D79FEB FFB061C6
Fingerprint SHA1: F7B40B94 5831D2AB 447AB8F2 25990732 227631BE
Use the show crypto pki trustpoint status command to show that enrollment has succeeded
and that five CA certificates were granted. The five certificates include the three
certificates just entered and the CA server certificate and the SRST router certificate.
Router# show crypto pki trustpoint status
Trustpoint 7970:
Issuing CA certificate configured:
Subject Name:
cn=CAP-RTP-002,o=Cisco Systems
Fingerprint MD5: F7E150EA 5E6E3AC5 615FC696 66415C9F
Fingerprint SHA1: 1BE2B503 DC72EE28 0C0F6B18 798236D8 D3B18BE6
State:
Keys generated ............. Yes (General Purpose)
Issuing CA authenticated ....... Yes
Certificate request(s) ..... None
Trustpoint 7960:
Subject Name:
cn=CAPF-508A3754,o=Cisco Systems Inc,c=US
Fingerprint MD5: 6BAE18C2 0BCE391E DAE2FE4C 5810F576
Fingerprint SHA1: B7735A2E 3A5C274F C311D7F1 3BE89942 355102DE
State:

9-262

Trustpoint PEM:
Subject Name:
cn=CAP-RTP-001,o=Cisco Systems
Fingerprint MD5: 233C8E33 8632EA4E 76D79FEB FFB061C6
Fingerprint SHA1: F7B40B94 5831D2AB 447AB8F2 25990732 227631BE
State:
Trustpoint srstcaserver:
Subject Name:
cn=srstcaserver
Fingerprint MD5: 6AF5B084 79C93F2B 76CC8FE6 8781AF5E
Fingerprint SHA1: 47D30503 38FF1524 711448B4 9763FAF6 3A8E7DCF
State:
Trustpoint srstca:
Subject Name:
cn=srstcaserver
Fingerprint MD5: 6AF5B084 79C93F2B 76CC8FE6 8781AF5E
Fingerprint SHA1: 47D30503 38FF1524 711448B4 9763FAF6 3A8E7DCF
Router General Purpose certificate configured:
Subject Name:
serialNumber=F3246544+hostname=c2611XM-sSRST.cisco.com
Fingerprint: 35471295 1C907EC1 45B347BC 7A9C4B86
State:
Certificate request(s) ..... Yes
Cisco Unified Communications Manager 5.0 and Later Versions Example
The following example shows the configuration for the four certificates (CAPF, CiscoCA,
CiscoManufactureCA, and CiscoRootCA2048) that are required for systems running
Cisco Unified Communications Manager 5.0:
Router(config)# crypto pki trustpoint CAPF
Router(config)# crypto pki authenticate CAPF

MIICKjCCAZOgAwIBAgIC8wEwDQYJKoZIhvcNAQEFBQAwQTELMAkGA1UEBhMCVVMx
GjAYBgNVBAoTEUNpc2NvIFN5c3RlbXMgSW5jMRYwFAYDVQQDEw1DQVBGLTU4RUFE
MkQyMB4XDTA2MDMwMTIxMjc1MloXDTIxMDIyNTIxMjc1MVowQTELMAkGA1UEBhMC
VVMxGjAYBgNVBAoTEUNpc2NvIFN5c3RlbXMgSW5jMRYwFAYDVQQDEw1DQVBGLTU4
RUFEMkQyMIGfMA0GCSqGSIb3DQEBAQUAA4GNADCBiQKBgQC99KgZT94qhozw4bOB

9-263
f8Z0tYwT2l4L++mC64O3s3AshDi8xe8Y8sN/f/ZKRRhNIxBlK4SWafXnHKJBqKZn
WtSgkRjJ3Dh0XtqcWYt8VS2sC69g8sX09lskKl3m+TpWsr2T/mDXv6CceaKN+mch
gcrrnNo8kamOOIG8OsQc4L6XzQIDAQABozEwLzAOBgNVHQ8BAf8EBAMCAoQwHQYD
quit
Fingerprint MD5: 1951DJ4E 76D79FEB FFB061C6 233C8E33
Fingerprint SHA1: 222891BE Z7B89B94 447AB8F2 5831D2AB 25990732
Router(config)# crypto pki trustpoint CiscoCA

Router(config)# crypto pki authenticate CiscoCA
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quit
Fingerprint MD5: 21956CBR 4B9706DF 0F3BA6B7 7P54AZ72
Fingerprint SHA1: A9917775 F86BB37A 7H130ED2 3E528BB8 286E8C2D
Router(config)# crypto pki trustpoint CiscoManufactureCA

Router(config)# crypto pki authenticate CiscoManufactureCA
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9-264
S25l3JNFBemvM2tnIwcGhiLa69yHz1khQhrpz3B1iOAkPV19TpY4gJfVb/Cbcdi6
YBmlsGGGrd1lZva5J6LuL2GbuqEwYf2+rDUU+bgtlwavw+9tzD0865XpgdOKXrbO
+nmka9eiV2TEP0zJ2+iC7AFm1BCIolblPFft6QKoSJFjB6thJksaE5/k3Npf
quit
Fingerprint MD5: 0F3BA6B7 4B9636DF 5F54BE72 24762SBR
Fingerprint SHA1: L92BB37A S9919925 5C130ED2 3E528UP8 286E8C2D
Router(config)# crypto pki trustpoint CiscoRootCA2048

Router(config)# crypto pki authenticate CiscoRootCA2048
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quit
Fingerprint MD5: 2G3LZ6B7 2R1995ER 6KE4WE72 3E528BB8
Fingerprint SHA1: M9912245 5C130ED2 24762JBC 3E528VF8 956E8S5H
Configuring Cisco Unified Communications Manager to the Secure Cisco

Unified SRST Router
The following tasks are performed in Cisco Unified Communications Manager:
• Adding an SRST Reference to Cisco Unified Communications Manager, page 265 (required)
• Configuring SRST Fallback on Cisco Unified Communications Manager, page 266 (required)
• Configuring CAPF on Cisco Unified Communications Manager, page 268 (required)
Adding an SRST Reference to Cisco Unified Communications Manager

The following procedure describes how to add an SRST reference to Cisco Unified Communications
Manager.
Before following this procedure, verify that credentials service is running in the Cisco Unified SRST
Router. Cisco Unified Communications Manager connects to the Cisco Unified SRST Router for its
device certificate. To enable credentials service, see the “Enabling Credentials Service on the Secure
Cisco Unified SRST Router” section on page 255.

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For complete information on adding Cisco Unified SRST to Cisco Unified Communications Manager,
see the “Survivable Remote Site Telephony Configuration” section for the Cisco Unified
Communications Manager version that you are running. All Cisco Unified CM administration guides are
at the following URL:
http://www.cisco.com/en/US/products/sw/voicesw/ps556/prod_maintenance_guides_list.html.
Step 1 In the menu bar in Cisco Unified Communications Manager, choose CCMAdmin > System > SRST.
Step 2 Click Add New SRST Reference.
Step 3 Enter the appropriate settings. Figure 9-3 shows the available fields in the SRST Reference
Configuration window.
a. Enter the name of the SRST gateway, the IP address, and the port.
b. Check the box asking if the SRST gateway is secure.
c. Enter the certificate provider (credentials service) port number. Credentials service runs on default
port 2445.
Figure 9-3 SRST Reference Configuration Window
Step 4 To add the new SRST reference, click Insert. The message “Status: Insert completed” displays.
Step 5 To add more SRST references, repeat Steps 2 to 4.
Configuring SRST Fallback on Cisco Unified Communications Manager

The following procedure describes how to configure SRST fallback on Cisco Unified Communications
Manager by assigning the Unified SRST reference to a device pool.

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For complete information about adding a device pool to Cisco Unified Communications Manager, see
the “Device Pool Configuration” section in Cisco Unified Communications Manager Administration
Guide for the Cisco Unified Communications Manager version that you are running. All Cisco Unified
CM administration guides are at the following URL:
http://www.cisco.com/en/US/products/sw/voicesw/ps556/prod_maintenance_guides_list.html
Step 1 In the menu bar in Cisco Unified Communications Manager, choose CCMAdmin > System > Device
Pool.
Step 2 Use one of the following methods to add a device pool:
• If a device pool already exists with settings that are similar to the one that you want to add, choose
the existing device pool to display its settings, click Copy, and modify the settings as needed.
Continue with Step 4.
• To add a device pool without copying an existing one, continue with Step 3.
Step 3 In the upper, right corner of the window, click the Add New Device Pool link. The Device Pool
Configuration window displays (see Figure 9-4).
Figure 9-4 Device Pool Configuration Window
Step 4 Enter the SRST reference.

Step 5 Click Update to save the device pool information in the database.

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Configuring CAPF on Cisco Unified Communications Manager

The Certificate Authority Proxy Function (CAPF) process allows supported devices, such as Cisco
Unified IP Phones to request LSC certificates from the CAPF service on Cisco Unified Communications
Manager. The CAPF utility generates a key pair and certificate that are specific for CAPF, and the utility
copies this certificate to all Cisco Unified Communications Manager servers in the cluster.
For complete instructions on configuring CAPF in Cisco Unified Communications Manager, see the
Cisco IP Phone Authentication and Encryption for Cisco Communications Manager documentation.
Enabling SRST Mode on the Secure Cisco Unified SRST Router

To configure secure SRST on the router to support the Cisco Unified IP Phone functions, use the
following commands beginning in global configuration mode.
SUMMARY STEPS
2. secondary-dialtone digit-string
3. transfer-system {blind | full-blind | full-consult | local-consult}
8. exit

9-268
DETAILED STEPS

Example:
Step 2 secondary-dialtone digit-string Activates a secondary dial tone when a digit string is
dialed.
Example:
Router(config-cm-fallback)# secondary-dialtone 9
Step 3 transfer-system {blind | full-blind | Defines the call-transfer method for all lines served by
full-consult | local-consult} the Cisco Unified SRST Router.
• blind: Calls are transferred without consultation
Example: with a single phone line using the Cisco proprietary
Router(config-cm-fallback)# transfer-system method.
full-consult
• full-blind: Calls are transferred without
consultation using H.450.2 standard methods.
• full-consult: Calls are transferred with consultation
using a second phone line if available. The calls
fallback to full-blind if the second line is
unavailable.
• local-consult: Calls are transferred with local
consultation using a second phone line if available.
The calls fallback to blind for nonlocal consultation
or nonlocal transfer target.
Phones through the specified IP addresses and provides
for strict IP address verification. The default port number
Example:
is 2000.
10.1.1.22 port 2000
Step 5 max-ephones max-phones Configures the maximum number of Cisco IP phones
that can be supported by the router. The maximum
number is platform dependent. The default is 0. See the
Example:
“Platform and Memory Support” section on page 26 for
further details.
Step 6 max-dn max-directory-numbers Sets the maximum number of directory numbers (DNs)
or virtual voice ports that can be supported by the router.
Router(config-cm-fallback)# max-dn 30 directory numbers or virtual voice ports supported
by the router. The maximum number is platform
dependent. The default is 0. See the “Platform and
Memory Support” section on page 26 for further
details.

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Step 7 transfer-pattern transfer-pattern Allows transfer of phone calls by Cisco Unified IP
Phones to specified phone number patterns.
Example: • transfer-pattern: String of digits for permitted call
Router(config-cm-fallback)# transfer-pattern transfers. Wildcards are allowed.
.....
Example:
Examples
The following example enables SRST mode on your router:
Router(config-cm-fallback)# secondary-dialtone 9
Router(config-cm-fallback)# transfer-system full-consult
Router(config-cm-fallback)# ip source-address 10.1.1.22 port 2000
Router(config-cm-fallback)# max-dn 30
Router(config-cm-fallback)# transfer-pattern .....
Configuring Secure SCCP SRST

• Prerequisites for Configuring Secure SCCP SRST, page 270
• Restrictions for Configuring Secure SCCP SRST, page 270
• Verifying Phone Status and Registrations, page 271 (required)
• Configuration Examples for Secure SCCP SRST, page 278
Prerequisites for Configuring Secure SCCP SRST

• Cisco Unified Communications Manager 4.1(2) or later must be installed and must support security
mode (authenticate and encryption mode).
• Unified SRST 12.3 or later releases for Secure SCCP support on Cisco 4000 Series Integrated
Services Routers and Cisco Analog Voice Gateways mentioned in the section Secure SCCP SRST
for Analog Voice Gateways, page 235. The configuration and behavior of Secure SCCP SRST
fallback aligns with the existing support offered on Cisco Integrated Services Router Generation 2,
unless specified otherwise.
Restrictions for Configuring Secure SCCP SRST

Not Supported in Secure SCCP SRST Mode (For Unified SRST 12.2 and prior releases)
• Cisco Unified Communications Manager versions before 4.1(2).
• Secure MOH; MOH stays active, but reverts to non-secure.
• Secure transcoding or conferencing.

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• Secure H.323 or SIP trunks.

• SIP phones interoperability.
• Hot Standby Routing Protocol (HSRP).
Not Supported in Secure SCCP SRST Mode (For Unified SRST 12.3 and later releases)
For information on the restrictions for Secure SCCP SRST support introduced on Unified SRST 12.3,
see the section SCCP SRST in Restrictions for Configuring Secure SRST, page 232.
Supported Calls in Secure SCCP SRST Mode (For Unified SRST 12.2 and prior releases)
Only voice calls are supported in secure SCCP SRST mode. Specifically, the following voice calls are
supported:
• Basic call
• Call transfer (consult and blind)
• Call forward (busy, no-answer, all)
• Shared line (IP phones)
• Hold and resume
For information on the features supported on Unified SRST 12.3 and later releases, see Feature Support
for Secure SRST (SCCP), Unified SRST Release 12.3, page 238.
Verifying Phone Status and Registrations

To verify or troubleshoot Cisco Unified IP Phone status and registration, complete the following steps
beginning in privileged EXEC mode.
Note You can verify Phone Status and Registrations in secure SCCP SRST after you have performed the
following steps:
• Enabling Credentials Service on the Secure Cisco Unified SRST Router, page 255
• Adding an SRST Reference to Cisco Unified Communications Manager, page 265
• Enabling SRST Mode on the Secure Cisco Unified SRST Router, page 268
SUMMARY STEPS
1. show ephone
2. show ephone offhook
3. show voice call status
4. debug ephone register
5. debug ephone state

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DETAILED STEPS

Step 1 show ephone Use this command to display registered Cisco
Unified IP Phones and their capabilities. The show
ephone command also displays authentication and
Example:
Router# show ephone
encryption status when used for secure SCCP SRST.
In this example, authentication and encryption status
ephone-1 Mac:1000.1111.0002 TCP socket:[5] is active with a TLS connection.
activeLine:0 REGISTERED in SCCP ver 5
+ Authentication + Encryption with TLS connection
mediaActive:0 offhook:0 ringing:0 reset:0
reset_sent:0 paging 0 debug:0
IP:10.1.1.40 32626 7970 keepalive 390 max_line 8
button 1: dn 14 number 2002 CM Fallback CH1 IDLE
ephone-2 Mac:1000.1111.000B TCP socket:[12]

activeLine:0 REGISTERED in SCCP ver
5 + Authentication + Encryption with TLS connection
ephone-3 Mac:1000.1111.000A TCP socket:[16]

Step 2 show ephone offhook Use this command to display Cisco IP Phone status
and quality for all phones that are off hook. In this
example, authentication and encryption status is
Example:
Router# show ephone offhook
active with a TLS connection, and there is an active
secure call.
ephone-1 Mac:1000.1111.0002 TCP socket:[5]
activeLine:1 REGISTERED in SCCP ver 5
+ Authentication + Encryption with TLS connection
reset_sent:0 paging 0
:0
button 1: dn 14 number 2002 CM Fallback CH1
CONNECTED
Active Secure Call on DN 14 chan 1 :2002 10.1.1.40
29632 to 10.1.1.40 25616 via 10.1.1.40
G711Ulaw64k 160 bytes no vad
Tx Pkts 295 bytes 49468 Rx Pkts 277 bytes 46531 Lost
0
Jitter 0 Latency 0 callingDn 22 calledDn -1
ephone-2 Mac:1000.1111.000B TCP socket:[12]

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button 1: dn 21 number 2011 CM Fallback CH1
CONNECTED
Active Secure Call on DN 21 chan 1 :2011 10.1.1.40
16382 to 10.1.1.40 16382 via 10.1.1.40
G711Ulaw64k 160 bytes no vad
Tx Pkts 295 bytes 49468 Rx Pkts 277 bytes 46531 Lost
0
Jitter 0 Latency 0 callingDn -1 calledDn 11

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Step 3 show voice call status Use this command to show the call status for all
voice ports on the Cisco Unified SRST router. This
command is not applicable for calls between two
Example:
CallID CID ccVdb Port DSP/Ch Called # Codec
POTS dial peers.
Dial-peers
0x1164 2BFE 0x8619A460 50/0/35.0 2014 g711ulaw
20035/20027
0x1165 2BFE 0x86144B78 50/0/27.0 *2014 g711ulaw
20027/20035
0x1166 2C01 0x861043D8 50/0/21.0 2012 g711ulaw
20021/20011
0x1168 2C01 0x860984C4 50/0/11.0 *2012 g711ulaw
20011/20021
0x1167 2C04 0x8610EC7C 50/0/22.0 2002 g711ulaw
20022/20014
0x1169 2C04 0x860B8894 50/0/14.0 *2002 g711ulaw
20014/20022
0x116A 2C07 0x860A374C 50/0/12.0 2010 g711ulaw
20012/20002
0x116B 2C07 0x86039700 50/0/2.0 *2010 g711ulaw
20002/20012
0x116C 2C0A 0x86119520 50/0/23.0 2034 g711ulaw
20023/20020
0x116D 2C0A 0x860F9150 50/0/20.0 *2034 g711ulaw
20020/20023
0x116E 2C0D 0x8608DC20 50/0/10.0 2022 g711ulaw
20010/20008
0x116F 2C0D 0x86078AD8 50/0/8.0 *2022 g711ulaw
20008/20010
0x1170 2C10 0x861398F0 50/0/26.0 2016 g711ulaw
20026/20028
0x1171 2C10 0x8614F41C 50/0/28.0 *2016 g711ulaw
20028/20026
0x1172 2C13 0x86159CC0 50/0/29.0 2018 g711ulaw
20029/20004
0x1173 2C13 0x8604E848 50/0/4.0 *2018 g711ulaw
20004/20029
0x1174 2C16 0x8612F04C 50/0/25.0 2026 g711ulaw
20025/20030
0x1175 2C16 0x86164F48 50/0/30.0 *2026 g711ulaw
20030/20025
0x1176 2C19 0x860D8C64 50/0/17.0 2032 g711ulaw
20017/20018
0x1177 2C19 0x860E4008 50/0/18.0 *2032 g711ulaw
20018/20017
0x1178 2C1C 0x860CE3C0 50/0/16.0 2004 g711ulaw
20016/20019
0x1179 2C1C 0x860EE8AC 50/0/19.0 *2004 g711ulaw
20019/20016
0x117A 2C1F 0x86043FA4 50/0/3.0 2008 g711ulaw
20003/20024
0x117B 2C1F 0x861247A8 50/0/24.0 *2008 g711ulaw
20024/20003
0x117C 2C22 0x8608337C 50/0/9.0 2020 g711ulaw
20009/20031
0x117D 2C22 0x8616F7EC 50/0/31.0 *2020 g711ulaw
20031/20009
0x117E 2C25 0x86063990 50/0/6.0 2006 g711ulaw
20006/20001

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0x117F 2C25 0x85C6BE6C 50/0/1.0 *2006 g711ulaw
20001/20006
0x1180 2C28 0x860ADFF0 50/0/13.0 2029 g711ulaw
20013/20034
0x1181 2C28 0x8618FBBC 50/0/34.0 *2029 g711ulaw
20034/20013
0x1182 2C2B 0x860C3B1C 50/0/15.0 2036 g711ulaw
20015/20005
0x1183 2C2B 0x860590EC 50/0/5.0 *2036 g711ulaw
20005/20015
0x1184 2C2E 0x8617A090 50/0/32.0 2024 g711ulaw
20032/20007
0x1185 2C2E 0x8606E234 50/0/7.0 *2024 g711ulaw
20007/20032
0x1186 2C31 0x861A56E8 50/0/36.0 2030 g711ulaw
20036/20033
0x1187 2C31 0x86185318 50/0/33.0 *2030 g711ulaw
20033/20036
18 active calls found
Step 4 debug ephone register Use this command to debug the process of Cisco IP
phone registration.
Example:
Router# debug ephone register
EPHONE registration debugging is enabled
*Jun 29 09:16:02.180: New Skinny socket accepted [2]
(0 active)
*Jun 29 09:16:02.180: sin_family 2, sin_port 51617,
in_addr 10.5.43.177
*Jun 29 09:16:02.180: skinny_socket_process: secure
skinny sessions = 1
*Jun 29 09:16:02.180: add_skinny_secure_socket: pid
=155, new_sock=0, ip address = 10.5.43.177
*Jun 29 09:16:02.180: skinny_secure_handshake: pid
=155, sock=0, args->pid=155, ip address =
10.5.43.177
*Jun 29 09:16:02.184: Start TLS Handshake 0
10.5.43.177 51617
*Jun 29 09:16:02.184: TLS Handshake retcode
OPSSLReadWouldBlockErr
*Jun 29 09:16:08.188: CRYPTO_PKI_OPSSL - Verifying 1
Certs
*Jun 29 09:16:08.212: TLS Handshake completes

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Step 5 debug ephone state Use this command to review call setup between two
secure Cisco Unified IP Phones. The debug ephone
state trace shows the generation and distribution of
Example:
Router# debug ephone state
encryption and decryption keys between the two
*Jan 11 18:33:09.231:%SYS-5-CONFIG_I:Configured from phones.
console by console
*Jan 11 18:33:11.747:ephone-2[2]:OFFHOOK
*Jan 11
18:33:11.747:ephone-2[2]:---SkinnySyncPhoneDnOverlay
s is onhook
*Jan 11 18:33:11.747:ephone-2[2]:SIEZE on activeLine
0 activeChan 1
*Jan 11 18:33:11.747:ephone-2[2]:SetCallState line 1
DN 2(-1) chan 1 ref 6 TsOffHook
*Jan 11 18:33:11.747:ephone-2[2]:Check Plar Number
*Jan 11 18:33:11.751:DN 2 chan 1 Voice_Mode
*Jan 11 18:33:11.751:dn_tone_control DN=2 chan 1
tonetype=33:DtInsideDialTone onoff=1 pid=232
tonetype=0:DtSilence onoff=0 pid=232
*Jan 11 18:33:16.039:ephone-2[2]:Skinny-to-Skinny
call DN 2 chan 1 to DN 4 chan 1 instance 1
DN 2(-1) chan 1 ref 6 TsProceed
DN 2(-1) chan 1 ref 6 TsRingOut
*Jan 11 18:33:16.039:ephone-2[2]::callingNumber 6000
*Jan 11 18:33:16.039:ephone-2[2]::callingParty 6000
*Jan 11 18:33:16.039:ephone-2[2]:Call Info DN 2 line
1 ref 6 call state 1 called 6001 calling 6000
origcalled
1 ref 6 called 6001 calling 6000 origcalled 6001
calltype 2
*Jan 11 18:33:16.039:ephone-2[2]:Call Info for chan
1
*Jan 11 18:33:16.039:ephone-2[2]:Original Called
Name 6001
*Jan 11 18:33:16.039:ephone-2[2]:6000 calling
*Jan 11 18:33:16.039:ephone-2[2]:6001
DN 4(4) chan 1 ref 7 TsRingIn
origcalled
calltype 1
1
Name 6001
*Jan 11 18:33:16.047:ephone-3[3]:6001
*Jan 11 18:33:16.047:ephone-3[3]:Ringer Inside Ring
On

9-276

tonetype=36:DtAlertingTone onoff=1 pid=232
*Jan 11 18:33:20.831:ephone-3[3]:OFFHOOK
*Jan 11
18:33:20.831:ephone-3[3]:---SkinnySyncPhoneDnOverlay
s is onhook
*Jan 11 18:33:20.831:ephone-3[3]:Ringer Off
*Jan 11 18:33:20.831:ephone-3[3]:ANSWER call
DN 4(-1) chan 1 ref 7 TsOffHook
*Jan 11
18:33:20.831:ephone-3[3][SEP000DEDAB3EBF]:Answer
Incoming call from ephone-(2) DN 2 chan 1
DN 4(-1) chan 1 ref 7 TsConnected
*Jan 11 18:33:20.831:defer_start for DN 2 chan 1 at
CONNECTED
DN 2(-1) chan 1 ref 6 TsConnected
origcalled
calltype 1
1
Name 6001
*Jan 11 18:33:20.835:ephone-3[3]:6001
*Jan 11 18:33:20.835:ephone-2[2]:Security Key
Generation
! Ephone 2 generates a security key.
*Jan 11 18:33:20.835:ephone-2[2]:OpenReceive DN 2
chan 1 codec 4:G711Ulaw64k duration 20 ms bytes 160
*Jan 11 18:33:20.835:ephone-2[2]:Send Decryption Key
! Ephone 2 sends the decryption key.
*Jan 11 18:33:20.835:ephone-3[3]:Security Key
Generation
!Ephone 3 generates its security key.
*Jan 11 18:33:20.835:ephone-3[3]:OpenReceive DN 4
chan 1 codec 4:G711Ulaw64k duration 20 ms bytes 160
*Jan 11 18:33:20.835:ephone-3[3]:Send Decryption Key
! Ephone 3 sends its decryption key.

tonetype=0:DtSilence onoff=0 pid=232
*Jan 11 18:33:21.091:DN 2 chan 1 End Voice_Mode
*Jan 11
18:33:21.095:ephone-2[2]:OpenReceiveChannelAck:IP
1.1.1.8, port=25552,
dn_index=2, dn=2, chan=1

9-277

*Jan 11 18:33:21.095:ephone-3[3]:StartMedia 1.1.1.8
port=25552
*Jan 11 18:33:21.095:DN 2 chan 1 codec 4:G711Ulaw64k
duration 20 ms bytes 160
*Jan 11 18:33:21.095:ephone-3[3]:Send Encryption Key
! Ephone 3 sends its encryption key.
*Jan 11
18:33:21.347:ephone-3[3]:OpenReceiveChannelAck:IP
1.1.1.9, port=17520,
dn_index=4, dn=4, chan=1
*Jan 11 18:33:21.347:ephone-2[2]:StartMedia 1.1.1.9
port=17520
*Jan 11 18:33:21.347:DN 2 chan 1 codec 4:G711Ulaw64k
duration 20 ms bytes 160
*Jan 11 18:33:21.347:ephone-2[2]:Send Encryption Key
!Ephone 2 sends its encryption key.*Jan 11
18:33:21.851:ephone-2[2]::callingNumber 6000
origcalled
calltype 2
1
Name 6001
*Jan 11 18:33:21.851:ephone-2[2]:6001
Configuration Examples for Secure SCCP SRST

This section provides the following configuration examples:
• Secure SCCP SRST: Example, page 278
• Control Plane Policing: Example, page 283
Note IP addresses and hostnames in examples are fictitious.
Secure SCCP SRST: Example
This section provides a configuration example to match the identified configuration tasks in the previous
sections. This example does not include using a third-party CA; it assumes the use of the Cisco IOS
certificate server to generate your certificates.
.
.
.
! Define Unified Communications Manager.
ccm-manager fallback-mgcp
ccm-manager mgcp
ccm-manager music-on-hold
ccm-manager config server 10.1.1.13
ccm-manager config

9-278
!
! Define root CA.
crypto pki server srstcaserver
database level complete
database url nvram
issuer-name CN=srstcaserver
!
crypto pki trustpoint srstca
enrollment url http://10.1.1.22:80
revocation-check none
!
crypto pki trustpoint srstcaserver
rsakeypair srstcaserver
!
! Define CTL/7970 trustpoint.
crypto pki trustpoint 7970
enrollment terminal
!
crypto pki trustpoint PEM
enrollment terminal
!
! Define CAPF/7960 trustpoint.
crypto pki trustpoint 7960
enrollment terminal
!
! SRST router device certificate.
crypto pki certificate chain srstca
certificate 02
308201AD 30820116 A0030201 02020102 300D0609 2A864886 F70D0101 04050030
17311530 13060355 0403130C 73727374 63617365 72766572 301E170D 30343034
31323139 35323233 5A170D30 35303431 32313935 3232335A 30343132 300F0603
55040513 08443042 39453739 43301F06 092A8648 86F70D01 09021612 6A61736F
32363931 2E636973 636F2E63 6F6D305C 300D0609 2A864886 F70D0101 01050003
4B003048 024100D7 0CC354FB 5F7C1AE7 7A25C3F2 056E0485 22896D36 6CA70C19
C98F9BAE AE9D1F9B D4BB7A67 F3251174 193BB1A3 12946123 E5C1CCD7 A23E6155
FA2ED743 3FB8B902 03010001 A330302E 300B0603 551D0F04 04030205 A0301F06
03551D23 04183016 8014F829 CE97AD60 18D05467 FC293963 C2470691 F9BD300D
06092A86 4886F70D 01010405 00038181 007EB48E CAE9E1B3 D1E7A185 D7F0D565
CB84B17B 1151BD78 B3E39763 59EC650E 49371F6D 99CBD267 EB8ADF9D 9E43A5F2
FB2B18A0 34AF6564 11239473 41478AFC A86E6DA1 AC518E0B 8657CEBB ED2BDE8E
B586FE67 00C358D4 EFDD8D44 3F423141 C2D331D3 1EE43B6E 6CB29EE7 0B8C2752
C3AF4A66 BD007348 D013000A EA3C206D CF
quit
certificate ca 01
30820207 30820170 A0030201 02020101 300D0609 2A864886 F70D0101 04050030
17311530 13060355 0403130C 73727374 63617365 72766572 301E170D 30343034
31323139 34353136 5A170D30 37303431 32313934 3531365A 30173115 30130603
55040313 0C737273 74636173 65727665 7230819F 300D0609 2A864886 F70D0101
01050003 818D0030 81890281 8100C3AF EE1E4BB1 9922A8DA 2BB9DC8E 5B1BD332
1051C9FE 32A971B3 3C336635 74691954 98E765B1 059E24B6 32154E99 105CA989
9619993F CC72C525 7357EBAC E6335A32 2AAF9391 99325BFD 9B8355EB C10F8963
9D8FC222 EE8AC831 71ACD3A7 4E918A8F D5775159 76FBF499 5AD0849D CAA41417
DD866902 21E5DD03 C37D4B28 0FAB0203 010001A3 63306130 0F060355 1D130101
FF040530 030101FF 300E0603 551D0F01 01FF0404 03020186 301D0603 551D0E04
160414F8 29CE97AD 6018D054 67FC2939 63C24706 91F9BD30 1F060355 1D230418
30168014 F829CE97 AD6018D0 5467FC29 3963C247 0691F9BD 300D0609 2A864886
F70D0101 04050003 8181007A F71B25F9 73D74552 25DFD03A D8D1338F 6792C805
47A81019 795B5AAE 035400BB F859DABF 21892B5B E71A8283 08950414 8633A8B2
C98565A6 C09CA641 88661402 ACC424FD 36F23360 ABFF4C55 BB23C66A C80A3A57

9-279
5EE85FF8 C1B1A540 E818CE6D 58131726 BB060974 4E1A2F4B E6195522 122457F3

quit
crypto pki certificate chain srstcaserver
certificate ca 01
30820207 30820170 A0030201 02020101 300D0609 2A864886 F70D0101 04050030
17311530 13060355 0403130C 73727374 63617365 72766572 301E170D 30343034
31323139 34353136 5A170D30 37303431 32313934 3531365A 30173115 30130603
55040313 0C737273 74636173 65727665 7230819F 300D0609 2A864886 F70D0101
01050003 818D0030 81890281 8100C3AF EE1E4BB1 9922A8DA 2BB9DC8E 5B1BD332
1051C9FE 32A971B3 3C336635 74691954 98E765B1 059E24B6 32154E99 105CA989
9619993F CC72C525 7357EBAC E6335A32 2AAF9391 99325BFD 9B8355EB C10F8963
9D8FC222 EE8AC831 71ACD3A7 4E918A8F D5775159 76FBF499 5AD0849D CAA41417
DD866902 21E5DD03 C37D4B28 0FAB0203 010001A3 63306130 0F060355 1D130101
FF040530 030101FF 300E0603 551D0F01 01FF0404 03020186 301D0603 551D0E04
160414F8 29CE97AD 6018D054 67FC2939 63C24706 91F9BD30 1F060355 1D230418
30168014 F829CE97 AD6018D0 5467FC29 3963C247 0691F9BD 300D0609 2A864886
F70D0101 04050003 8181007A F71B25F9 73D74552 25DFD03A D8D1338F 6792C805
47A81019 795B5AAE 035400BB F859DABF 21892B5B E71A8283 08950414 8633A8B2
C98565A6 C09CA641 88661402 ACC424FD 36F23360 ABFF4C55 BB23C66A C80A3A57
5EE85FF8 C1B1A540 E818CE6D 58131726 BB060974 4E1A2F4B E6195522 122457F3
quit
crypto pki certificate chain 7970
certificate ca 353FB24BD70F14A346C1F3A9AC725675
308203A8 30820290 A0030201 02021035 3FB24BD7 0F14A346 C1F3A9AC 72567530
0D06092A 864886F7 0D010105 0500302E 31163014 06035504 0A130D43 6973636F
20537973 74656D73 31143012 06035504 03130B43 41502D52 54502D30 3032301E
170D3033 31303130 32303138 34395A17 0D323331 30313032 30323733 375A302E
31163014 06035504 0A130D43 6973636F 20537973 74656D73 31143012 06035504
03130B43 41502D52 54502D30 30323082 0120300D 06092A86 4886F70D 01010105
00038201 0D003082 01080282 010100C4 266504AD 7DC3FD8D 65556FA6 308FAE95
B570263B 575ABD96 1CC8F394 5965D9D0 D8CE02B9 F808CCD6 B7CD8C46 24801878
57DC4440 A7301DDF E40FB1EF 136212EC C4F3B50F BCAFBB4B CD2E5826 34521B65
01555FE4 D4206776 03368357 83932638 D6FC953F 3A179E44 67255A73 45C69DEE
FB4D221B 21D7A3AD 38184171 8FD8C271 42183E65 09461434 736C77CC F380EEBF
632C7B3F A5F92AA6 A8EF3490 8724A84F 4DAF7FD7 0928F585 764D3558 3C0FE9AF
1ED8763F A299A802 970004AD 1912D265 7DE335B4 BCB6F789 DC68B9FA C8FDF85E
8A28AD8F 0F4883C0 77112A47 141DBEE0 948FBE53 FE67B308 D40C8029 87BD790E
CDAB9FD7 A190C1A2 A462C5F2 4A6E0B02 0103A381 C33081C0 300B0603 551D0F04
04030201 86300F06 03551D13 0101FF04 05300301 01FF301D 0603551D 0E041604
1452922B E288EE2E 098A4E7E 702C56A5 9AB4D49B 96306F06 03551D1F 04683066
3064A062 A060862D 68747470 3A2F2F63 61702D72 74702D30 30322F43 65727445
6E726F6C 6C2F4341 502D5254 502D3030 322E6372 6C862F66 696C653A 2F2F5C5C
6361702D 7274702D 3030325C 43657274 456E726F 6C6C5C43 41502D52 54502D30
30322E63 726C3010 06092B06 01040182 37150104 03020100 300D0609 2A864886
F70D0101 05050003 82010100 56838CEF C4DA3AD1 EA8FBB15 2FFE6EE5 50A1972B
D4D7AF1F D298892C D5A2A76B C3462866 13E0E55D DC0C4B92 5AA94B6E 69277F9B
FC73C697 11266E19 451C0FAB A55E6A28 901A48C5 B9911EE6 348A8920 0AEDE1E0
B6EA781C FFD97CA4 B03C0E34 0E5B0649 8B0A34C9 B73A654E 09050C1F 4DA53E44
BF78443D B08C3A41 2EEEB873 78CB8089 34F9D16E 91512F0D 3A8674AD 0991ED1A
92841E76 36D7740E CB787F11 685B9E9D 0C67E85D AF6D05BA 3488E86D 7E2F7F65
6918DE0F BD3C7F67 D8A33F70 9C4A596E D9F62B3B 1EDEE854 D5882AD4 3D71F72B
8FAB7F3C 0B5F0759 D9828F83 954D7BB1 57A638EC 7D72BFF1 8933C16F 760BCA94
4C5B1931 67947A4F 89A1BDB5
quit
crypto pki certificate chain PEM
certificate ca 7612F960153D6F9F4E42202032B72356
308203A8 30820290 A0030201 02021076 12F96015 3D6F9F4E 42202032 B7235630
0D06092A 864886F7 0D010105 0500302E 31163014 06035504 0A130D43 6973636F
20537973 74656D73 31143012 06035504 03130B43 41502D52 54502D30 3031301E
170D3033 30323036 32333237 31335A17 0D323330 32303632 33333633 345A302E
31163014 06035504 0A130D43 6973636F 20537973 74656D73 31143012 06035504
03130B43 41502D52 54502D30 30313082 0120300D 06092A86 4886F70D 01010105

9-280
00038201 0D003082 01080282 010100AC 55BBED18 DE9B8709 FFBC8F2D 509AB83A

21C1967F DEA7F4B0 969694B7 80CC196A 463DA516 54A28F47 5D903B5F 104A3D54
A981389B 2FC7AC49 956262B8 1C143038 5345BB2E 273FA7A6 46860573 CE5C998D
55DE78AA 5A5CFE14 037D695B AC816409 C6211F0B 3BBF09CF B0BBB2D4 AC362F67
0FD145F1 620852B3 1F07E2F1 AA74F150 367632ED A289E374 AF0C5B78 CE7DFB9F
C8EBBE54 6ECF4C77 99D6DC04 47476C0F 36E58A3B 6BCB24D7 6B6C84C2 7F61D326
BE7CB4A6 60CD6579 9E1E3A84 8153B750 5527E865 423BE2B5 CB575453 5AA96093
58B6A2E4 AA3EF081 C7068EC1 DD1EBDDA 53E6F0D6 E2E0486B 109F1316 78C696A3
CFBA84CC 7094034F C1EB9F81 931ACB02 0103A381 C33081C0 300B0603 551D0F04
04030201 86300F06 03551D13 0101FF04 05300301 01FF301D 0603551D 0E041604
14E917B1 82C71FCF ACA91B6E F4A9269C 70AE05A0 9A306F06 03551D1F 04683066
3064A062 A060862D 68747470 3A2F2F63 61702D72 74702D30 30312F43 65727445
6E726F6C 6C2F4341 502D5254 502D3030 312E6372 6C862F66 696C653A 2F2F5C5C
6361702D 7274702D 3030315C 43657274 456E726F 6C6C5C43 41502D52 54502D30
30312E63 726C3010 06092B06 01040182 37150104 03020100 300D0609 2A864886
F70D0101 05050003 82010100 AB64FDEB F60C32DC 360F0E10 5FE175FA 0D574AB5
02ACDCA3 C7BBED15 A4431F20 7E9286F0 770929A2 17E4CDF4 F2629244 2F3575AF
E90C468C AE67BA08 AAA71C12 BA0C0E79 E6780A5C F814466C 326A4B56 73938380
73A11AED F9B9DE74 1195C48F 99454B8C 30732980 CD6E7123 8B3A6D68 80B97E00
7F4BD4BA 0B5AB462 94D9167E 6D8D48F2 597CDE61 25CFADCC 5BD141FB 210275A2
0A4E3400 1428BA0F 69953BB5 50D21F78 43E3E563 98BCB2B1 A2D4864B 0616BACD
A61CD9AE C5558A52 B5EEAA6A 08F96528 B1804B87 D26E4AEE AB7AFFE9 2FD2A574
BAFE0028 96304A8B 13FB656D 8FC60094 D5A53D71 444B3CEF 79343385 3778C193
74A2A6CE DC56275C A20A303D
quit
crypto pki certificate chain 7960
certificate ca F301
308201F7 30820160 A0030201 020202F3 01300D06 092A8648 86F70D01 01050500
3041310B 30090603 55040613 02555331 1A301806 0355040A 13114369 73636F20
53797374 656D7320 496E6331 16301406 03550403 130D4341 50462D33 35453038
33333230 1E170D30 34303430 39323035 3530325A 170D3139 30343036 32303535
30315A30 41310B30 09060355 04061302 5553311A 30180603 55040A13 11436973
636F2053 79737465 6D732049 6E633116 30140603 55040313 0D434150 462D3335
45303833 33323081 9F300D06 092A8648 86F70D01 01010500 03818D00 30818902
818100C8 BD9B6035 366B44E8 0F693A47 250FF865 D76C35F7 89B1C4FD 1D122CE0
F5E5CDFF A4A87EFF 41AD936F E5C93163 3E55D11A AF82A5F6 D563E21C EB89EBFA
F5271423 C3E875DC E0E07967 6E1AAB4F D3823E12 53547480 23BA1A09 295179B6
85A0E83A 77DD0633 B9710A88 0890CD4D DB55ADD0 964369BA 489043BB B667E60F
93954B02 03010001 300D0609 2A864886 F70D0101 05050003 81810056 60FD3AB3
6F98D2AD 40C309E2 C05B841C 5189271F 01D864E8 98BCE665 2AFBCC8C 54007A84
8F772C67 E3047A6C C62F6508 B36A6174 B68C1D78 C2228FEA A89ECEFB CC8BA9FC
0F30E151 431670F9 918514D9 868D1235 18137F1E 50DFD32E 1DC29CB7 95EF4096
421AF22F 5C1D5804 B83F8E8E 95B04F45 86563BFE DF976C5B FB490A
quit
!
!
no crypto isakmp enable
!
! Enable IPSec.
crypto isakmp policy 1
authentication pre-share
lifetime 28800
crypto isakmp key cisco123 address 10.1.1.13
! The crypto key should match the key configured on Cisco Unified Communications Manager.
!
! The crypto IPSec configuration should match your Cisco Unified Communications Manager
configuration.
crypto ipsec transform-set rtpset esp-des esp-md5-hmac

!
!
crypto map rtp 1 ipsec-isakmp
set peer 10.1.1.13
set transform-set rtpset

9-281
match address 116

!
!
interface FastEthernet0/0
ip address 10.1.1.22 255.255.255.0
duplex auto
speed auto
crypto map rtp
!
no ip address
shutdown
duplex auto
speed auto
!
ip classless
!
ip http server
no ip http secure-server
!
!
! Define traffic to be encrypted by IPSec.
access-list 116 permit ip host 10.1.1.22 host 10.1.1.13
!
!
control-plane
!
!
call application alternate DEFAULT
!
!
voice-port 1/0/0
!
voice-port 1/0/1
!
voice-port 1/0/2
!
voice-port 1/0/3
!
voice-port 1/1/0
timing hookflash-out 50
!
voice-port 1/1/1
!
voice-port 1/1/2
!
voice-port 1/1/3
!
! Enable MGCP voice protocol.
mgcp
mgcp call-agent 10.1.1.13 2427 service-type mgcp version 0.1
mgcp dtmf-relay voip codec all mode out-of-band
mgcp rtp unreachable timeout 1000 action notify
mgcp package-capability rtp-package
mgcp package-capability sst-package
no mgcp package-capability fxr-package
no mgcp timer receive-rtcp
mgcp sdp simple
mgcp fax t38 inhibit
mgcp rtp payload-type g726r16 static
!
!
!

9-282

application mgcpapp
port 1/1/0
forward-digits all
!
application mgcpapp
port 1/0/0
!
application mgcpapp
port 1/0/0
!
application mgcpapp
port 1/1/0
!
!
! Enable credentials service on the gateway.
credentials
trustpoint srstca
!
!
! Enable SRST mode.
transport-tcp-tls
secondary-dialtone 9
max-ephones 15
max-dn 30
transfer-pattern .....
.
.
.
Control Plane Policing: Example
This section provides a configuration example for the security best practice of protecting the credentials
service port using control plane policing. Control plane policing protects the gateway and maintains
packet forwarding and protocol states despite a heavy traffic load. For more information on control
planes, see the Control Plane Policing documentation.
.
.
.
! Allow trusted host traffic.
access-list 140 deny tcp host 10.1.1.11 any eq 2445
! Rate-limit all other traffic.

access-list 140 permit tcp any any eq 2445
access-list 140 deny ip any any
! Define class-map "sccp-class."

class-map match-all sccp-class
match access-group 140
policy-map control-plane-policy
class sccp-class

9-283
police 8000 1500 1500 conform-action drop exceed-action drop

! Define aggregate control plane service for the active Route Processor.
control-plane
service-policy input control-plane-policy
Configuring Secure SIP Call Signaling and SRTP Media with Cisco SRST
Cisco Unified Survivable Remote Site Telephony (Cisco SRST) provides secure call signaling and
Secure Real-time Transport Protocol (SRTP) for media encryption to establish a secure, encrypted
connection between Cisco Unified IP Phones and gateway devices.
• Prerequisites for Configuring Secure SIP Call Signaling and SRTP Media with Cisco SRST,
page 284
• Restrictions for Configuring Secure SIP Call Signaling and SRTP Media with Cisco SRST, page 284
• Information About Cisco Unified SIP SRST Support of Secure SIP Signaling and SRTP Media,
page 285
• Configuring Cisco Unified Communications Manager, page 285
• Configuring Phones, page 286
• Configuring SIP options for Secure SIP SRST, page 287
• Configuring SIP SRST Security Policy, page 288 (optional)
• Configuring SIP User Agent for Secure SIP SRST, page 289 (optional)
• Verifying the Configuration, page 291
• Configuration Example for Cisco Unified SIP SRST, page 293
Prerequisites for Configuring Secure SIP Call Signaling and SRTP Media with Cisco SRST
• Cisco IOS Release 15.0(1)XA and later releases.
• Cisco Unified IP Phone firmware release 8.5(3) or later.
• Complete the prerequisites and necessary tasks found in Prerequisites for Configuring SIP SRST
Features Using Back-to-Back User Agent Mode.
• Prepare the Cisco Unified SIP SRST device to use certificates as documented in Preparing the Cisco
Unified SRST Router for Secure Communication.
Restrictions for Configuring Secure SIP Call Signaling and SRTP Media with Cisco SRST
SIP phones may be configured on the Cisco Unified CM with an authenticated device security mode.
The Cisco Unified CM ensures integrity and authentication for the phone using a TLS connection with
NULL-SHA cipher for signaling. If an authenticated SIP phone fails over to the Cisco Unified SRST
device, it will register using TCP instead of TLS/TCP, thus disabling the authenticated mode until the
phone fails back to the Cisco Unified CM.
• By default, non-secure TCP SIP phones are permitted to register to the SRST device on failover from
the primary call control. Support for TCP SIP phones requires the secure SRST configuration
described in this section even if no encrypted phones are deployed. Without the secure SIP SRST
configuration, TCP phones will register to the SRST device using UDP for signaling transport.

9-284
Information About Cisco Unified SIP SRST Support of Secure SIP Signaling and SRTP Media
Beginning with Cisco IP Phone firmware 8.5(3) and Cisco IOS Release 15.0(1)XA, Cisco SRST
supports SIP signaling over UDP, TCP, and TLS connections, providing both RTP and SRTP media
connections based on the security settings of the IP phone.
Cisco SRST SIP-to-SIP and SIP-to-PSTN support includes the following features:
• Basic calling
• Hold/resume
• Conference
• Transfer
• Blind transfer
• Call forward
Cisco SRST SIP-to-other (including SIP-to-SCCP) support includes basic calling, although other
features may work.
Configuring Cisco Unified Communications Manager

Like SCCP-controlled devices, SIP-controlled devices will use the SRST Reference profile that is listed
in their assigned Device Pool. The SRST Reference profile must have the "Is SRST Secure" check box
selected if SIP/TLS communication is desired in the event of a WAN failure.
Note All Cisco Unified IP Phones must have their firmware updated to version 8.5(3) or later. Devices with
firmware earlier than 8.5(3) will need to have a separate Device Pool and SRST Reference profile created
without the "Is SRST Secure" option selected; SIP-controlled devices in this Device Pool will use SIP
over UDP to attempt to register to the SRST router.
In Cisco Unified CM Administration, under System > SRST:

• For the secure SRST profile, Is SRST Secure? must be checked. The SIP port must be 5061.
• For the non-secure SRST profile, the Is SRST Secure? checkbox should NOT be checked and the
SIP port should be 5060.
Under Device > Phone:
• Secure phones must belong to the pool that uses the secure SRST profile.
• Non-secure phones must belong to the pool that uses the non-secure SRST profile.
Note SIP phones will use the transport method assigned to them by their Phone Security Profile.

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Configuring Phones
This section specifies that SRTP should be used to enable secure calls and allows non-secure calls to
"fallback" to using RTP media.
SUMMARY STEPS
1. enable
4. srtp
5. allow-connections sip to h323
7. end

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DETAILED STEPS

Example:
Router> enable
Example:
Example::
Step 4 srtp Specifies that SRTP be used to enable secure calls.
Example::
Router(config-voi-serv)# srtp
Step 5 allow-connections sip to h323 (Optional) Allows connections from SIP endpoints to H.323
endpoints.
Example:
to h323
Step 6 allow-connections sip to sip Allows connections from SIP endpoints to SIP endpoints.
Example:
to sip
Step 7 end Ends the current configuration session and returns to
Example:
Router(conf-voi-serv)# end
Configuring SIP options for Secure SIP SRST

This section explains how to configure secure SIP SRTP.
SUMMARY STEPS
1. enable
4. sip
5. url sip | sips
6. srtp negotiate cisco

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7. end
DETAILED STEPS

Example:
Router> enable
Example:
Example::
Example:
Step 5 url sip | sips To configure secure mode, use the sips keyword to generate
URLs in SIP secure (SIPS) format for VoIP calls.
Example: To configure device-default mode, use the sip keyword to
Router(conf-serv-sip)# url sips generate URLs in SIP format for VoIP calls.
Step 6 srtp negotiate cisco Enables a Cisco IOS SIP gateway to negotiate the sending
and accepting of RTP profiles in response to SRTP offers.
Example:
Router(conf-serv-sip)# srtp negotiate cisco
Example:
Configuring SIP SRST Security Policy

This section explains how to secure mode to block registration of non-secure phones to the SRST router.
SUMMARY STEPS

2. security-policy secure | no security-policy
3. end

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DETAILED STEPS

Step 1 voice register global Enters voice register global configuration mode.
Example:
Step 2 security-policy secure Configures SIP registration security policy so that only
SIP/TLS/TCP connections are allowed. For device-default
mode, use the no security-policy command. Device-default
Example:
Router(config-register-global)# security-policy
mode allows non-secure devices to register without using
secure TLS.
Note We recommend that security-policy secure is
configured for the Secure SRST feature, so that
non-secure phones do not fall back on Secure SRST.
Example:
Configuring SIP User Agent for Secure SIP SRST

This section explains how the strict-cipher limits the allowed encryption algorithms.
SUMMARY STEPS
1. sip-ua
2. registrar ipv4:destination-address expires seconds
3. xfer target dial-peer
4. crypto signaling default trustpoint string [strict-cipher]
5. crypto signaling remote-addr {ip address |subnet mask} trustpoint trustpoint-name
6. end

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DETAILED STEPS

Example:
Step 2 registrar ipv4:destination-address expires Enables the gateway to register E.164 telephone numbers
seconds with primary and secondary external SIP registrars.
destination-address is the IP address of the primary SIP
Example: registrar server.
Router(config-sip-ua)# registrar
ipv4:192.168.2.10 expires 3600
Step 3 xfer target dial-peer Specifies that SRST should use the dial-peer as a transfer
target instead of what is in the message body.
Example:
Router(config-sip-ua)# xfer target dial-peer
Step 4 crypto signaling default trustpoint string Identifies the trustpoint string keyword and argument used
[strict-cipher] during the TLS handshake. The trustpoint string keyword
and argument refer to the gateway’s certificate generated as
Example: part of the enrollment process, using Cisco IOS public-key
Router(config-sip-ua)# crypto signaling default infrastructure (PKI) commands. The strict-cipher keyword
trustpoint 3745-SRST strict-cipher restricts support to TLS RSA encryption with the Advanced
Encryption Standard-128 (AES-128) cipher-block-chaining
(CBC) Secure Hash Algorithm (SHA)
(TLS_RSA_WITH_AES_128_CBC_SHA) cipher suite.
To configure device-default mode, omit the strict-cipher
keyword.
Step 5 crypto signaling remote-addr {ip address The trustpoint label refers to the CUBE’s certificate that is
|subnet mask} trustpoint trustpoint-name generated with the Cisco IOS PKI commands as part of the
enrollment process.
Keywords and arguments are as follows:
Example:
Router(config-sip-ua)# crypto signaling • remote-addr ip address—Associates an IP address to a
remote-addr 8.41.20.20 255.255.0.0 trustpoint trustpoint.
srst-trunk1
• trustpoint trustpoint-name—Refers to the SIP
gateways certificate generated as part of the enrollment
process using Cisco IOS PKI commands
Example:
Router(config-sip-ua)# end
Multiple Trustpoints
Use the default trustpoint configuration under sip-ua config mode for phones registering to Unified
SRST in secure mode. For example, srstca is the default trustpoint for Secure SRST. This default
signaling trustpoint is used for all SIP TLS interactions from SIP phones to Unified Secure SRST router.

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In a deployment scenario with multiple trustpoints, communication with a service provider over a secure
trunk with certificate issued by CA is achieved using the CLI command crypto signaling remote-addr
8.41.20.20 255.255.0.0 trustpoint srst-trunk1 under sip-ua config mode.
Example
The following example shows a sample configuration of multiple trustpoints for a Unified SRST
deployment. In this example, the srst-trunk1 trustpoint points to the network with IP address 8.39.0.0,
and srst-trunk2 trustpoint points to the network with IP address 8.41.20.20.
sip-ua
crypto signaling remote-addr 8.39.0.0 255.255.0.0 trustpoint srst-trunk1
crypto signaling remote-addr 8.41.20.20 255.255.0.0 trustpoint srst-trunk2
crypto signaling default trustpoint secsrst
Verifying the Configuration
The following examples show a sample configuration displayed by the show sip-ua status registrar
command and the show voice register global command.
The show sip-ua status registrar command in privileged EXEC mode displays all SIP endpoints that
are currently registered with the contact address.
transport call-id
peer
============ =============== ============ ===============
3029991 192.168.2.108 388 192.168.2.108
TLS [email protected]
40004
3029993 192.168.2.103 382 192.168.2.103
TCP [email protected]
40011
3029982 192.168.2.106 406 192.168.2.106
UDP [email protected]
40001
3029983 192.168.2.106 406 192.168.2.106
UDP [email protected]
40003
3029992 192.168.2.107 414 192.168.2.107
TLS [email protected]
40005
The show voice register global command in privileged EXEC mode displays all global configuration
parameters associated with SIP phones.
CONFIG [Version=8.0]
========================
Version 8.0
Mode is srst
Max-pool is 50
Max-dn is 100
Outbound-proxy is enabled and will use global configured value
Security Policy: DEVICE-DEFAULT
timeout interdigit 10
network-locale[0] US (This is the default network locale for this box)
network-locale[1] US
user-locale[0] US (This is the default user locale for this box)

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user-locale[1] US
user-locale[2] US
user-locale[3] US
user-locale[4] US
Router#

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Configuration Example for Cisco Unified SIP SRST

!
! Last configuration change at 05:34:06 UTC Tue Jun 13 2017
! NVRAM config last updated at 11:57:03 UTC Thu Jun 8 2017
!
version 16.7
platform qfp utilization monitor load 80
no platform punt-keepalive disable-kernel-core
!
hostname router
!
boot-start-marker
boot-end-marker
!
vrf definition Mgmt-intf

!
address-family ipv4
exit-address-family
!
address-family ipv6
exit-address-family
!
! card type command needed for slot/bay 0/3
no logging queue-limit
logging buffered 20000000
no logging rate-limit
no logging console
enable password xxxx
!
no aaa new-model
!
subscriber templating
!
multilink bundle-name authenticated
!
crypto pki server SRST-CA-2
database level complete
no database archive
grant auto
!
crypto pki trustpoint TRUSTPT-SRST-CA-2
serial-number
rsakeypair srstcakey 2048
rsakeypair SRST-CA-2
!
crypto pki trustpoint SRST-CA-2
revocation-check crl
rsakeypair SRST-CA-2
!
crypto pki trustpoint Cisco_Manufacturing_CA
enrollment terminal
!
crypto pki trustpoint CAPF-3a66269a
enrollment terminal

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!
crypto pki trustpoint Cisco_Root_CA_2048
enrollment terminal
!
!
crypto pki certificate chain TRUSTPT-SRST-CA-2
certificate 02
3082020B 30820174 A0030201 02020102 300D0609 2A864886 F70D0101 05050030
14311230 10060355 04031309 53525354 2D43412D 32301E17 0D313730 36303831
31333131 325A170D 31383036 30383131 33313132 5A303231 30301206 03550405
130B4647 4C313735 31313150 42301A06 092A8648 86F70D01 0902160D 416E7473
41726D79 2D343430 3030819F 300D0609 2A864886 F70D0101 01050003 818D0030
81890281 81009E24 6259A98D A61C1973 45A95DA8 DE83ECAD C2B1B448 741F7E64
3D753BF1 19BD54FB 9A4D4A8E 7A2BA416 B93C40B3 A63A7C4D 7303498F 098EF07F
96F26F5F 49AD4E39 EC113DF4 696CB887 607D545A 52A11469 958F4C04 05868DF9
317456F6 3D23837C D46331FA 69FB29E8 3211E01C A7AB19A3 94DAC09F 97601196
A08D7073 76210203 010001A3 4F304D30 0B060355 1D0F0404 030205A0 301F0603
551D2304 18301680 142110B8 F25BD9BD E1D401EC 9D11DC0E AE52CDB8 2F301D06
03551D0E 04160414 2110B8F2 5BD9BDE1 D401EC9D 11DC0EAE 52CDB82F 300D0609
2A864886 F70D0101 05050003 8181003A DC409694 26D08A31 7B4F495F 002D4E57
B28669A9 10E93C68 A9556659 97D326EC A5508201 C1A86659 B1CDC910 73097FCA
F6174794 1057DDDE DBA666D6 0BAFC503 96A10BE5 5FCA3B93 5D377ABE BC9B2774
3732DF01 CE3BF12B 1899AA69 F7EC8726 A1964C5A D6A99A0E E27EE2A0 15A7D364
793C6C8D 961C77E4 397F9CB4 C6A271
quit
certificate ca 01
30820201 3082016A A0030201 02020101 300D0609 2A864886 F70D0101 04050030
14311230 10060355 04031309 53525354 2D43412D 32301E17 0D313730 36303831
31323135 305A170D 32303036 30373131 32313530 5A301431 12301006 03550403
13095352 53542D43 412D3230 819F300D 06092A86 4886F70D 01010105 0003818D
00308189 02818100 9E246259 A98DA61C 197345A9 5DA8DE83 ECADC2B1 B448741F
7E643D75 3BF119BD 54FB9A4D 4A8E7A2B A416B93C 40B3A63A 7C4D7303 498F098E
F07F96F2 6F5F49AD 4E39EC11 3DF4696C B887607D 545A52A1 1469958F 4C040586
8DF93174 56F63D23 837CD463 31FA69FB 29E83211 E01CA7AB 19A394DA C09F9760
1196A08D 70737621 02030100 01A36330 61300F06 03551D13 0101FF04 05300301
01FF300E 0603551D 0F0101FF 04040302 0186301F 0603551D 23041830 16801421
10B8F25B D9BDE1D4 01EC9D11 DC0EAE52 CDB82F30 1D060355 1D0E0416 04142110
B8F25BD9 BDE1D401 EC9D11DC 0EAE52CD B82F300D 06092A86 4886F70D 01010405
00038181 0018859E D39C6A05 63509442 8746D970 BB716DE2 E82BA822 58AA55AD
AC37260F 36BFDFE6 F2D0E489 A8D23690 791AD903 F19AC857 5002E621 A5927ACC
DCB759C0 B126ACAB C53BF054 1F62D895 A895C50A E3AE83E3 EC68F346 50B88D39
BB053EE9 5D466AE4 C6B4593D 7EFA7A78 213C0766 7307A051 78FED92E 5A34AAB6
98D2A59C 31
quit
crypto pki certificate chain SRST-CA-2
certificate ca 01
30820201 3082016A A0030201 02020101 300D0609 2A864886 F70D0101 04050030
14311230 10060355 04031309 53525354 2D43412D 32301E17 0D313730 36303831
31323135 305A170D 32303036 30373131 32313530 5A301431 12301006 03550403
13095352 53542D43 412D3230 819F300D 06092A86 4886F70D 01010105 0003818D
00308189 02818100 9E246259 A98DA61C 197345A9 5DA8DE83 ECADC2B1 B448741F
7E643D75 3BF119BD 54FB9A4D 4A8E7A2B A416B93C 40B3A63A 7C4D7303 498F098E
F07F96F2 6F5F49AD 4E39EC11 3DF4696C B887607D 545A52A1 1469958F 4C040586
8DF93174 56F63D23 837CD463 31FA69FB 29E83211 E01CA7AB 19A394DA C09F9760
1196A08D 70737621 02030100 01A36330 61300F06 03551D13 0101FF04 05300301
01FF300E 0603551D 0F0101FF 04040302 0186301F 0603551D 23041830 16801421
10B8F25B D9BDE1D4 01EC9D11 DC0EAE52 CDB82F30 1D060355 1D0E0416 04142110
B8F25BD9 BDE1D401 EC9D11DC 0EAE52CD B82F300D 06092A86 4886F70D 01010405
00038181 0018859E D39C6A05 63509442 8746D970 BB716DE2 E82BA822 58AA55AD
AC37260F 36BFDFE6 F2D0E489 A8D23690 791AD903 F19AC857 5002E621 A5927ACC
DCB759C0 B126ACAB C53BF054 1F62D895 A895C50A E3AE83E3 EC68F346 50B88D39
BB053EE9 5D466AE4 C6B4593D 7EFA7A78 213C0766 7307A051 78FED92E 5A34AAB6
98D2A59C 31

9-294
quit
crypto pki certificate chain Cisco_Manufacturing_CA
certificate ca 6A6967B3000000000003
308204D9 308203C1 A0030201 02020A6A 6967B300 00000000 03300D06 092A8648
86F70D01 01050500 30353116 30140603 55040A13 0D436973 636F2053 79737465
6D73311B 30190603 55040313 12436973 636F2052 6F6F7420 43412032 30343830
1E170D30 35303631 30323231 3630315A 170D3239 30353134 32303235 34325A30
39311630 14060355 040A130D 43697363 6F205379 7374656D 73311F30 1D060355
04031316 43697363 6F204D61 6E756661 63747572 696E6720 43413082 0120300D
06092A86 4886F70D 01010105 00038201 0D003082 01080282 010100A0 C5F7DC96
943515F1 F4994EBB 9B41E17D DB791691 BBF354F2 414A9432 6262C923 F79AE7BB
9B79E807 294E30F5 AE1BC521 5646B0F8 F4E68E81 B816CCA8 9B85D242 81DB7CCB
94A91161 121C5CEA 33201C9A 16A77DDB 99066AE2 36AFECF8 0AFF9867 07F430EE
A5F8881A AAE8C73C 1CCEEE48 FDCD5C37 F186939E 3D71757D 34EE4B14 A9C0297B
0510EF87 9E693130 F548363F D8ABCE15 E2E8589F 3E627104 8726A415 620125AA
D5DFC9C9 5BB8C9A1 077BBE68 92939320 A86CBD15 75D3445D 454BECA8 DA60C7D8
C8D5C8ED 41E1F55F 578E5332 9349D5D9 0FF836AA 07C43241 C5A7AF1D 19FFF673
99395A73 67621334 0D1F5E95 70526417 06EC535C 5CDB6AEA 35004102 0103A382
01E73082 01E33012 0603551D 130101FF 04083006 0101FF02 0100301D 0603551D
0E041604 14D0C522 26AB4F46 60ECAE05 91C7DC5A D1B047F7 6C300B06 03551D0F
04040302 01863010 06092B06 01040182 37150104 03020100 30190609 2B060104
01823714 02040C1E 0A005300 75006200 43004130 1F060355 1D230418 30168014
27F3C815 1E6E9A02 0916AD2B A089605F DA7B2FAA 30430603 551D1F04 3C303A30
38A036A0 34863268 7474703A 2F2F7777 772E6369 73636F2E 636F6D2F 73656375
72697479 2F706B69 2F63726C 2F637263 61323034 382E6372 6C305006 082B0601
05050701 01044430 42304006 082B0601 05050730 02863468 7474703A 2F2F7777
772E6369 73636F2E 636F6D2F 73656375 72697479 2F706B69 2F636572 74732F63
72636132 3034382E 63657230 5C060355 1D200455 30533051 060A2B06 01040109
15010200 30433041 06082B06 01050507 02011635 68747470 3A2F2F77 77772E63
6973636F 2E636F6D 2F736563 75726974 792F706B 692F706F 6C696369 65732F69
6E646578 2E68746D 6C305E06 03551D25 04573055 06082B06 01050507 03010608
2B060105 05070302 06082B06 01050507 03050608 2B060105 05070306 06082B06
01050507 0307060A 2B060104 0182370A 0301060A 2B060104 01823714 02010609
2B060104 01823715 06300D06 092A8648 86F70D01 01050500 03820101 0030F330
2D8CF2CA 374A6499 24290AF2 86AA42D5 23E8A2EA 2B6F6923 7A828E1C 4C09CFA4
4FAB842F 37E96560 D19AC6D8 F30BF5DE D027005C 6F1D91BD D14E5851 1DC9E3F7
38E7D30B D168BE8E 22A54B06 E1E6A4AA 337D1A75 BA26F370 C66100A5 C379265B
A719D193 8DAB9B10 11291FA1 82FDFD3C 4B6E65DC 934505E9 AF336B67 23070686
22DAEBDC 87CF5921 421AE9CF 707588E0 243D5D7D 4E963880 97D56FF0 9B71D8BA
6019A5B0 6186ADDD 6566F6B9 27A2EE2F 619BBAA1 3061FDBE AC3514F9 B82D9706
AFC3EF6D CC3D3CEB 95E981D3 8A5EB6CE FA79A46B D7A25764 C43F4CC9 DBE882EC
0166D410 88A256E5 3C57EDE9 02A84891 6307AB61 264B1A13 9FE4DCDA 5F
quit
crypto pki certificate chain CAPF-3a66269a
certificate ca 583BD5B4844C8BC172B8C4979092A067
308203C3 308202AB A0030201 02021058 3BD5B484 4C8BC172 B8C49790 92A06730
0D06092A 864886F7 0D01010B 05003071 310B3009 06035504 06130249 4E310E30
0C060355 040A0C05 63697363 6F311230 10060355 040B0C09 75637467 2D656467
65311630 14060355 04030C0D 43415046 2D336136 36323639 61311230 10060355
04080C09 6B61726E 6174616B 61311230 10060355 04070C09 62616E67 616C6F72
65301E17 0D313730 35323931 30333631 335A170D 32323035 32383130 33363132
5A307131 0B300906 03550406 1302494E 310E300C 06035504 0A0C0563 6973636F
31123010 06035504 0B0C0975 6374672D 65646765 31163014 06035504 030C0D43
4150462D 33613636 32363961 31123010 06035504 080C096B 61726E61 74616B61
31123010 06035504 070C0962 616E6761 6C6F7265 30820122 300D0609 2A864886
F70D0101 01050003 82010F00 3082010A 02820101 00BC774F BAED3986 05BDFFBC
4EABBFA7 1F73D150 2989EFF2 902502F6 248DA7AB 261E474C 08A4BB6F 35B10449
0A6A3D94 E2C6EB98 57BECE0C 34F30517 CA6CC9B2 710B511B 8826E0AB 733FF26F
F7ADC4B9 76118300 6156072C 43F78E5E 3AD7C92B 54CB5BDB 00B53FC8 875100C4
056BC4A7 0F96CE69 E58B1C22 194CCEC6 968ECF9B 08B7B7B2 0FF0800E 43764BB1
E6ED36C0 A738F762 81A88F6D E464E2A5 FD74207F 1EC7ACAC 2F63B04D E0E9DA4C
901A1710 E3D1C069 82EFF77E 0597254D 149C1263 EC67DAE9 305FD8BF C7410B17
8C6DE9FF 28A37514 86AF828C BC698DD5 F18A3B66 9D8D895A 5562E08D 383F790A
A5C7F6F6 915CB558 042E5B99 71F7169D B3AFA699 2B020301 0001A357 3055300B

9-295
0603551D 0F040403 0202A430 13060355 1D25040C 300A0608 2B060105 05070301

301D0603 551D0E04 16041475 71EC5D35 1A431511 7E8C8462 6E65E570 7C551930
12060355 1D130101 FF040830 060101FF 02010030 0D06092A 864886F7 0D01010B
05000382 0101008F 0D3E9F3E 3574100D 97AD876D B4015C21 300A1BD0 59D5C9BF
41A8448D 597CD278 718A6431 BA94C042 7EC64BA0 71F04501 C33C1664 16484373
F3C226A7 256363A9 8BE97291 6B25B8B4 E3DB84C3 3DDB63E7 A9D8D577 6B8F37B3
7CFCE019 D6F09573 946191F7 C4028465 B072DF74 9D6DED45 CA9E6A3B 1401D1A3
5449EDCE 9FA593E3 2FD71031 C7C7EB9C 045DAAFE C67603BF DAB40EE0 352C009F
EAAA6816 A11F6D8B 7C406211 1045A0C6 488B34E1 AF968FAF 3705A364 1EE21A1D
B7080EDC 40D4AA15 E110C5F1 D8A57561 DB2B09F1 0779B855 3998CE22 C471B5CB
09605E24 99855176 2D1CA40E BEBC2F23 7434CA2B 8D1C5EFB 822147CC 81F98825
47A1A14F DC5480
quit
crypto pki certificate chain Cisco_Root_CA_2048
certificate ca 5FF87B282B54DC8D42A315B568C9ADFF
30820343 3082022B A0030201 0202105F F87B282B 54DC8D42 A315B568 C9ADFF30
0D06092A 864886F7 0D010105 05003035 31163014 06035504 0A130D43 6973636F
20537973 74656D73 311B3019 06035504 03131243 6973636F 20526F6F 74204341
20323034 38301E17 0D303430 35313432 30313731 325A170D 32393035 31343230
32353432 5A303531 16301406 0355040A 130D4369 73636F20 53797374 656D7331
1B301906 03550403 13124369 73636F20 526F6F74 20434120 32303438 30820120
300D0609 2A864886 F70D0101 01050003 82010D00 30820108 02820101 00B09AB9
ABA7AF0A 77A7E271 B6B46662 94788847 C6625584 4032BFC0 AB2EA51C 71D6BC6E
7BA8AABA 6ED21588 48459DA2 FC83D0CC B98CE026 68704A78 DF21179E F46105C9
15C8CF16 DA356189 9443A884 A8319878 9BB94E6F 2C53126C CD1DAD2B 24BB31C4
2BFF8344 6FB63D24 7709EABF 2AA81F6A 56F6200F 11549781 75A725CE 596A8265
EFB7EAE7 E28D758B 6EF2DD4F A65E629C CF100A64 D04E6DCE 2BCC5BF5 60A52747
8D69F47F CE1B70DE 701B20D6 6ECDA601 A83C12D2 A93FA06B 5EBB8E20 8B7A91E3
B568EEA0 E7C40174 A8530B2B 4A9A0F65 120E824D 8E63FDEF EB9B1ADB 53A61360
AFC27DD7 C76C1725 D473FB47 64508180 944CE1BF AE4B1CDF 92ED2E05 DF020103
A351304F 300B0603 551D0F04 04030201 86300F06 03551D13 0101FF04 05300301
01FF301D 0603551D 0E041604 1427F3C8 151E6E9A 020916AD 2BA08960 5FDA7B2F
AA301006 092B0601 04018237 15010403 02010030 0D06092A 864886F7 0D010105
05000382 0101009D 9D8484A3 41A97C77 0CB753CA 4E445062 EF547CD3 75171CE8
E0C6484B B6FE4C3A 198156B0 56EE1996 62AA5AA3 64C1F64E 5433C677 FEC51CBA
E55D25CA F5F0939A 83112EE6 CBF87445 FEE705B8 ABE7DFCB 4BE13784 DAB98B97
701EF0E2 8BD7B0D8 0E9DB169 D62A917B A9494F7E E68E95D8 83273CD5 68490ED4
9DF62EEB A7BEEB30 A4AC1F44 FC95AB33 06FB7D60 0ADEB48A 63B09CA9 F2A4B953
0187D068 A4277FAB FFE9FAC9 40388867 B439C684 6F57C953 DBBA8EEE C043B2F8
09836EFF 66CF3EEF 17B35818 2509345E E3CBD614 B6ECF292 6F74E42F 812AD592
91E0E097 3C326805 854BD1F7 57E2521D 931A549F 0570C04A 71601E43 0B601EFE
A3CE8119 E10B35
quit
!
voice service voip
no ip address trusted authenticate
media bulk-stats
media disable-detailed-stats
srtp
no supplementary-service sip handle-replaces
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
!
default mode
no allow-hash-in-dn
security-policy secure
max-dn 50
max-pool 40
!

9-296

id network 10.0.0.1 mask 255.255.0.0
dtmf-relay rtp-nte
codec g711ulaw
!
final 89898
list 1008,2005
timeout 5
pilot 1111
!
voice-card 0/1
no watchdog
!
voice-card 0/2
no watchdog
!
voice-card 0/3
no watchdog
!
voice-card 1/0
no watchdog
!
license udi pid ISR4451-X/K9 sn FOC1743565L
license accept end user agreement
license boot level uck9
license boot level securityk9
no license smart enable
diagnostic bootup level minimal
!
spanning-tree extend system-id
!
redundancy
mode none
!
ip address 10.0.0.1 255.255.0.0
negotiation auto
!
no ip address
negotiation auto
!
ip address 10.0.0.1 255.0.0.0
negotiation auto
!
no ip address
negotiation auto
!
interface Service-Engine0/1/0
shutdown
!
shutdown
!
!
!
interface GigabitEthernet0
vrf forwarding Mgmt-intf
no ip address

9-297
negotiation auto
!
ip forward-protocol nd
ip http server
ip route 0.0.0.0 0.0.0.0 10.0.0.1
!
ip ssh server algorithm encryption aes128-ctr aes192-ctr aes256-ctr
ip ssh client algorithm encryption aes128-ctr aes192-ctr aes256-ctr
!
control-plane
!
!
voice-port 0/1/0
!
voice-port 0/1/1
!
voice-port 0/2/0
!
voice-port 0/2/1
!
voice-port 0/2/2
!
voice-port 0/2/3
!
mgcp behavior rsip-range tgcp-only
mgcp behavior comedia-role none
mgcp behavior comedia-check-media-src disable
mgcp behavior comedia-sdp-force disable
!
!
sip-ua
crypto signaling default trustpoint TRUSTPT-SRST-CA-2
!
!
credentials
trustpoint TRUSTPT-SRST-CA-2
!
!
max-ephones 50
max-dn 50
fac standard
!
!
line con 0
exec-timeout 0 0
length 0
transport input none
stopbits 1
line aux 0
stopbits 1
line vty 0 4
exec-timeout 0 0
password xxxx
no login
length 0
transport preferred none
transport input telnet ssh

9-298
!
end
The following sections provide references related to this feature.

9-299
Related Documents
Related Topic Document Title
Cisco IOS voice configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Voice Command Reference
Cisco Unified Communications Manager • Cisco Unified Communications Manager Documentation Guide
Documentation Guide for Release 8.0(2) for Release 8.0(2)
Cisco Unified SRST configuration • Cisco Unified SRST and SIP SRST Command Reference
Cisco Unified SRST • Cisco Unified SRST 8.0 Supported Firmware, Platforms,
Cisco Unified Communications Operating System • Security
Administration Guide, Release 6.1(1)
Configuring a Secure Survivable Remote Site • Configuring a Secure Survivable Remote Site Telephony
Telephony (SRST) Reference (SRST) Reference
Standards
Standard Title
No new or modified standards are supported by this —
feature, and support for existing standards has not been
modified by this feature.
MIBs
MIB MIBs Link
No new or modified MIBs are supported by this To locate and download MIBs for selected platforms, Cisco IOS
feature, and support for existing MIBs has not been releases, and feature sets, use Cisco MIB Locator found at the
modified by this feature. following URL:
http://www.cisco.com/go/mibs
RFCs
RFC Title
No new or modified RFCs are supported by this —
feature, and support for existing RFCs has not been
modified by this feature.

9-300
Command Reference
Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies.
To receive security and technical information about
your products, you can subscribe to various services,
such as the Product Alert Tool (accessed from Field
Notices), the Cisco Technical Services Newsletter, and
Really Simple Syndication (RSS) Feeds.
Access to most tools on the Cisco Support website
requires a Cisco.com user ID and password.
Command Reference
The following commands are introduced or modified in the feature or features documented in this
section. For information about these commands, see the Cisco IOS Voice Command Reference at
http://www.cisco.com/en/US/docs/ios/voice/command/reference/vr_book.html. For information about
all Cisco IOS commands, use the Command Lookup Tool at http://tools.cisco.com/Support/CLILookup
or Cisco IOS Master Command List, All Releases at
http://www.cisco.com/en/US/docs/ios/mcl/allreleasemcl/all_book.html.
• security-policy
• show voice register global
• show voice register all

9-301
Feature Information for Secure SCCP and SIP SRST
Feature Information for Secure SCCP and SIP SRST

Table 9-4 lists the release history for this feature.
Not all commands may be available in your Cisco IOS software release. For release information about a
specific command, see the command reference documentation.
Use Cisco Feature Navigator to find information about platform support and software image support.
Cisco Feature Navigator enables you to determine which Cisco IOS and Catalyst OS software images
support a specific software release, feature set, or platform. To access Cisco Feature Navigator, go to
http://www.cisco.com/go/cfn. An account on Cisco.com is not required.
Note Table 9-4 lists only the Cisco IOS software release that introduced support for a given feature in a given
Cisco IOS software release train. Unless noted otherwise, subsequent releases of that Cisco IOS
software release train also support that feature.
Table 9-4 Feature Information for Secure SIP Call Signaling and SRTP Media with Cisco SRST
Feature Name Releases Feature Information

Secure SIP Call Signaling and SRTP Media 15.0(1)XA Adds Session Initiation Protocol/Transport Layer
with Cisco SRST Security/Transmission Control Protocol (SIP/TLS/TCP)
support for secure call signaling and Secure Real-time
Transport Protocol (SRTP) for media encryption to
establish a secure, encrypted connection between Cisco
Unified IP Phones and a failover device using Cisco Unified
Survivable Remote Site Telephony (Cisco SRST). The
following commands were introduced or modified:
security-policy, show voice register global, show voice
register all
Where to Go Next
If you require voicemail, see the voice-mail configuration instructions in the “Integrating Voicemail with
Cisco Unified SRST” section on page 321.

9-302
CHAPTER 10
Configuring SIP Trunking on Unified SRST
This chapter describes how to configure SIP trunking on Cisco Unified Survivable Remote Site
Telephony (Unified SRST).
Contents
This chapter describes the configuration recommendations and details on the various line side and SIP
trunking features on Unified SRST. Also, details are provided on the co-location of Unified Border
Element and Unified SRST.
• Configuration Recommendations for Unified SRST and Unified Border Element Co-location,
page 305
Unified SRST and Unified Border Element Co-location

For Unified SRST Release 12.1 and later releases, you can deploy product instances of Cisco Unified
Border Element and Unified SRST (only for SIP) on the same Cisco 4000 Series Integrated Services
Router. Co-location of Unified SRST and Unified Border element is supported from the release Cisco
IOS XE Fuji 16.7.1. All the Cisco SIP IP Phones are supported for this deployment. The phone support
includes, but is not limited to:
• Cisco IP Phone 7800 Series
• Cisco IP Phone 8800 Series
• Cisco Unified IP Phone 9900 Series
When the Wide Area Network (WAN) is available, the router acts as a pure Cisco Unified Border
Element, and not as a Unified SRST.
During a WAN outage, the phones registered to the Unified Communications Manager fall back on the
Unified SRST. However, phones registered to Unified SRST can place or receive PSTN calls through
SIP trunk.
The Unified SRST and the Unified Border Element feature set is limited to the features mentioned. The
following features are supported on the phone when registered to Unified SRST:
• Incoming or Outgoing Basic Call
• Hold/Resume

10-303
Chapter 10 Configuring SIP Trunking on Unified SRST
• Call Forward
• Call Transfer
• Conference (Built-in Bridge)
• Hunt Groups
• MOH (for SIP lines in SRST mode)
The list of SIP trunk features supported for Unified SRST and Unified Border Element co-location are:
• SIP-UA Registration/Authentication, Registrar, Register/Register Refresh
• SIP-Server, Outbound Proxy
• DNS Service Record
• Bind Global / Dial-peer
• SRTP / TLS, SRTP – RTP Interworking
• Connection Reuse
• IP Trust List
• Voice class tenant
• RTP-NTE DTMF
• P-Called-Party ID, Privacy Header (PAI)
• SIP Normalization
For more information on configuring tenants on SIP trunks, see Cisco Unified Border Element
Configuration Guide. For more information on the recommended configurations for the Unified Border
Element co-location, see Configuration Recommendations for Unified SRST and Unified Border
Element Co-location, page 305.
Figure 10-1 shows a co-located deployment of Unified SRST with Cisco Unified Border Element.
Figure 10-1 Co-located Deployment of Unifed SRST and Cisco Unified Border Elelement

10-304
. Primary PSTN calls use central SIP Trunk

IP PSTN . Local (Secondary) PSTN SIP Trunk for
(Secondary) . Emergency
. Central SIP Trunk Unavailable
. WAN Unavailable / SRST
. Unified CM Unavailable / SRST
xDSL / SIP Trunk / Registered

xDSL / SIP-TRUNK / Registered
xDSL / SIP-TRUNK / Registered
Unified SRST / CUBE Unified SRST / CUBE Unified SRST / CUBE
IP PSTN
(Primary)
. SRST Calls:
. Local outgoing calls
. No local Incoming call
Static SIP Trunk (Centralized)
. Incoming / Outgoing calls through central Enterprise
. SIP Trunk, if Unified CM is unavailable IP WAN
Unified CM Centralized CUBE

Unified CM SIP Trunk / CUBE Dial-Peer Cluster
WAN Link
Datacenter / HQ
393597
xDSL / SIP Trunk Registered
Unified SRST to CUBE Dial-Peer
Configuration Recommendations for Unified SRST and Unified Border

Element Co-location
The dial-peers created after the phones (registered to Unified Communications Manager) fall back on
Unified SRST are dynamic dial-peers. Hence, the configurations under voice service voip and sip-ua
are inherited by these dynamic dial-peers. Move voice service voip and sip-ua configurations under
voice class tenant configuration mode to avoid configuration conflict. The voice class tenant is
included in the SIP trunk dial-peer configuration.
Similarly, the relevant global configurations are grouped under a voice class tenant and can be applied
on the dial-peer toward Unified Communications Manager as well. These configurations grouped under
the voice class tenant are used whenever the Unified Communications Manager is available (WAN is
available). For sample configurations of the co-located deployment of Unified SRST and Unified Border
Element, see Examples, page 308.
The following are the configuration recommendations for the Unified SRST and Unified Border Element
co-location:
• Move SIP trunk specific voice service voip and sip-ua configurations under voice class tenant. This
is to avoid configuration conflict between SIP trunk and line side dial-peer configurations. When
tenant is configured under dial-peer, the configurations are applied in the following order of
preference:
– Dial-peer configuration
– Tenant configuration
– Global configuration

10-305
Note Certain CLI commands which need to be moved under tenant, are moved under dial-peer
configuration mode. This is because these CLIs are not available under voice class tenant. For
example, the CLI command srtp fallback needs to be configured under dial-peer, not voice
class tenant configuration mode.
• Use dial-peer groups feature to group multiple outbound dial-peers into a dial-peer group and
configure this dial-peer group as the destination of an inbound dial-peer (Unified CM trunk). For
more information on dial-peer groups, see Dial Peer Configuration Guide.
• Configure SIP Options Request Keepalives to monitor reachability towards Unified
Communications Manager. For example:
voice class sip-options-keepalive 101
up-interval 30
retry 3
transport tcp
Options keepalive under dialpeer

description **CUCM/PBX**
voice-class sip options-keepalive profile 101
• The relevant CLI commands for configuring dial-peer groups are:

– voice class dpg dial-peer-group-id (Creates a dial-peer group.)
– destination dpg dial-peer-group-id (Specifies the dial-peer group from which the outbound
dial-peer(s) is chosen.)
• Avoid configuring dial-peer groups on the SIP trunk dial-peer pointing to the Service Provider
router.
• Configure the destination pattern (.T) on the dial-peer that points to Unified Communications
Manager.
• It is mandatory to configure voice class tenant on the dial-peers pointing towards the Service
Provider router. A configuration with voice class tenant on the dial-peer pointing towards Unified
Communications Manager is also validated, though it is not mandatory.
• Configure the CLI command destination dpg dial-peer-group-id (destination dpg 101) on the
dial-peer pointing to inbound dial-peer for Unified Communications Manager SIP trunk. This dpg
configuration has dial-peer information pointing to the Service Provider. You can configure
preferences for the dial-peers within the dial-peer group:
voice class dpg 1
dial-peer 2900 preference 2
dial-peer 3900 preference 1
• Do not configure incoming called-number (.T), from the dial-peer towards the Service Provider.
Match the incoming call from SIP trunk using the dial-peer address information ‘From URI’, after
removing incoming called-number (.T).
voice class uri 201 sip
host dns:sip-trunk.sample
Under dial-peer:
incoming uri from 201
• Configure the CLI command transport tcp tls v1.2 under sip-ua configuration mode, not voice
class tenant.

10-306
• Avoid modification of contact header in a Secure SIP to SIP (and vice versa) call flow, as it leads to
call establishment issues. If sip-profiles are used to modify header information from sips: to sip: in
SIP REQUESTS and RESPONSES, there must be rules to include ‘transport=tls’ in the contact
header.
• If dial-peers are using voice class codec, configure the same voice class codec under voice register
pool too.
• Ensure that an srtp voice-class is created using the voice class srtp-crypto crypto-tag command. A
sample configuration is as follows:
voice class srtp-crypto 1
crypto 1 AES_CM_128_HMAC_SHA1_32
• Configure the SIP Registrar under voice service voip sip configuration mode with maximum and
minimum expiry time for an incoming registration using the CLI command registrar server
[expires [max sec] [min sec]].
– registrar server expires max 120 min 60
• Move all the CLI commands related to SIP Bind feature under voice class tenant configuration
mode. For example, it is recommended to have the CLI commands voice-class sip bind control, and
voice-class sip bind media, under voice class tenant configuration mode.
• Exclude SIP ports from NAT services, if NAT is configured on the router. The recommended CLIs
for excluding SIP ports from NAT services are:
– no ip nat service sip udp port 5060
– no ip nat service sip tcp port 5060
• Configure the CLI commands no supplementary-service sip refer, no supplementary-service sip
moved-temporarily, supplementary-service media-renegotiate under voice service voip
configuration mode.
• For the co-located deployment of Unified SRST and Unified Border Element, do not configure the
CLI command no transport udp under sip-ua configuration mode. This is because, phones register
to the Unified SRST device using UDP for signaling transport with the non-secure SIP SRST
configuration.
• Playback of MOH from the flash memory of the router is supported for SIP lines in SRST mode in
a co-located deployment of Unified SRST and Cisco Unified Border Element. Cisco IOS XE Fuji
16.7.1 and later releases support this feature.
• Configure Media Inactivity Timer to enable router to monitor and disconnect calls if no Real-Time
Protocol (RTP) packets are received within a configurable time period. A sample configuration is as
follows:
ip rtcp report interval 9000
gateway
media-inactivity-criteria all
timer receive-rtp 1200
timer receive-rtcp 5
Restrictions
The following restrictions are observed for a co-located deployment of Unified SRST and Unified
Border Element:
• You need to disable the NAT firewall support for SIP trunk side, using the CLI commands no ip nat
service sip udp port 5060 and no ip nat service sip tcp port 5060.

10-307
• All the SIP trunk features are not supported in a Unified SRST and Unified Border Element
co-location deployment. For the list of supported features, see Unified SRST and Unified Border
Element Co-location.
Examples
The following is a sample configuration for a voice class tenant:
voice class tenant 1
registrar ipv4:10.64.86.64:5061:5061 scheme sips expires 240 tcp tls auth-realm
sip-trunk.sample
credentials number +492281844672 username xxxx password xxxx realm sip-trunk.sample
authentication username xxxx password xxxx realm sip-trunk.sample
no remote-party-id
timers expires 900000
timers register 100
sip-server dns:sip-trunk.sample:5061
connection-reuse
asserted-id pai
bind control source-interface GigabitEthernet0/0/1
bind media source-interface GigabitEthernet0/0/1
conn-reuse
sip-profiles 3000
outbound-proxy dns:reg.sip-trunk.sample
privacy-policy passthru
call-route p-called-party-id
midcall-signaling preserve-codec
In the following configuration, the voice class tenant configured in the previous example is part of the
dial-peer on the SIP trunk.
description **SIP-TRUNK.SAMPLE**
session target sip-server
session transport tcp tls
destination e164-pattern-map 201
voice-class codec 1
voice-class sip url sips
voice-class sip asserted-id pai
voice-class sip outbound-proxy dns:reg.sip-trunk.sample
voice-class sip tenant 1
voice-class sip srtp-crypto 1
voice-class sip bind control source-interface GigabitEthernet0/0/1
voice-class sip bind media source-interface GigabitEthernet0/0/1
dtmf-relay rtp-nte
srtp
fax-relay ecm disable
fax rate 14400
ip qos dscp cs6 signaling
clid strip name
no vad

10-308
The following example provides the show running-config command output for the co-located
deployment of Unified SRST and Unified Border Element:

!
! Last configuration change at 17:52:50 IST Tue Jul 4 2017
! NVRAM config last updated at 17:52:54 IST Tue Jul 4 2017
!
version 16.7
service sequence-numbers
platform qfp utilization monitor load 80
no platform punt-keepalive disable-kernel-core
platform shell
platform trace runtime slot F0 bay 0 process forwarding-manager module aom level debug
platform trace runtime slot F0 bay 0 process forwarding-manager module dsp level verbose
platform trace runtime slot F0 bay 0 process forwarding-manager module sbc level debug
platform trace runtime slot R0 bay 0 process forwarding-manager module dsp level verbose
platform trace runtime slot R0 bay 0 process forwarding-manager module om level debug
platform trace runtime slot R0 bay 0 process forwarding-manager module sbc level debug
!
hostname be4k-technium
!
boot-start-marker
boot-end-marker
!
!
vrf definition Mgmt-intf
!
address-family ipv4
exit-address-family
!
address-family ipv6
exit-address-family
!
! card type command needed for slot/bay 0/1
no logging queue-limit
logging buffered 100000000
no logging rate-limit
no logging console
!
no aaa new-model
process cpu statistics limit entry-percentage 10 size 7200
clock timezone IST 5 30
!
!
!
ip host gauss-lnx.cisco.com 10.64.86.64
ip name-server 8.41.20.1
ip dhcp excluded-address 8.39.23.13 8.39.23.50
!
ip dhcp pool phones
network 8.39.0.0 255.255.0.0
default-router 8.39.23.13
domain-name cisco.com
dns-server 8.39.23.13
!
!
!
!

10-309
!
!
!
!
!
!
subscriber templating
!
!
!
!
!
!
!
multilink bundle-name authenticated
!
!
!
!
!
!
trunk group 1
xsvc
!
!
crypto pki trustpoint sipgw1
serial-number
ip-address 8.39.23.13
subject-name CN=sipgw1
revocation-check crl
rsakeypair cisco123
!
!
crypto pki certificate chain sipgw1
certificate 02
30820234 3082019D A0030201 02020102 300D0609 2A864886 F70D0101 05050030
13311130 0F060355 04031308 63617365 72766572 301E170D 31373036 32383134
32393330 5A170D31 38303632 38313432 3933305A 305C310F 300D0603 55040313
06736970 67773131 49301206 03550405 130B4644 4F323031 31413132 33301706
092A8648 86F70D01 0908130A 382E3339 2E32332E 3133301A 06092A86 4886F70D
01090216 0D626534 6B2D7465 63686E69 756D3081 9F300D06 092A8648 86F70D01
01010500 03818D00 30818902 818100B5 3CE45902 52517DBE E735F0B5 9D6A412F
FBF398A8 F306F28F A4C79A41 198A19D7 06025696 F5EC6237 EFCB1BBD C7430263
1D0D3C7E AF06B4B2 0D30547C F049A3CD CC4FCFA1 335DA8C5 602A2D18 F91ECC32
E0A7E279 60945941 DF5B53F9 102B9067 8782C1E0 874D6CBC DB0CDA82 C64B7423
E56C5C33 2E13C729 9AB7FEEA 068E7102 03010001 A34F304D 300B0603 551D0F04
04030205 A0301F06 03551D23 04183016 8014265B 6595680C E517CC42 F54AE9EC
1F328FBE BF33301D 0603551D 0E041604 14BA096E DE4E2289 12E8F4D8 95E06E4A
F93876E7 96300D06 092A8648 86F70D01 01050500 03818100 9B172FF6 291C193A
E505ABE9 45AC3202 621BBE2B 6BA45F19 AE0DA7A0 EF5FBC19 5197094E 7A50BCF3
CC49656E A0D991AC FED14749 EAB50892 0239E39C 345ED555 7CD74760 66B0DF49
7E26B654 B8F9E1B1 72FD4039 8A13C9AC EBE75F21 B457D8E3 24BA70E3 F1B3A0C9
5C3153FA B3C744B7 D81F706F B836617F 9E95AD51 813F20AD
quit
certificate ca 01
308201FF 30820168 A0030201 02020101 300D0609 2A864886 F70D0101 04050030
13311130 0F060355 04031308 63617365 72766572 301E170D 31373036 32383134
32383131 5A170D32 30303632 37313432 3831315A 30133111 300F0603 55040313
08636173 65727665 7230819F 300D0609 2A864886 F70D0101 01050003 818D0030
81890281 8100A3AC A4003239 62667AB4 6E8ACE2B 90672DD8 1E2A2952 AFC8A1F6
D56173C9 269F9176 747E93D1 6F699B6F 0C2E600D 8C864F27 4379ED8A E88187F7
17A77C63 B87B7EF6 1556D949 43C743F6 01D9941D 946FCEC8 880B342C 97CC9CEA
9F015EAC A667F30B 505281AA 29EB10A3 F1C75A99 2A224653 F3B985DD F17BC8DD

10-310
40C8C609 62C90203 010001A3 63306130 0F060355 1D130101 FF040530 030101FF

300E0603 551D0F01 01FF0404 03020186 301F0603 551D2304 18301680 14265B65
95680CE5 17CC42F5 4AE9EC1F 328FBEBF 33301D06 03551D0E 04160414 265B6595
680CE517 CC42F54A E9EC1F32 8FBEBF33 300D0609 2A864886 F70D0101 04050003
81810077 C36A6C9A B7C18856 EBDA4504 C38565F0 CF6385EE 29AFC38B 8B90C741
B20C8C36 E979FD72 7B849B34 0BBE3EFA 191E1776 C28FDCF8 5D5F7CFF 170CF615
B4105ABD CD6E0318 4B576FFD 44D115FF 2817E279 78B2794E 577F694F DD129820
B500BB08 E57BFAA9 87835645 4EA53352 B80B51AD 2CC0633A AB9974EB E523A944 0EC230
quit
!
!
!
!
voice service voip
ip address trusted list
ipv4 8.55.0.0 255.255.0.0
ipv4 10.64.0.0 255.255.0.0
address-hiding
mode border-element license capacity 50
media statistics
media bulk-stats
media disable-detailed-stats
no supplementary-service sip moved-temporarily
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
!
!
host ipv4:10.64.86.136
!
host dns:sip-trunk.sample
!
host ipv4:10.64.86.138
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g722-64
codec preference 3 g711ulaw
!
!
voice class sip-profiles 3000
rule 1 request REGISTER sip-header SIP-Req-URI modify "sips:(.*)" "sip:\1"
rule 2 request REGISTER sip-header To modify "<sips:(.*)" "<sip:\1"
rule 3 request REGISTER sip-header From modify "<sips:(.*)" "<sip:\1"
rule 4 request REGISTER sip-header Contact modify "<.*:.*@(.*)>"
"<sip:\1;transport=tls;bnc>"
rule 6 request REGISTER sip-header Proxy-Require add "Proxy-Require: gin"
rule 7 request REGISTER sip-header Require add "Require: gin"
!
rule 1 request ANY sip-header P-Asserted-Identity modify "<sips:(.*)>"
"<sip:[email protected]>"
rule 2 request ANY sip-header SIP-Req-URI modify "sips:(.*)" "sip:\1"
rule 3 request ANY sip-header To modify "<sips:(.*)" "<sip:\1"
rule 4 request ANY sip-header From modify "<sips:(.*)" "<sip:\1"
rule 5 request ANY sip-header Contact modify "<sips:(.*)>" "<sip:\1;transport=tls>"
rule 6 response ANY sip-header To modify "<sips:(.*)" "<sip:\1"
rule 7 response ANY sip-header From modify "<sips:(.*)" "<sip:\1"
rule 8 response ANY sip-header Contact modify "<sips:(.*)>" "<sip:\1;transport=tls>"

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rule 9 request ANY sip-header Min-SE remove

rule 10 request ANY sip-header Diversion remove
rule 11 request ANY sdp-header Connection-Info remove
rule 12 response ANY sdp-header Connection-Info remove
rule 13 request INVITE sip-header Allow-Header modify "INFO," ""
!
rule 1 request INVITE sip-header Supported modify "100rel," ""
!
rule 1 request INVITE sip-header Privacy add "Privacy:id"
rule 2 request INVITE sip-header P-Called-Party-ID add "P-Called-Party-ID:
sip:[email protected]"
!
!
voice class sip-copylist 201
sip-header FROM
!
voice class e164-pattern-map 101
e164 +492284229322T
!
!
e164 11[02]
e164 11[68]T
e164 11[025]
e164 +T
e164 0T
e164 2...
!
!
e164 3...
!
!
voice class dpg 201
!
voice class dpg 101
dial-peer 201
!
voice class dpg 301
dial-peer 301
!
voice class server-group 1
ipv4 10.64.86.136
description **CUCM Server Group**
!
voice class sip-options-keepalive 101
up-interval 30
retry 3
transport tcp
sip-profiles 3000
!
registrar dns:sip-trunk.sample:5061 scheme sips expires 240 tcp tls auth-realm
sip-trunk.sample
credentials number +492281844672 username xxxx password 7 060506324F41 realm
sip-trunk.sample
authentication username xxxx password 7 121A0C041104 realm sip-trunk.sample
no remote-party-id
timers register 100
timers buffer-invite 1000
timers dns registrar-cache ttl

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connection-reuse
asserted-id pai
no pass-thru content custom-sdp
conn-reuse
sip-profiles 3000
!
registrar dns:sip-trunk.sample:5060 expires 240 tcp auth-realm sip-trunk.sample
credentials number +492281844673 username xxxx password 7 030752180500 realm
sip-trunk.sample
authentication username xxxx password 7 121A0C041104 realm sip-trunk.sample
no remote-party-id
timers register 100
connection-reuse
asserted-id pai
conn-reuse
sip-profiles 3000
!
registrar dns:sipp.sample:6600 expires 240 auth-realm sip-trunk.sample
credentials number +492281844672 username xxxx password 7 121A0C041104 realm
sip-trunk.sample
authentication username xxxx password 7 05080F1C2243 realm sip-trunk.sample
no remote-party-id
timers register 500
sip-server dns:sipp.sample
connection-reuse
asserted-id pai
conn-reuse
sip-profiles 3000
outbound-proxy dns:sipp.sample:6600
!
connection-reuse
asserted-id pai

10-313

!
voice class srtp-crypto 1
!
!
!
default mode
no allow-hash-in-dn
max-dn 40
max-pool 40
!
id network 8.55.0.0 mask 255.255.0.0
dtmf-relay rtp-nte
voice-class codec 1
!
voice hunt-group 1 parallel
list 1001,1002,1003
timeout 15
statistics collect
pilot 1234
!
!
list 1002,1003,1004
timeout 5
statistics collect
pilot 2345
!
!
!
!
!
!
voice-card 0/1
dsp services dspfarm
no watchdog
!
license udi pid ISR4321/K9 sn FDO201115PV
license boot level uck9
license boot level securityk9
no license smart enable
diagnostic bootup level minimal
!
spanning-tree extend system-id
!
!
!
username xxxx privilege 15 password 0 cisco
username xxxx password 0 cisco
!
redundancy
mode none
!
!
!

10-314
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
template 1
!
!
!
!
!
ip address 8.39.23.13 255.255.0.0
ip nat inside
media-type rj45
negotiation auto
!
ip address 10.64.86.64 255.255.0.0
ip nat outside
negotiation auto
!
!
interface GigabitEthernet0
vrf forwarding Mgmt-intf
no ip address
negotiation auto
!
no ip nat service sip tcp port 5060
no ip nat service sip udp port 5060
ip nat pool pool1 8.39.0.0 8.39.255.255 netmask 255.255.0.0
ip nat inside source list 100 interface GigabitEthernet0/0/1 overload
ip forward-protocol nd
ip http server
ip tftp source-interface GigabitEthernet0/0/0
ip tftp blocksize 1520
ip rtcp report interval 9000
ip route 0.0.0.0 0.0.0.0 8.39.0.1
ip route 10.0.0.0 255.0.0.0 10.64.86.1
!
ip ssh server algorithm encryption aes128-ctr aes192-ctr aes256-ctr
ip ssh client algorithm encryption aes128-ctr aes192-ctr aes256-ctr
!
!
ip access-list extended nat-list
access-list 100 permit ip 8.39.23.0 0.0.0.255 any
!
!
tftp-server flash:fbi88xx.BE-01-010.sbn
tftp-server flash:kern88xx.12-0-1MN-113.sbn

10-315
tftp-server flash:rootfs88xx.12-0-1MN-113.sbn
tftp-server flash:sb288xx.BE-01-020.sbn
tftp-server flash:sip88xx.12-0-1MN-113.loads
tftp-server flash:vc488xx.12-0-1MN-113.sbn
!
!
ipv6 access-list preauth_v6
permit udp any any eq domain
permit tcp any any eq domain
permit icmp any any nd-ns
permit icmp any any nd-na
permit icmp any any router-solicitation
permit icmp any any router-advertisement
permit icmp any any redirect
permit udp any eq 547 any eq 546
permit udp any eq 546 any eq 547
deny ipv6 any any
!
control-plane
!
!
voip trunk group 1
xsvc
!
uc wsapi
message-exchange max-failures 99
response-timeout 2
source-address 8.39.23.13
probing interval keepalive 60
probing max-failures 2
provider xcc
remote-url http://8.39.23.13:8090/xcc
!
!
provider xsvc
!
!
!
mgcp behavior rsip-range tgcp-only
mgcp behavior comedia-role none
mgcp behavior comedia-check-media-src disable
mgcp behavior comedia-sdp-force disable
!
!
!
!
!
session transport tcp tls
voice-class codec 1
voice-class sip url sips
voice-class sip profiles 201
dtmf-relay rtp-nte
srtp
fax rate 14400

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clid strip name

no vad
!
session transport tcp
voice-class codec 1
voice-class sip url sip
dtmf-relay rtp-nte
srtp fallback
fax rate 14400
clid strip name
no vad
!
destination-pattern 4...
session transport udp
voice-class codec 1
voice-class sip url sip
dtmf-relay rtp-nte
fax rate 14400
clid strip name
no vad
!
description **CUCM/PBX**
destination-pattern .T
session transport tcp
session server-group 1
destination dpg 101
incoming uri via 101
voice-class codec 1
no voice-class sip outbound-proxy
voice-class sip srtp negotiate cisco
voice-class sip profiles 102 inbound
voice-class sip options-keepalive profile 101
dtmf-relay rtp-nte
srtp fallback
fax rate 14400
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711alaw
no vad
!
!
presence
!
gateway

10-317
media-inactivity-criteria all
timer receive-rtcp 5
timer receive-rtp 180
!
sip-ua
transport tcp tls v1.2
crypto signaling default trustpoint sipgw1
!
alias exec cl clear logg
alias exec rtp show voip rtp connections
alias exec pool show voice register pool all brief
!
line con 0
exec-timeout 0 0
password cisco
width 0
transport input none
stopbits 1
line aux 0
stopbits 1
line vty 0 4
exec-timeout 0 0
password cisco
login local
length 0
transport input all
!
!
!
!
!
!
end

10-318
Feature Information for Configuring SIP Trunking on Unified SRST
Feature Information for Configuring SIP Trunking on

Unified SRST
Table 10-1 lists the release history for this feature.
Not all commands may be available in your Cisco IOS software release. For release information about a
specific command, see the command reference documentation.
Cisco Feature Navigator enables you to determine which Cisco IOS and Catalyst OS software images
support a specific software release, feature set, or platform. To access Cisco Feature Navigator, go to
Note Table lists only the Cisco IOS software release that introduced support for a given feature in a given
Cisco IOS software release train. Unless noted otherwise, subsequent releases of that Cisco IOS
software release train also support that feature.
Table 10-1 Feature Information for Configuring SIP Trunking on Unified SRST

Unified SRST and Unified Border Element Cisco IOS Added Support for co-location of Unified SRST and
Co-location XE Fuji Unified Border Element on Cisco 4000 Series Integrated
16.7.1 Services Router.

10-319
Feature Information for Configuring SIP Trunking on Unified SRST

10-320
CHAPTER 11
Integrating Voicemail with Cisco Unified
SRST
This chapter describes how to make your existing voicemail system run on phones connected to a
Cisco Unified SRST router during Cisco Unified Communications Manager fallback.
Cisco Unified SRST also supports incoming and outgoing Session Initiation Protocol (SIP) calls to and
from Cisco Unified IP phones and router voice gateway voice ports. SIP may be used in situations where
the Cisco Unified SRST Router is separate from the PSTN gateway and the SRST and PSTN gateways
are linked together using SIP (instead of H.323).
For more information about SIP, see Cisco IOS SIP Configuration Guide.
Contents
• Information About Integrating Voicemail with Cisco Unified SRST, page 321
• How to Integrate Voicemail with Cisco Unified SCCP and SIP SRST, page 323
• Configuring Message Waiting Indication (SIP Phones in SRST Mode), page 335
• How to Configure DTMF Relay for SIP Applications and Voicemail, page 340
Information About Integrating Voicemail with

Cisco Unified SRST
Cisco Unified SRST can send and receive voicemail messages from Cisco Unity and other voicemail
systems during Cisco Unified CM fallback. When the WAN is down, a voicemail system with BRI or
PRI access to the Cisco Unified SRST system uses ISDN signaling (see Figure 11-1). Systems with
Foreign Exchange Office (FXO) or Foreign Exchange Station (FXS) access connect to a PSTN and use
in-band dual tone multifrequency (DTMF) signaling (see Figure 11-2).
From Unified SRST Release 12.0 onwards, Unified SRST supports voicemail on IPv6 protocols for SIP
IP phones.

11-321
Chapter 11 Integrating Voicemail with Cisco Unified SRST
Information About Integrating Voicemail with Cisco Unified SRST
Figure 11-1 Cisco Unified Communications Manager Fallback with BRI or PRI
Cisco Unified
Communications
Cisco Unified Manager
SRST gateway gateway
IP
BRI/PRI
IP
WAN failure Cisco Unified
Communications
Manager
IP
146615
Voicemail Server
WAN
Figure 11-2 Cisco Unified Communications Manager Fallback with PSTN
Cisco Unified
Communications
Manager gateway
IP
FXS FXO
PSTN
IP
WAN failure Cisco Unified
Communications
Manager
IP
WAN
155102
Voicemail Server
Both configurations allow phone message buttons to remain active and calls to busy or unanswered
numbers to be forwarded to the dialed numbers’ mailboxes.
Calls that reach a busy signal, calls that are unanswered, and calls made by pressing the message button
are forwarded to the voicemail system. To make this happen, you must configure access from the dial
peers to the voicemail system and establish routing to the voicemail system for busy and unanswered
calls and for message buttons.
If the voicemail system is accessed over FXO or FXS, you must configure instructions (DTMF patterns)
for the voicemail system so that it can access the correct voicemail system mailbox. If your voicemail
system is accessed over BRI or PRI, no instructions are necessary because the voicemail system can
log in to the calling phone’s mailbox directly.

11-322
How to Integrate Voicemail with Cisco Unified SCCP and SIP SRST
How to Integrate Voicemail with Cisco Unified SCCP and

SIP SRST
• Configuring Direct Access to Voicemail, page 323 (Required)
• Configuring Message Buttons, page 326 (Required)
• Redirecting to Cisco Unified Communications Manager Gateway, page 329 (Required for BRI or
PRI))
• Configuring Call Forwarding to Voicemail, page 329 (Required FXO or FXS)
• Configuring Message Waiting Indication (Cisco Unified SRST Routers), page 333 (Optional)
Note Support for SIP SRST is added from IOS release 15.1(4)M3 and 15.2(1)T2.
Configuring Direct Access to Voicemail

You can configure direct access to voicemail system using BRI/PRI or FXO/FXS. To access voicemail
messages with BRI/PRI or FXO/FXS access, you must have POTS dial peers configured with a
destination pattern that matches the voicemail system’s number. Also, you must associate the dial peer
with the port to which the voicemail system is accessed.
Both sets of configurations are done in dial-peer configuration mode. The summary and detailed steps
below include only the basic commands necessary to perform this task. You may require additional
commands for your particular dial-peer configuration.
SUMMARY STEPS
1. dial-peer voice tag {pots | voatm | vofr | voip}

2. destination-pattern [+] string [T]
3. port {slot-number/subunit-number/port | slot/port:ds0-group-no}
4. forward-digits {num-digit | all | extra}
5. exit

11-323
DETAILED STEPS

Step 1 dial-peer voice tag {pots | voatm | vofr | (FXO or FXS and BRI or PRI) Defines a particular dial peer,
voip} specifies the method of voice encapsulation, and enters
dial-peer configuration mode. The dial-peer command
Example: provides different syntax for individual routers. This example is
Router(config)# dial-peer voice 1002 pots syntax for Cisco 3600 series routers.
• tag: Digits that define a particular dial peer. Range is from
1 to 2147483647.
• pots: Indicates that this is a POTS dial peer that uses VoIP
encapsulation on the IP backbone.
• voatm: Specifies that this is a VoATM dial peer that uses
real-time AAL5 voice encapsulation on the ATM backbone
network.
• vofr: Specifies that this is a VoFR dial peer that uses
FRF.11 encapsulation on the Frame Relay backbone
network.
• voip: Indicates that this is a VoIP dial peer that uses voice
encapsulation on the POTS network.
Step 2 destination-pattern [+] string [T] (FXO or FXS and BRI or PRI) Specifies either the prefix or the
full E.164 telephone number (depending on your dial plan) to
be used for a dial peer.
Example:
Router(config-dial-peer)# destination-pattern • +: (Optional) Character that indicates an E.164 standard
1100T number.
• string: See Table 11-1.
• T: (Optional) Control character that indicates that the
destination-pattern value is a variable-length dial string.
Step 3 port {slot-number/subunit-number/port | (FXO or FXS and BRI or PRI) Associates a dial peer with a
slot/port:ds0-group-no} specific voice port on Cisco routers.
• slot-number: Number of the slot in the router in which the
Example: voice interface card (VIC) is installed. Valid entries are
Router(config-dial-peer)# port 1/1/1 from 0 to 3, depending on the slot in which it is installed.
• subunit-number: Subunit on the VIC in which the voice
port is located. Valid entries are 0 or 1.
• port: Voice port number. Valid entries are 0 and 1.
• ds0-group-no: Specifies the DS0 group number. Each
defined DS0 group number is represented on a separate
voice port. This allows you to define individual DS0s on
the digital T1/E1 card.

11-324

Step 4 forward-digits {num-digit | all | extra} (Optional for FXO or FXS) Specifies which digits to forward
for voice calls.
Example: • num-digit: The number of digits to be forwarded. If the
Router(config-dial-peer)# forward-digits all number of digits is greater than the length of a destination
phone number, the length of the destination number is used.
Range is 0 to 32. Setting the value to 0 is equivalent to
entering the no forward-digits command.
• all: Forwards all digits. If all is entered, the full length of
the destination pattern is used.
• extra: If the length of the dialed digit string is greater than
the length of the dial-peer destination pattern, the extra
right-justified digits are forwarded. However, if the
dial-peer destination pattern is variable length and ends
with the character “T” (for example: T, 123T, 123...T),
extra digits are not forwarded.
Step 5 exit (FXO or FXS and BRI or PRI) Exits dial-peer configuration
mode.
Example:
Router(config-dial-peer)# exit
Table 11-1 Valid Entries for the String Argument in the destination-pattern command
Entry Description
Digits 0 to 9 —
Letters A through D —
Asterisk (*) and pound sign (#) These appear on standard touch-tone dial pads.
Comma (,) Inserts a pause between digits.
Period (.) Matches any entered digit (this character is used as a wildcard).
Percent sign (%) Indicates that the preceding digit occurred zero or more times; similar to the wildcard
usage.
Plus sign (+) Indicates that the preceding digit occurred one or more times.
Note The plus sign used as part of a digit string is different from the plus sign that
can be used in front of a digit string to indicate that the string is an E.164
standard number.
Circumflex (^) Indicates a match to the beginning of the string.
Parentheses ( ( ) ), which indicate a pattern and are the same as the regular expression
rule.
Dollar sign ($) Matches the null string at the end of the input string.
Backslash symbol (\) Is followed by a single character and matches that character. Can be used with a single
character with no other significance (matching that character).
Question mark (?) Indicates that the preceding digit occurred zero or one time.
Brackets ( [ ] ) Indicates a range. A range is a sequence of characters enclosed in the brackets; only
numeric characters from 0 to 9 are allowed in the range.

11-325
Examples
The following FXO and FXS example sets up a POTS dial peer named 1102, matches dial-peer 1102 to
voicemail extension 1101, and assigns dial-peer 1102 to voice-port 1/1/1 where the voicemail system is
connected. Other dial peers are configured for direct access to voicemail.
voice-port 1/1/1
timing digit 250
timing inter-digit 250

port 1/1/1
forward-digits all

port 1/1/1
forward-digits all

port 1/1/1
forward-digits all
The following example sets up a POTS dial peer named 1102 to go directly to 1101 through port 2/0:23:
controller T1 2/0
framing esf
clock source line primary
linecode b8zs
cablelength short 133
pri-group timeslots 21-24
no ip address
isdn T309-enable
no cdp enable
voice-port 2/0:23

port 2/0:23
Configuring Message Buttons

To activate the message buttons on Cisco Unified IP phones connected to the Cisco Unified SCCP and
SIP SRST router during Cisco Unified Communications Manager fallback, you must program a
speed-dial number to the voicemail system. The speed-dial number is dialed when message buttons on
phones connected to the Cisco Unified SCCP and SIP SRST router are pressed during Cisco Unified CM
fallback. In addition, call forwarding must be configured so that calls to busy and unanswered numbers
are sent to the voicemail number.
This configuration is required for FXO or FXS and BRI or PRI.

11-326
SUMMARY STEPS
2. voicemail phone-number
3. call-forward busy directory-number
4. call-forward noan directory-number timeout seconds
5. exit
9. exit
DETAILED STEPS

Example:
Step 2 voicemail phone-number Configures the telephone number that is dialed when the
message button on a Cisco Unified SCCP IP Phone is
pressed.
Example:
Router(config-cm-fallback)# voicemail 5550100 • phone-number: Phone number configured as a
speed-dial number for retrieving messages.
Step 3 call-forward busy directory-number Configures call forwarding to another number when the
Cisco SCCP IP phone is busy.
Router(config-cm-fallback)# call-forward busy representing a fully qualified E.164 number. This
2000 number can contain “.” wildcard characters that
number extension.
Step 4 call-forward noan directory-number timeout Configures call forwarding to another number when no
seconds answer is received from the Cisco SCCP IP phone.
Example: representing a fully qualified E.164 number. This
Router(config-cm-fallback)# call-forward noan number can contain “.” wildcard characters that
2000 timeout 10
number extension.

11-327

Example:
Example:
Step 7 call-forward b2bua busy directory-number Configures call forwarding to another number when the
Cisco SIP IP phone is busy.
Router(config-register-pool)# call-forward representing a fully qualified E.164 number. This
b2bua busy 2000 number can contain “.” wildcard characters that
number extension.
Step 8 call-forward b2bua noan directory-number Configures call forwarding to another number when no
timeout seconds answer is received from the Cisco SIP IP phone.
Example: representing a fully qualified E.164 number. This
Router(config-register-pool)# call-forward noan number can contain “.” wildcard characters that
2000 timeout 10
number extension.
Step 9 exit Exits voice register pool configuration mode.
Example:
Router(config-register-pool)# exit
Examples
The following example specifies 1101 as the speed-dial number that is issued when message buttons are
pressed on Cisco Unified IP Phones connected to the Cisco Unified SRST router. All busy and
unanswered calls are configured to be forwarded to the voicemail number (1101).
voicemail 1101

11-328
Redirecting to Cisco Unified Communications Manager Gateway
Note The following task is required for voicemail systems with BRI or PRI access.
In addition to supporting message buttons for retrieving personal messages, Cisco Unified SRST allows
the automatic forwarding of calls to busy and unanswered numbers to voicemail systems. Voicemail
systems with BRI or PRI access can log in to the calling phone’s mailbox directly. For this to happen,
some Cisco Unified CM configuration is recommended. If your voicemail system supports Redirected
Dialed Number Identification Service (RDNIS), RDNIS must be included in the outgoing SETUP
message to Cisco Unified CM to declare the last redirected number and the originally dialed number to
and from configured devices and applications.
Step 1 From any page in Cisco Unified CM, click Device and Gateway.
Step 2 From the Find and List Gateways page, click Find.
Step 3 From the Find and List Gateways page, choose a device name.
Step 4 From the Gateway Configuration page, check Redirecting Number IE Delivery - Outgoing.
Configuring Call Forwarding to Voicemail
Note The following task is required for voicemail systems with FXO or FXS access.
In addition to supporting message buttons for retrieving personal messages, Cisco Unified SRST allows
the automatic forwarding of calls to busy or unanswered numbers to voicemail systems. The forwarded
calls can be routed to almost any location in the voicemail system. Typically, calls are forwarded to a
location in the called number’s mailbox where the caller can leave messages.
Call Routing Instructions Using DTMF Digit Patterns

Cisco Unified SRST call-routing instructions are required so that forwarded calls can be sent to the
correct voicemail boxes. These instructions consist of DTMF digits configured in patterns that match the
dial sequences required by the voicemail system to get to a particular voicemail location. For example,
a voicemail system may be designed so that callers must do the following to leave a message:
1. Dial the central voicemail number (1101) and press #.
2. Dial an extension number (6000) and press #.
3. Dial 2 to select the menu option for leaving messages in the extension number’s mailbox.
For Cisco Unified SRST to forward a call to a busy or unanswered number to extension 6000’s mailbox,
it must be programmed to issue a sequence of 1101#6000#2. As shown in Figure 11-3, this is
accomplished through the voicemail and pattern commands.

11-329
Figure 11-3 How Voicemail Dial Sequence 1101#6000#2 Is Configured in Cisco Unified SRST
voicemail 1101
1101 #6000#2
pattern ext-to-ext busy # cgn #2
pattern ext-to-ext busy # cdn #2
pattern ext-to-ext busy # fdn #2
pattern ext-to-ext no-answer # cgn #2
pattern ext-to-ext no-answer # cdn #2
pattern ext-to-ext no-answer # fdn #2
pattern trunk-to-ext busy # cgn #2
pattern trunk-to-ext busy # cdn #2
pattern trunk-to-ext busy # fdn #2
pattern trunk-to-ext no-answer # cgn #2
88978
pattern trunk-to-ext no-answer # cdn #2
pattern trunk-to-ext no-answer # fdn #2
The # cgn #2, # cdn #2, and # fdn #2 portions of the pattern commands shown in Figure 11-3 are DTMF
digit patterns. These patterns are composed of tags and tokens. Tags are sets of characters representing
DTMF tones. Tokens consist of three command keywords (cgn, cdn, and fdn) that declare the state of
an incoming call transferred to voicemail.
A tag can be up to three character from the DTMF tone set (A to D, 0 to 9, # and *). Voicemail systems
can use limited sets of DTMF tones. For example, Cisco Unity uses all DTMF tones but A to D. Tones
can be defined in multiple ways. For example, when the star (*) is placed in front of a token by itself, it
can mean “dial the following token number,” or, if it is at the end of a token, it can mark the end of a
token number. If the asterisk is between other tag characters, it can mean dial *. The use of tags depends
on how DTMF tones are defined by your voicemail system.
Tokens tell Cisco Unified SRST what telephone number in the call forwarding chain to use in the pattern.
As shown in Figure 11-4, there are three types of tokens that correspond to three possible call states
during voicemail forwarding.
Figure 11-4 How Numbers Are Extracted from Tokens
(cgn=calling number) (fdn=forwarding number) (cdn=called number)
1000 calls 2000 1000 is forwarded Cisco Unified

IP IP Communications
ext. 1000 ext. 2000 Manager
ext. 3000
pattern ext-to-ext busy # cdn # 2 = pattern ext-to-ext busy # 3000 # 2

88979
pattern ext-to-ext busy # fdn # 2 = pattern ext-to-ext busy # 2000 # 2

pattern ext-to-ext busy # cgn # 2 = pattern ext-to-ext busy # 1000 # 2
Sets of tags and tokens or patterns activate a voicemail system when one of the following occurs:
• A user presses the message button on a phone (pattern direct command).
• An internal extension attempts to connect to a busy extension and the call is forwarded to voicemail
(pattern ext-to-ext busy command).
• An internal extension fails to connect to an extension and the call is forwarded to voicemail (pattern
ext-to-ext no-answer command).
• An external trunk call reaches a busy extension and the call is forwarded to voicemail (pattern
trunk-to-ext busy command).

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• An external trunk call reaches an unanswered extension and the call is forwarded to voicemail
(pattern trunk-to-ext no-answer command).
Prerequisites
• FXO hairpin-forwarded calls to voicemail systems must have disconnect supervision from the
central office. For further information, see the FXO Answer and Disconnect Supervision document.
• To configure patterns that your voicemail system will interpret correctly, you must know how the
system routes voicemail calls and interprets DTMF tones (see the “Call Routing Instructions Using
DTMF Digit Patterns” section on page 329).
You can find information about how Cisco Unity handles voicemail calls in the How to Transfer a
Caller Directly into a Cisco Unity Mailbox document. Additional call-handling information can be
found in the “Subscriber and Operator Orientation” chapters of any Cisco Unity system
administration guide.
For other voicemail systems, see the analog voicemail integration configuration guide or
information about the system’s call handling.
SUMMARY STEPS
1. vm-integration
2. pattern direct tag1 {CGN | CDN | FDN} [tag2 {CGN | CDN | FDN}]
[tag3 {CGN | CDN | FDN}] [last-tag]
3. pattern ext-to-ext busy tag1 {CGN | CDN | FDN} [tag2 {CGN | CDN | FDN}]
4. pattern ext-to-ext no-answer tag1 {CGN | CDN | FDN} [tag2 {CGN | CDN | FDN}]
5. pattern trunk-to-ext busy tag1 {CGN | CDN | FDN} [tag2 {CGN | CDN | FDN}]
6. pattern trunk-to-ext no-answer tag1 {CGN | CDN | FDN} [tag2 {CGN | CDN | FDN}]

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DETAILED STEPS

Step 1 vm-integration Enters voicemail integration mode and enables voicemail
integration with DTMF and analog voicemail systems.
Example:
Router(config)# vm-integration
Step 2 pattern direct tag1 {CGN | CDN | FDN} [tag2 Configures the DTMF digit pattern forwarding necessary to
{CGN | CDN | FDN}] [tag3 {CGN | CDN | FDN}] activate the voicemail system when the user presses the
[last-tag]
messages button on the phone.
• tag1: Alphanumeric string fewer than four DTMF digits
Example: in length. The alphanumeric string consists of a
Router(config-vm-int)# pattern direct 2 CGN *
combination of four letters (A, B, C, and D), two
symbols (* and #), and ten digits (0 to 9). The tag
numbers match the numbers defined in the voicemail
system’s integration file, immediately preceding either
the number of the calling party, the number of the called
party, or a forwarding number.
• tag2 and tag3: (Optional) See tag1.
• last-tag: See tag1. This tag indicates the end of the
pattern.
• CGN: Calling number (CGN) information is sent to the
voicemail system.
• CDN: Called number (CDN) information is sent to the
voicemail system.
• FDN: Forwarding number (FDN) information is sent to
the voicemail system.
Step 3 pattern ext-to-ext busy tag1 {CGN | CDN | Configures the DTMF digit pattern forwarding necessary to
FDN} [tag2 {CGN | CDN | FDN}] [tag3 {CGN | activate the voicemail system once an internal extension
CDN | FDN}] [last-tag]
attempts to connect to a busy extension and the call is
forwarded to voicemail. For argument and keyword
Example: information, see Step 2.
Router(config-vm-int)# pattern ext-to-ext
busy 7 FDN * CGN *
Step 4 pattern ext-to-ext no-answer tag1 {CGN | CDN Configures the DTMF digit pattern forwarding necessary to
| FDN} [tag2 {CGN | CDN | FDN}] [tag3 {CGN | activate the voicemail system once an internal extension fails
to connect to an extension and the call is forwarded to
voicemail. For argument and keyword information, see
Example: Step 2.
Router(config-vm-int)# pattern ext-to-ext
no-answer 5 FDN * CGN *

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Step 5 pattern trunk-to-ext busy tag1 {CGN | CDN | Configures the DTMF digit pattern forwarding necessary to
FDN} [tag2 {CGN | CDN | FDN}] [tag3 {CGN | activate the voicemail system once an external trunk call
reaches a busy extension and the call is forwarded to
Example: Step 2.
Router(config-vm-int)# pattern trunk-to-ext
busy 6 FDN * CGN *
Step 6 pattern trunk-to-ext no-answer tag1 {CGN | Configures the DTMF digit pattern forwarding necessary to
CDN | FDN} [tag2 {CGN | CDN | FDN}] [tag3 activate the voicemail system when an external trunk call
{CGN | CDN | FDN}] [last-tag]
reaches an unanswered extension and the call is forwarded to
Example: Step 2.
Router(config-vm-int)# pattern trunk-to-ext
no-answer 4 FDN * CGN *
Examples
For the following configuration, if the voicemail number is 1101, and 3001 is a phone with a message
button, 1101*3001 would be dialed automatically when the 3001 message button is pressed. Under these
circumstances, 3001 is considered to be a calling number or inbound call number.
vm-integration
pattern direct * CGN
For the following configuration, if 3001 calls 3006 and 3006 does not answer, the Unified SRST router
will forward 3001 to the voicemail system (1101) and send to the voicemail system the DTMF pattern #
3006 #2. This pattern is intended to select voicemail box number 3006 (3006’s voice mailbox). For this
pattern to be sent, 3001 must be a forwarding number.
vm-integration
pattern ext-to-ext no-answer # FDN #2
For the following configuration, if 3006 is busy and 3001 calls 3006, the Unified SRST router will
forward 3001 to the voicemail system (1101) and send to the voicemail system the DTMF pattern # 3006
#2. This pattern is intended to select voice mailbox number 3006 (3006’s voice mailbox). For this pattern
to be sent, 3001 must be a forwarding number.
vm-integration
pattern ext-to-ext busy # FDN #2
Configuring Message Waiting Indication (Cisco Unified SRST Routers)

The Message Waiting Indication (MWI) relay mechanism is initiated after someone leaves a voicemail
message on the remote voicemail message system. MWI relay is required when one Cisco Unity
Voicemail system is shared by multiple Cisco Unified SRST routers. Unified SRST routers use the SIP
Subscribe and Notify methods for MWI. See Configuring Cisco IOS SIP Configuration Guide for more
information on SIP MWI and the Subscribe and Notify methods. The Unified SRST router that is the SIP
MWI relay server acts as the SIP notifier. The other remote routers act as the SIP subscribers.

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Restriction
• MWI is not supported during a fallback to Unified SRST. The MWI (the phone LED indication) will
not correctly reflect when new messages arrive or when all messages have been listened to. We
recommend resynchronizing MWIs after the WAN link is available, and connection with Unified
Communications Manager is reestablished. The MWI behavior is consistent across voicemail
support for IPv4 as well as IPv6 on Unified SRST.
SUMMARY STEPS
2. mwi relay
3. mwi reg-e164
4. exit
5. sip-ua
6. mwi-server {ipv4:destination-address | dns:host-name} [expires seconds] [port port]
[transport {tcp | udp}] [unsolicited]
7. exit
DETAILED STEPS
Command Purpose
Example:
Step 2 mwi relay Enables the Unified SRST router to relay MWI
information to remote Cisco IP phones.
Example:
Router(config-cm-fallback)# mwi relay
Step 3 mwi reg-e164 Registers E.164 numbers rather than extension
numbers with a SIP proxy or registrar.
Example:
Router(config-cm-fallback)# mwi reg-e164
Example:
Example:

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Command Purpose
Step 6 mwi-server {ipv4:destination-address | Configures voicemail server settings on a voice
dns:host-name} [expires seconds] [port port] gateway or user agent. The IP address and port for the
[transport {tcp | udp}] [unsolicited]
SIP-based MWI server should be in the same LAN as
the voicemail server. The MWI server is a
Example: Cisco Unified SRST router. Keywords and arguments
Router(config-sip-ua)# mwi-server ipv4:10.0.2.254 are as follows:
• ipv4:destination-address: IP address of the
voicemail server.
• dns:host-name: The argument should contain the
complete hostname to be associated with the
target address; for example, dns:test.cisco.com.
• expires seconds: Subscription expiration time, in
seconds. Range is from 1 to 999999. Default
is 3600.
• port port: Port number on the voicemail server.
Default is 5060.
• transport: Transport protocol to the voicemail
server. Valid values are tcp and udp. Default is
UDP.
• unsolicited: Requires the voicemail server to
send a SIP notification message to the voice
gateway or UA if the mailbox status changes.
Removes the requirement that the voice gateway
subscribe for MWI service.
Step 7 exit Exits SIP user-agent configuration mode.
Example:
Configuring Message Waiting Indication (SIP Phones in SRST Mode)

On SIP phones operating in the SIP SRST mode, you can use the mwi unsolicited command to configure
a message-waiting notification when a message is sent by the Cisco Unity Express (CUE). The SIP
phone then displays the notification when indicated by the voice messaging system. To configure
message-waiting notification, perform the following steps.
SUMMARY STEPS
1. enable
3. sip-ua
4. mwi-server {ipv4:destination-address | dns:host-name} [unsolicited]
5. exit

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7. mwi unsolicited
8. end
DETAILED STEPS
Command Purpose
Enter your password if prompted.
Example:
Router> enable
Example:
Step 3 sip-ua Enters Session Initiation Protocol (SIP) user agent
(ua) configuration mode for configuring the user
agent.
Example:
Step 4 mwi-server {ipv4:destination-address | Configures voicemail server settings on a voice
dns:host-name} [unsolicited] gateway or user agent. Keywords and arguments are
as follows:
Example: • ipv4:destination-address: IP address of the
Router(config-sip-ua)# mwi-server ipv4:10.0.2.254 voicemail server.
unsolicited
• dns:host-name: The argument should contain the
Or complete hostname to be associated with the
Router(config-sip-ua)# mwi-server target address; for example, dns:test.cisco.com.
dns:server.yourcompany.com unsolicited • unsolicited: Requires the voicemail server to
send a SIP notification message to the voice
gateway or UA if the mailbox status changes.
Removes the requirement that the voice gateway
subscribe for MWI service.
Example:
parameters for all supported SIP phones in SIP SRST
mode.
Example:

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Configuration Examples for Unified SRST
Command Purpose
Step 7 mwi unsolicited Enables all SIP phones to receive MWI notification.
Example:
Router(config-register-global)# mwi unsolicited
Example:

This section provides the following configuration examples:
• Configuring Local Voicemail System (FXO and FXS): Example, page 337
• Configuring Central Location Voicemail System (FXO and FXS): Example, page 338
• Configuring Voicemail Access over FXO and FXS: Example, page 338
• Configuring Voicemail Access over BRI and PRI: Example, page 339
• Message Waiting Indication for SIP SRST: Example, page 340
Configuring Local Voicemail System (FXO and FXS): Example

The “Dial-Peer Configuration for Integration of Voicemail with Cisco Unified SRST” section of the
example below shows a legacy dial-peer configuration for a local voicemail system. The “Cisco Unified
SRST Voicemail Integration Pattern Configuration” section must be compatible with your voicemail
system configuration.
! Dial-Peer Configuration for Integration of voicemail with Cisco Unified SRST
!
port 3/0/0
!
preference 1
port 3/0/1
!
preference 2
port 3/1/0
!
port 3/1/1
!
! Cisco Unified SRST configuration
!
max-ephones 24
max-dn 144

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voicemail 14011
! Cisco Unified SRST voicemail Integration Pattern Configuration

!
vm-integration
pattern direct 2 CGN *
pattern ext-to-ext no-answer 5 FDN * CGN *
pattern ext-to-ext busy 7 FDN * CGN *
pattern trunk-to-ext no-answer 4 FDN * CGN *
pattern trunk-to-ext busy 6 FDN * CGN *
Configuring Central Location Voicemail System (FXO and FXS): Example

The “Dial-Peer Configuration for Integration of voicemail with Cisco Unified SRST in Central
Location” section of the example shows a legacy dial-peer configuration for a central voicemail system.
The “Cisco Unified SRST Voicemail Integration Pattern Configuration” section must be compatible with
your voicemail system configuration.
Note Message waiting indicator (MWI) integration is not supported for PSTN access to voicemail systems at
central locations.
! Dial-Peer Configuration for Integration of voicemail with Cisco Unified SRST in Central
! Location
!
port 3/0/0
!
! Cisco Unified SRST configuration
!
max-ephones 24
max-dn 144
voicemail 14011
!
! Cisco Unified SRST Voicemail Integration Pattern Configuration
!
vm-integration
pattern direct 2 CGN *
pattern ext-to-ext no-answer 5 FDN * CGN *
pattern ext-to-ext busy 7 FDN * CGN *
pattern trunk-to-ext no-answer 4 FDN * CGN *
pattern trunk-to-ext busy 6 FDN * CGN *
Configuring Voicemail Access over FXO and FXS: Example

The following example shows how to configure the Cisco Unified SRST router to forward unanswered
calls to voicemail. In this example, the voicemail number is 1101, the voicemail system is connected to
FXS voice port 1/1/1, and the voice mailbox numbers are 3001, 3002, and 3006.

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voice-port 1/1/1
timing digit 250
timing inter-digit 250

port 1/1/1
max-ephones 24
max-dn 24
transfer-pattern 3...
voicemail 1101
moh minuet.au
vm-integration
pattern direct * CGN
pattern ext-to-ext no-answer # FDN #2
pattern ext-to-ext busy # FDN #2
pattern trunk-to-ext no-answer # FDN #2
pattern trunk-to-ext busy # FDN #2
Configuring Voicemail Access over BRI and PRI: Example

The following example shows how to configure the Cisco Unified SRST router to forward unanswered
calls to voicemail. In this example, the voicemail number is 1101, the voicemail system is connected to
a BRI or PRI voice port, and the voice mailbox numbers are 3001, 3002, and 3006.
controller T1 2/0
framing esf
clock source line primary
linecode b8zs
cablelength short 133
pri-group timeslots 21-24
no ip address
isdn T309-enable
no cdp enable
voice-port 2/0:23

direct-inward-dial
port 2/0:23
max-ephones 24
max-dn 24
transfer-pattern 3...

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How to Configure DTMF Relay for SIP Applications and Voicemail
voicemail 1101
moh minuet.au
Message Waiting Indication for SIP SRST: Example

The following is an example of a NOTIFY message received at SRST indicating that there is a voicemail
for extension 32002:
Received:
NOTIFY sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.4.49.66:5060;branch=z9hG4bK.D6.7wAl9CN6khf305D1MQ~~194
Max-Forwards: 70
To: <sip:[email protected]:5060>
From: <sip:[email protected]:5060>;tag=dsd3d29b2f
Call-ID: f0e7ae97-1227@sip:[email protected]:5060
CSeq: 1 NOTIFY
Content-Length: 112
Contact: <sip:[email protected]:5060>
Content-Type: application/simple-message-summary
Event: message-summary
Messages-Waiting: yes
Message-Account: sip:[email protected]
Voice-Message: 1/0 (1/0)
Fax-Message: 0/0 (0/0)
How to Configure DTMF Relay for SIP Applications and

Voicemail
DTMF relay for SIP applications can be used in two voicemail situations:
• DTMF Relay Using SIP RFC 2833, page 340
• DTMF Relay Using SIP Notify (Nonstandard), page 342
For SIP SRST forwarding call to voicemail configuration, see the “” section on page 175.
Note Voicemail number associate with SIP phone message button in SRST is configured by Cisco
Unified Communications Manager (CUCM), and not configurable by SIP SRST. The
administrator needs to know the voicemail number set by CUCM to configure proper dial peer
to voicemail system in SIP SRST.
DTMF Relay Using SIP RFC 2833

Cisco Unified Skinny Client Control Protocol (SCCP) Phones, such as those used with Cisco Unified
SRST systems, provide only out-of-band DTMF digit indications. To enable SCCP phones to send digit
information to remote SIP-based IVR and voicemail applications, Cisco Unified SRST 3.2 and later
versions provide conversion from the out-of-band SCCP digit indication to the SIP standard for DTMF
relay, which is RFC 2833. You select this method in the SIP VoIP dial peer using the dtmf-relay rtp-nte
command.
The SIP DTMF relay method is needed in the following situations:

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• When SIP is used to connect a Cisco Unified SRST system to a remote SIP-based IVR or voicemail
application, such as Cisco Unity.
• When SIP is used to connect a Cisco Unified SRST system to a remote SIP-PSTN voice gateway
that goes through the PSTN to a voicemail or IVR application.
Note The need to use out-of-band DTMF relay conversion is limited to SCCP phones. SIP phones natively
support in-band DTMF relay as specified in RFC 2833.
To enable SIP DTMF relay using RFC 2833, the commands in this section must be used on both
originating and terminating gateways.
SUMMARY STEPS
1. dial-peer voice tag voip

2. dtmf-relay rtp-nte
3. exit
4. sip-ua
5. notify telephone-event max-duration time
6. exit
DETAILED STEPS

Step 1 dial-peer voice tag voip Enters dial-peer configuration mode.
Example:
Protocol (RTP) with the Named Telephone Event
(NTE) payload type.
Example:
Router(config-dial-peer)# dtmf-relay rtp-nte
Example:
Step 4 sip-ua Enables SIP user-agent configuration mode.
Example:

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Step 5 notify telephone-event max-duration time Configures the maximum time interval allowed
between two consecutive NOTIFY messages for a
single DTMF event.
Example:
Router(config-sip-ua)# notify telephone-event • max-duration time: Time interval between
max-duration 2000 consecutive NOTIFY messages for a single
DTMF event, in milliseconds. Range is from
500 to 3000. Default is 2000.
Example:
Troubleshooting Tips
The dial-peer section of the show running-config command output displays DTMF relay status when it
is configured, as shown in this excerpt:
destination-pattern [12]...
monitor probe icmp-ping
dtmf-relay rtp-nte
DTMF Relay Using SIP Notify (Nonstandard)

To use voicemail on a SIP network that connects to a Cisco Unity Express system, use a nonstandard SIP
Notify format. To configure the Notify format, use the sip-notify keyword with the dtmf-relay
command. Using the sip-notify keyword may be required for backward compatibility with Cisco
Unified SRST Versions 3.0 and 3.1.
SUMMARY STEPS

2. dtmf-relay sip-notify
3. exit
4. sip-ua
5. notify telephone-event max-duration time
6. exit

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DETAILED STEPS

Example:
Step 2 dtmf-relay sip-notify Forwards DTMF tones using SIP NOTIFY messages.
Example:
Router(config-dial-peer)# dtmf-relay sip-notify
Example:
Step 4 sip-ua Enables SIP user-agent configuration mode.
Example:
Step 5 notify telephone-event max-duration time Configures the maximum time interval allowed
between two consecutive NOTIFY messages for a
single DTMF event.
Example:
Router(config-sip-ua)# notify telephone-event • max-duration time: Time interval between
max-duration 2000 consecutive NOTIFY messages for a single
DTMF event, in milliseconds. Range is from 500
to 3000. Default is 2000.
Example:
The show sip-ua status command output displays the time interval between consecutive NOTIFY
messages for a telephone event. In the following example, the time interval is 2000 ms:
Router# show sip-ua status
SIP User Agent Status
SIP User Agent for UDP :ENABLED
SIP User Agent for TCP :ENABLED
SIP User Agent bind status(signaling):DISABLED
SIP User Agent bind status(media):DISABLED
SIP early-media for 180 responses with SDP:ENABLED
SIP max-forwards :6
SIP DNS SRV version:2 (rfc 2782)
NAT Settings for the SIP-UA
Role in SDP:NONE
Check media source packets:DISABLED
Maximum duration for a telephone-event in NOTIFYs:2000 ms
SIP support for ISDN SUSPEND/RESUME:ENABLED

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Where to Go Next
Redirection (3xx) message handling:ENABLED
SDP application configuration:

Version line (v=) required
Owner line (o=) required
Timespec line (t=) required
Media supported:audio image
Network types supported:IN
Address types supported:IP4
Transport types supported:RTP/AVP udptl
Where to Go Next
If you want to configure video parameters, see the “Setting Video Parameters” section on page 345.

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CHAPTER 12
Setting Video Parameters
This chapter describes how to set video parameters for a Cisco Unified Survivable Remote Site
Telephony (SRST) Router.
Contents
• Prerequisites for Setting Video Parameters, page 345
• Restrictions for Setting Video Parameters, page 346
• Information About Setting Video Parameters, page 346
• How to Set Video Parameters for Cisco Unified SRST, page 349
• Troubleshooting Video for Cisco Unified SRST, page 358
Prerequisites for Setting Video Parameters

• Ensure that you are using Cisco Unified SRST 4.0 or a later version.
• Ensure that you are using Cisco Unified CM 4.0 or a later version.
• Ensure that the Cisco IP phones are registered with the Cisco Unified SRST router. Use the show
ephone registered command to verify ephone registration.
• Ensure that the connection between the Cisco Unified Video Advantage application and the
Cisco Unified IP phone is up.
From a PC with Cisco Unified Video Advantage 1.02 or a later version installed, ensure that the line
between the Cisco Unified Video Advantage and the Cisco Unified IP phone is green. For more
information, see Cisco Unified Video Advantage End User Guides.
• Ensure that the correct video firmware is installed on the Cisco Unified IP phone. Use the show
ephone phone-load command to view current ephone firmware. The following lists the minimum
firmware version for video-enabled Cisco Unified IP phones:
– Cisco Unified IP Phone 7940G version 6.0(4)

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Chapter 12 Setting Video Parameters
Restrictions for Setting Video Parameters
• Perform basic Cisco Unified SRST configuration. For more information, see Cisco Unified SRST
V4.0: Setting Up the Network.
• Perform basic ephone configuration. For more information, see Cisco Unified SRST V4.0: Setting
Up Cisco Unified IP Phones.
Restrictions for Setting Video Parameters

• This feature supports only the following video codecs:
– H.261
– H.263
– H.264 (for CUVA from SRST 7.1)
• This feature supports only the following video formats:
– Common Intermediate Format (CIF): Resolution 352x288
– One-Quarter Common Intermediate Format (QCIF): Resolution 176x144
– Sub QIF (SQCIF): Resolution 128x96
– 4CIF: Resolution 704x576
– 16CIF: Resolution 1408x1152
• The call start fast feature is not supported with an H.323 video connection. You must configure
call start slow for H.323 video.
• Video capabilities are configured per ephone, not per line.
• All call feature controls (for example, mute and hold) apply to both audio and video calls, if
applicable.
• This feature does not support the following:
– Dynamic addition of video capability: The video capability must be present before the call setup
starts to allow the video connection.
– T-120 data connection between two SCCP endpoints
– Video security
– Far-end camera control (FECC) for SCCP endpoints
– Video codec renegotiation: The negotiated video codec must match or the call falls back to
audio-only. The negotiated codec for the existing call can be used for a new call.
– Video codec transcoding
• When a video-capable endpoint connects to an audio-only endpoint, the call falls back to audio-only.
During audio-only calls, video messages are skipped.
Information About Setting Video Parameters

This feature allows you to set video parameters for the Cisco Unified SRST to maintain close feature
parity with Cisco Unified CM. When the Cisco Unified SRST is enabled, Cisco Unified IP phones do
not have to be reconfigured for video capabilities because all ephones retain the same configuration used

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with Cisco Unified CM. However, you must enter call-manager-fallback configuration mode to set video
parameters for Cisco Unified SRST. The feature set for video is the same as that for Cisco Unified SRST
audio calls.
To set video parameters, you should understand the following concepts:
• Matching Endpoint Capabilities, page 347
• Retrieving Video Codec Information, page 347
• Call Fallback to Audio-Only, page 347
• Call Setup for Video Endpoints, page 347
• Flow of the RTP Video Stream, page 348
Matching Endpoint Capabilities

Endpoint capabilities are stored in the Cisco Unified SRST during phone registration. These capabilities
are used to match with other endpoints during call setup. Endpoints can update at any time; however, the
router recognizes endpoint-capability changes only during call setup. If a video feature is added to a
phone, the information about it is updated in the router’s internal data structure, but that information does
not take effect until the next call. If a video feature is removed, the router continues to see the video
capability until the call is terminated but no video stream is exchanged between the two endpoints.
Note The endpoint-capability match is executed every time a new call is set up or an existing call is resumed.
Retrieving Video Codec Information

Voice gateways use dial-peer configurations to retrieve codec information for audio codecs. Video codec
selection is done by the endpoints and is not controlled by the H.323 service-provider interface (SPI)
through dial-peer or other configuration. The video-codec information is retrieved from the SCCP
endpoint using a capabilities request during call setup.
Call Fallback to Audio-Only

When a video-capable endpoint connects to an audio-only endpoint, the call falls back to an audio-only
connection. Also, for certain features such as conferencing, where video support is not available, the call
falls back to audio-only.
Cisco Unified SRST routers use a call-type flag to indicate whether the call is video-capable or
audio-only. The call-type flag is set to video when the video capability is matched or set to audio-only
when connecting to an audio-only TDM or an audio-only SIP endpoint.
Note During an audio-only connection, all video-related media messages are skipped.
Call Setup for Video Endpoints

The process for handling SCCP video endpoints is the same as that for handling SCCP audio endpoints.
The video call must be part of the audio call. If the audio call setup fails, the video call fails.

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During call setup for video, media setup handling determines if a video-media path is required or not. If
so, the corresponding video-media-path setup actions are taken.
• For an SCCP endpoint, video-media-path setup includes sending messages to the endpoints to open
a multimedia path and start the multimedia transmission.
• For an H.323 endpoint, video-media-path setup includes an exchange between the endpoints to open
a logical channel for the video stream.
A call-type flag is set during call setup on the basis of the endpoint-capability match. After call setup,
the call-type flag is used to determine whether an additional video-media path is required. Call signaling
is managed by the Cisco Unified CME router, and the media stream is directly connected between the
two video-enabled SCCP endpoints on the same router. Video-related commands and flow-control
messages are forwarded to the other endpoint. Routers do not interpret these messages.
Call Setup Between Two Local SCCP Endpoints

For interoperation between two local SCCP endpoints (that exist on the same router), video call setup
uses all existing audio-call-setup handling, except during media setup. During media setup, a message
is sent to establish the video-media path. If the endpoint responds, the video-media path is established
and a start-multimedia-transmission function is called.
Call Setup Between SCCP and H.323 Endpoints

Call setup between SCCP and H.323 endpoints is the same as it is between SCCP endpoints except that,
if video capability is selected, the event is posted to the H.323 call leg to send out a video open logical
channel (OLC) and the gateway generates an OLC for the video channel. Because the router needs to
both terminate and originate the media stream, video must be enabled on the router before call setup
begins.
Call Setup Between Two SCCP Endpoints Across an H.323 Network

If call setup between SCCP endpoints occurs across an H.323 network, the setup is a combination of the
processes listed in the previous two sections. The router controls the video media setup between the two
endpoints, and the event is posted to the H.323 call leg so that the gateway can generate an OLC.
Flow of the RTP Video Stream

For video streams between two local SCCP endpoints, the Real-Time Transport Protocol (RTP) stream
is in flow-around mode. For video streams between SCCP and H.323 endpoints or two SCCP endpoints
on different Cisco Unified CME routers, the RTP stream is in flow-through mode.
• Media flow-around mode enables RTP packets to stream directly between the endpoints of a VoIP
call without the involvement of the gateway. By default, the gateway receives the incoming media,
terminates the call, and then reoriginates it on the outbound call leg. In flow-around mode, only
signaling data is passed to the gateway, improving scalability and performance.
• Media flow-through mode involves the same video-media path as for an audio call. Media packets
flow through the gateway, thus hiding the networks from each other.

12-348
How to Set Video Parameters for Cisco Unified SRST
To display information about RTP named-event packets, such as caller-ID number, IP address, and port
for both the local and remote endpoints, use the show voip rtp connection command as shown in the
following sample output:
Router# show voip rtp connections
VoIP RTP active connections :

No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP
1 102 103 18714 18158 10.1.1.1 192.168.1.1
2 105 104 17252 19088 10.1.1.1 192.168.1.1
Found 2 active RTP connections
============================

When the Cisco Unified SRST is enabled, Cisco Unified IP phones do not have to be reconfigured for
video capabilities because all ephones retain the same configuration used with Cisco Unified CM.
However, you can set video parameters for Cisco Unified SRST.
Setting Video parameters for Cisco Unified SRST involves the following tasks:
• Configuring Slow Connect Procedures, page 349
• Verifying Cisco Unified SRST, page 350
• Setting Video Parameters for Cisco Unified SRST, page 356
Configuring Slow Connect Procedures

Video streams require slow-connect procedures for Cisco Unified SRST. H.323 endpoints require a slow
connect because the endpoint-capability match occurs after the connect message.
Note For more information about slow-connect procedures, see Configuring Quality of Service for Voice.
Use the following procedure to configure slow-connect procedures.
SUMMARY STEPS
1. enable
4. h323
5. call start slow

12-349
DETAILED STEPS

Example:
Router> enable
Example:
Step 3 voice service voip Enters voice-service configuration mode.
Example:
Step 4 h323 Enters H.323 voice-service configuration mode.
Example:
Router(config-voi-serv)# h323
Step 5 call start slow Forces an H.323 gateway to use slow-connect procedures
for all VoIP calls.
Example:
Router(config-serv-h323)# call start slow
Verifying Cisco Unified SRST

Use the following procedure to verify that the Cisco Unified SRST feature is enabled and to verify
Cisco Unified IP phone configuration settings.
SUMMARY STEPS
1. enable
2. show running config
3. show call-manager-fallback all

12-350
DETAILED STEPS

Example:
Router> enable
Step 2 show running config Displays the entire contents of the running configuration
file.
Example:
Router# show running config
Step 3 show call-manager-fallback all Displays the detailed configuration of all Cisco Unified IP
phones, directory numbers, voice ports, and dial peers in
your network while in fallback mode.
Example:
Router# show call-manager-fallback all
Note Use the Settings display on the Cisco Unified IP phones in your network to verify that the default router
IP address on the phones matches the IP address of the Cisco Unified SRST router.
Examples
The following example shows output from the show call-manager-fallback all command:
Router# show call-manager-fallback all
CONFIG (Version=3.3)
=====================
Version 3.3
For on-line documentation please see:
www.cisco.com/univercd/cc/td/doc/product/access/ip_ph/ip_ks/index.htm

max-video-bit-rate 384(kbps)
max-ephones 52
max-dn 110
dspfarm units 0
dspfarm transcode sessions 0
huntstop
dialplan-pattern 1 4084442... extension-length 4
voicemail 6001
moh music-on-hold.au
time-format 24
date-format dd-mm-yy
timezone 0 Greenwich Standard Time
call-forward pattern .T
transfer-pattern .T
keepalive 45
timeout interdigit 10
timeout busy 10
timeout ringing 180
caller-id name-only: enable
Limit number of DNs per phone:

12-351
7910: 34
7935: 34
7936: 34
7940: 34
7960: 34
7970: 34
Log (table parameters):
max-size: 150
retain-timer: 15
local directory service: enabled.
ephone-dn 1
number 1001
name 1001
description 1001
label 1001
preference 0 secondary 9
huntstop
call-waiting beep
ephone-dn 2
number 1002
name 1002
description 1002
huntstop
call-waiting beep
ephone-dn 3
huntstop
call-waiting beep
ephone-dn 4
huntstop
call-waiting beep
ephone-dn 5
huntstop
call-waiting beep
ephone-dn 6
huntstop
call-waiting beep
ephone-dn 7
huntstop
call-waiting beep
ephone-dn 8
huntstop
call-waiting beep
ephone-dn 9

12-352
huntstop
call-waiting beep
ephone-dn 10
huntstop
call-waiting beep
ephone-dn 11
huntstop
call-waiting beep
ephone-dn 12
huntstop
call-waiting beep
ephone-dn 13
huntstop
call-waiting beep
ephone-dn 14
huntstop
call-waiting beep
ephone-dn 15
huntstop
call-waiting beep
ephone-dn 16
huntstop
call-waiting beep
ephone-dn 17
huntstop
call-waiting beep
ephone-dn 18
huntstop
call-waiting beep
ephone-dn 19
huntstop
call-waiting beep
ephone-dn 20
huntstop
call-waiting beep
Number of Configured ephones 0 (Registered 2)
voice-port 50/0/1
station-id number 1001
station-id name 1001

12-353
timeout ringing 8
!
voice-port 50/0/2
station-id number 1002
station-id name 1002
timeout ringing 8
!
voice-port 50/0/3
!
voice-port 50/0/4
!
voice-port 50/0/5
!
voice-port 50/0/6
!
voice-port 50/0/7
!
voice-port 50/0/8
!
voice-port 50/0/9
!
voice-port 50/0/10
!
voice-port 50/0/11
!
voice-port 50/0/12
!
voice-port 50/0/13
!
voice-port 50/0/14
!
voice-port 50/0/15
!
voice-port 50/0/16
!
voice-port 50/0/17
!
voice-port 50/0/18
!
voice-port 50/0/19
!
voice-port 50/0/20
!

huntstop
call-forward noan 6001
progress_ind setup enable 3
port 50/0/1

huntstop
call-forward noan 6001
port 50/0/2

huntstop
port 50/0/3

12-354

huntstop
port 50/0/4

huntstop
port 50/0/5

huntstop
port 50/0/6

huntstop
port 50/0/7

huntstop
port 50/0/8

huntstop
port 50/0/9

huntstop
port 50/0/10

huntstop
port 50/0/11

huntstop
port 50/0/12

huntstop
port 50/0/13

huntstop
port 50/0/14

huntstop
port 50/0/15

huntstop

12-355
port 50/0/16

huntstop
port 50/0/17

huntstop
port 50/0/18

huntstop
port 50/0/19

huntstop
port 50/0/20
tftp-server system:/its/SEPDEFAULT.cnf
tftp-server system:/its/SEPDEFAULT.cnf alias SEPDefault.cnf
tftp-server system:/its/XMLDefault.cnf.xml alias XMLDefault.cnf.xml
tftp-server system:/its/ATADefault.cnf.xml
tftp-server system:/its/united_states/7960-tones.xml alias United_States/7960-tones.xml
tftp-server system:/its/united_states/7960-font.xml alias
English_United_States/7960-font.xml
tftp-server system:/its/united_states/7960-dictionary.xml alias
English_United_States/7960-dictionary.xml
tftp-server system:/its/united_states/7960-kate.xml alias
English_United_States/7960-kate.xml
tftp-server system:/its/united_states/SCCP-dictionary.xml alias
English_United_States/SCCP-dictionary.xml
tftp-server system:/its/XMLDefault7960.cnf.xml alias SEP003094C2772E.cnf.xml
tftp-server system:/its/XMLDefault7960.cnf.xml alias SEP001201372DD1.cnf.xml
tftp-server system:/its/XMLDefault7960.cnf.xml alias SEPFFDD00000001.cnf.xml
tftp-server system:/its/XMLDefault7960.cnf.xml alias SEPFFDD0000000A.cnf.xml
tftp-server system:/its/XMLDefault7960.cnf.xml alias SEPFFDD0000000B.cnf.xml
tftp-server system:/its/XMLDefault7960.cnf.xml alias SEPFFDD0000000C.cnf.xml
tftp-server system:/its/XMLDefault7960.cnf.xml alias SEPFFDD0000000D.cnf.xml
tftp-server system:/its/XMLDefault7960.cnf.xml alias SEPFFDD0000000E.cnf.xml
tftp-server system:/its/XMLDefault7960.cnf.xml alias SEPFFDD0000000F.cnf.xml
Setting Video Parameters for Cisco Unified SRST

Using the following procedure to set the maximum bit rate for all video-capable phones in a
Cisco Unified SRST system.

12-356
SUMMARY STEPS
1. enable
4. video
5. maximum bit-rate value
DETAILED STEPS

Example:
Router> enable
Example:
Example:
Step 4 video Enters call-manager-fallback video configuration mode.
Example:
Router(config-call-manager-fallback)# video
Step 5 maximum bit-rate value Sets the maximum IP phone video bandwidth, in kbps. The
range is 0 to 10000000. The default is 10000000.
Example:
Router(conf-cm-fallback-video)# maximum
bit-rate 256
Examples
The following example shows the configuration for video with Cisco Unified SRST:
video
maximum bit-rate 384
max-ephones 52
max-dn 110
dialplan-pattern 1 4084442... extension-length 4
transfer-pattern .T
keepalive 45
voicemail 6001
call-forward pattern .T

12-357
Troubleshooting Video for Cisco Unified SRST

time-format 24
date-format dd-mm-yy
!
Troubleshooting Video for Cisco Unified SRST

Use the following commands to troubleshoot Video for Cisco Unified SRST.
• For SCCP endpoint troubleshooting, use the following debug commands:
– debug cch323 video: Enables video debugging trace on the H.323 SPI.
– debug ephone detail: Debugs all Cisco Unified IP phones that are registered to the router and
displays error and state levels.
– debug h225 asn1: Displays Abstract Syntax Notation One (ASN.1) contents of H.225 messages
that have been sent or received.
– debug h245 asn1: Displays ASN.1 contents of H.245 messages that have been sent or received.
– debug voip ccapi inout: Displays the execution path through the call-control-application
programming interface (CPI).
• For ephone troubleshooting, use the following debug commands:
– debug ephone message: Enables message tracing between Cisco ephones.
– debug ephone register: Sets registration debugging for ephones.
– debug ephone video: Sets ephone video traces, which provide information about different
video states for the call, including video capabilities selection, start, and stop.
• For basic video-to-video call checking, use the following show commands:
– show call active video: Displays call information for SCCP video calls in progress.
– show ephone offhook: Displays information and packet counts for ephones that are currently
off hook.
– show voip rtp connections: Displays information about RTP named-event packets, such as
caller ID number, IP address, and port, for both the local and remote endpoints.
Where to Go Next
To monitor and maintain Cisco Unified SRST, see the “Monitoring and Maintaining Cisco Unified
SRST” section on page 359.

12-358
CHAPTER 13
Monitoring and Maintaining Cisco Unified
SRST
To monitor and maintain Cisco Unified Survivable Remote Site Telephony (SRST), use the following
commands in privileged EXEC mode.
Command Purpose
Router# show call-manager-fallback all Displays the detailed configuration of all the
Cisco Unified IP phones, voice ports, and dial peers of the
Router# show call-manager-fallback dial-peer Displays the output of the dial peers of the
Router# show call-manager-fallback ephone-dn Displays Cisco Unified IP Phone destination numbers when in
call manager fallback mode.
Router# show call-manager-fallback voice-port Displays output for the voice ports.
Router# show dial-peer voice summary Displays a summary of all voice dial peers.
Router# show ephone phone Displays Cisco Unified IP Phone status.
Router# show ephone offhook Displays Cisco Unified IP Phone status for all phones that are
off hook.
Router# show ephone registered Displays Cisco Unified IP Phone status for all phones that are
currently registered.
Router# show ephone remote Displays Cisco Unified IP Phone status for all nonlocal phones
(phones that have no Address Resolution Protocol [ARP] entry).
Router# show ephone ringing Displays Cisco Unified IP Phone status for all phones that are
ringing.
Router# show ephone summary Displays a summary of all Cisco Unified IP Phones.
Router# show ephone telephone-number phone-number Displays Unified IP Phone status for a specific phone number.
Router# show ephone unregistered Displays Unified IP Phone status for all unregistered phones.
Router# show ephone-dn tag Displays Unified IP Phone destination numbers.
Router# show ephone-dn summary Displays a summary of all Cisco Unified IP Phone destination
numbers.
Router# show ephone-dn loopback Displays Cisco Unified IP Phone destination numbers in
loopback mode.

13-359
Chapter 13 Monitoring and Maintaining Cisco Unified SRST
Command Purpose
Router# show running-config Displays the configuration.
Router # show sip-ua status registrar Display SIP registrar clients.
Router# show voice port summary Displays a summary of all voice ports.
Router # show voice register all Displays all SIP SRST configurations , SIP phone registrations
and dial peer info.
Router # show voice register global Displays voice register global config.
Router # show voice register pool all Displays all config SIP phone voice register pool detail info.
Router # show voice register pool <tag> Displays specific SIP phone voice register pool detail info.
Router # show voice register dial-peers Displays SIP-SRST created dial peer.
Router # show voice register dn all Displays all config voice register dn detail info.
Router # show voice register dn <tag> Displays specific voice register dn detail info.

13-360
A P P E NDIX A
Configuring Cisco Unified SIP SRST Features
Using Redirect Mode
Note This chapter applies to version 3.0 only.
This chapter describes Cisco Unified Session Initiation Protocol (SIP) Survivable Remote Site
Telephony (SRST) features using redirect mode.
Contents
• Prerequisites for Cisco Unified SIP SRST Features Using Redirect Mode, page A-1
• Restrictions for Cisco Unified SIP SRST Features Using Redirect Mode, page A-1
• Information About Cisco Unified SIP SRST Features Using Redirect Mode, page A-2
• How to Configure Cisco Unified SIP SRST Features Using Redirect Mode, page A-2
• Configuration Examples for Cisco Unified SIP SRST Features Using Redirect Mode, page A-6
• Where to Go Next, page A-8
Prerequisites for Cisco Unified SIP SRST Features Using

Redirect Mode
Complete the prerequisites documented in the “Prerequisites for Configuring Cisco Unified SIP SRST”
section on page 1-9 in the “Cisco Unified SRST Feature Overview” section on page 1-1.
Restrictions for Cisco Unified SIP SRST Features Using

Redirect Mode
See the restrictions documented in the “Restrictions for Configuring Cisco Unified SIP SRST” section
on page 1-9 section in the “Cisco Unified SRST Feature Overview” section on page 1-1.

A-1
Appendix A Configuring Cisco Unified SIP SRST Features Using Redirect Mode
Information About Cisco Unified SIP SRST Features Using Redirect Mode
Information About Cisco Unified SIP SRST Features Using

Redirect Mode
Cisco Unified SIP SRST provides backup to an external SIP call control (IP-PBX) by providing basic
registrar and redirect services. These services are used by a SIP IP phone in the event of a WAN
connection outage when the SIP phone is unable to communicate with its primary SIP proxy. The Cisco
Unified SIP SRST device also provides PSTN gateway access for placing and receiving PSTN calls.
To make maximum use of the Cisco Unified SIP SRST service, the local SIP IP phones should support
dual (concurrent) registration with both their primary SIP proxy or registrar and the
Cisco Unified SIP SRST backup registrar. Cisco Unified SIP SRST works for the following types of
calls:
• Local SIP IP phone to local SIP phone, if the main proxy is unavailable.
• Additional services like class of restriction (COR) for local SIP IP phones to the outgoing PSTN.
For example, to block outgoing 1-900 numbers.
How to Configure Cisco Unified SIP SRST Features Using

Redirect Mode
• Configuring Call Redirect Enhancements to Support Calls Between SIP IP Phones for Cisco Unified
SIP SRST, page A-2 (required)
• Configuring Sending 300 Multiple Choice Support, page A-5 (required)
Configuring Call Redirect Enhancements to Support Calls Between SIP IP

Phones for Cisco Unified SIP SRST
The call redirect enhancement supports calls from a local SIP phone to another local SIP phone through
the Cisco IOS voice gateway. Prior to this enhancement, an attempt by a SIP phone to contact another
local SIP phone using the Cisco IOS voice gateway as if it were a SIP proxy or redirect server would
fail. However, the Cisco IOS voice gateway can now act as a SIP redirect server. The voice gateway
responds to the originator with a SIP Redirect message, allowing the SIP phone that originated the call
to establish a call to its destination.
The redirect ip2ip (voice service) and redirect ip2ip (dial-peer) commands allow you to enable the SIP
functionality, globally or on a specific inbound dial peer. The default application on Cisco Unified SIP
SRST supports IP-to-IP redirection.
• Configuring Call Redirect Enhancements to Support Calls Globally, page A-3
• Configuring Call Redirect Enhancements to Support Calls on a Specific VoIP Dial Peer, page A-4

A-2
How to Configure Cisco Unified SIP SRST Features Using Redirect Mode
Configuring Call Redirect Enhancements to Support Calls Globally

To enable global IP-to-IP call redirection for all VoIP dial peers, use voice service configuration mode.
Note When IP-to-IP redirection is configured in dial-peer configuration mode, the configuration for the
specific dial peer takes precedence over the global configuration entered under voice service
configuration mode.
SUMMARY STEPS
1. enable
4. redirect ip2ip
5. end
DETAILED STEPS

Example:
Router> enable
Example:
Example:
Step 4 redirect ip2ip Redirects SIP phone calls to SIP phone calls globally on a
gateway using the Cisco IOS voice gateway.
Example:
Router(config-voi-srv)# redirect ip2ip
Example:
Router(config-voi-srv)# end

A-3
Configuring Call Redirect Enhancements to Support Calls on a Specific VoIP Dial Peer
To enable IP-to-IP call redirection for a specific VoIP dial peer, configure it on an inbound dial peer in
dial-peer configuration mode. The default application on Cisco Unified SIP SRST supports IP-to-IP
redirection.
Note When IP-to-IP redirection is configured in dial-peer configuration mode, the configuration for the
specific dial peer takes precedence over the global configuration entered under voice service
configuration mode.
Restrictions
The redirect ip2ip command must be configured on an inbound dial peer of the gateway.
SUMMARY STEPS
1. enable
5. redirect ip2ip
6. end
DETAILED STEPS

Example:
Router> enable
Example:
• tag—A number that uniquely identifies the dial peer
Example: (this number has local significance only).
• voip—Indicates that this is a VoIP peer using voice
encapsulation on the POTS network and is used for
configuring redirect.

A-4

Step 4 application application-name Enables a specific application on a dial peer.
• For SIP, the default Tool Command Language (Tcl)
Example: application (from the Cisco IOS image) is session and
Router(config-dial-peer)# application session can be applied to both VoIP and POTS dial peers.
• The application must support IP-to-IP redirection.
Step 5 redirect ip2ip Redirects SIP phone calls to SIP phone calls on a specific
VoIP dial peer using the Cisco IOS voice gateway.
Example:
Router(config-dial-peer)# redirect ip2ip
Example:
Configuring Sending 300 Multiple Choice Support

Prior to Cisco IOS Release 12.2(15)ZJ, when a call was redirected, the SIP gateway would send a 302
Moved Temporarily message. The first longest match route on a gateway (dial-peer destination pattern)
was used in the Contact header of the 302 message. With Release 12.2(15)ZJ, if multiple routes to a
destination exist for a redirected number (multiple dial peers are matched), the SIP gateway sends a 300
Multiple Choice message, and the multiple routes in the Contact header are listed.
The configuration below allows users to choose the order in which the routes appear in the Contact
header.
SUMMARY STEPS
1. enable
4. sip
5. redirect contact order [best-match | longest-match]
6. end

A-5
Configuration Examples for Cisco Unified SIP SRST Features Using Redirect Mode
DETAILED STEPS

Example:
Router> enable
Example:
Example:
Example:
Step 5 redirect contact order [best-match | longest- Sets the order of contacts in the 300 Multiple Choice
match] message. The keywords are defined as follows:
• best-match—(Optional) Uses the current system
Example: configuration to set the order of contacts.
Router(conf-serv-sip)# redirect contact order
best-match • longest-match—(Optional) Sets the contact order by
using the destination pattern longest match first, and
then the second longest match, the third longest match,
and so on. This is the default.
Example:
Router(config-serv-sip)# end
Configuration Examples for Cisco Unified SIP SRST

Features Using Redirect Mode
This section provides the following configuration example:
• Cisco Unified SIP SRST: Example, page A-7

A-6
Configuration Examples for Cisco Unified SIP SRST Features Using Redirect Mode
Cisco Unified SIP SRST: Example

This section provides a configuration example to match the configuration tasks in the previous sections.
!
! Sets up the registrar server and enables IP-to-IP redirection and 300
! Multiple Choice support.
!
voice service voip
redirect ip2ip
sip
redirect contact order best-match
!
! Configures the voice-class codec with G.711uLaw and G729 codecs. The codecs are
! applied to the voice register pools.
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729br8
!
! The voice register pools define various pools that are used to match
! incoming REGISTER requests and create corresponding dial peers.
!
id mac 0030.94C2.A22A
preference 5
cor incoming call91 1 91011
translate-outgoing called 1
proxy 10.2.161.187 preference 1 monitor probe icmp-ping
alias 1 94... to 91011 preference 8
voice-class codec 1
!
id ip 192.168.0.3 mask 255.255.255.255
preference 5
cor outgoing call95 1 91021
proxy 10.2.161.187 preference 1
voice-class codec 1
!
id network 10.2.161.0 mask 255.255.255.0
number 1 95... preference 1
preference 5
cor incoming call95 1 95011
cor outgoing call95 1 95011
proxy 10.2.161.187 preference 1 monitor probe icmp-ping
max registrations 5
voice-class codec 1
!
id network 10.2.161.0 mask 255.255.255.0
number 1 94... preference 1
preference 5
cor incoming everywhere default
cor outgoing everywhere default
proxy 10.2.161.187 preference 1
max registrations 2
voice-class codec 1
!
! Configures translation rules to be applied in the voice register pools.
!
translation-rule 1

A-7
Where to Go Next
Rule 0 94 91
!
! Sets up proxy monitoring.
!
call fallback active
!
name 95
name 94
name 91
!
! Configures COR values to be applied to the voice register pool.
!
member 95
!
member 94
!
member 91
!
dial-peer cor list everywhere
member 95
member 94
member 91
!
! Configures a voice port and a POTS dial peer for calls to and from the PSTN endpoints.
voice-port 1/0/0
!
corlist incoming call91
corlist outgoing call91
port 1/0/0
!
Where to Go Next
For additional information, see the “Additional References” section on page 1-30 in the “Cisco Unified
SRST Feature Overview” section on page 1-1 chapter.

A-8
Where to Go Next

A-9
Where to Go Next

A-10
A P P E NDIX B
Integrating Cisco Unified Communications
Manager and Cisco Unified SRST to Use Cisco
Unified SRST as a Multicast MOH Resource
This chapter describes how to configure Cisco Unified CM and Cisco Unified SRST to allow Cisco
Unified CM to use Cisco Unified SRST gateways as multicast music-on-hold (MOH) resources during
fallback and normal Cisco Unified CM operation. A distributed MOH design with local gateways
providing MOH eliminates the need to stream MOH across a WAN and saves bandwidth.
Finding Feature Information in This Module

Your Cisco Unified CME version may not support all of the features documented in this module. For a
list of the versions in which each feature is supported, see the “Feature Information for Cisco Unified SRST
as a Multicast MOH Resource” section on page B-42.
Contents
• Prerequisites for Using Cisco Unified SRST Gateways as a Multicast MOH Resource, page B-12
• Restrictions for Using Cisco Unified SRST Gateways as a Multicast MOH Resource, page B-12
• Information About Using Cisco Unified SRST Gateways as a Multicast MOH Resource, page B-13
• How to Use Cisco Unified SRST Gateways as a Multicast MOH Resource, page B-18
• Configurations Examples for Cisco Unified SRST Gateways, page B-41
• Where to Go Next, page B-43

B-11
Appendix B Integrating Cisco Unified Communications Manager and Cisco Unified SRST to Use Cisco
Prerequisites for Using Cisco Unified SRST Gateways as a Multicast MOH Resource
Prerequisites for Using Cisco Unified SRST Gateways as a

Multicast MOH Resource
• Multicast MOH for H.323 and MGCP is supported on Cisco Unified CM 3.1.1 and higher versions.
• Cisco Unified CM must be configured as follows:
– With multicast MOH enabled.
– With Media Resource Groups (MRGs) and Media Resource Group Lists (MRGLs) controlling
which devices receive multicast MOH and which devices receive unicast MOH.
– With Cisco Unified CM regions assigned so that G.711 is used whenever a Cisco Unified SRST
multicast MOH resource is invoked.
• The Cisco Unified SRST gateways must run on Cisco Unified SRST 3.0 on Cisco IOS
Release 12.2(15)ZJ2 or a later release.
• Cisco Unified SRST must be registered to Cisco Unified CM using protocol such as H.323, MGCP,
or SIP.
• For branches that do not run Cisco Unified SRST, Cisco Unified CM multicast MOH packets must
cross the WAN. To accomplish this, you must have multicast routing enabled in your network. For
more information about multicast routing, see the “IP Multicast” section of Cisco IOS IP
Configuration Guide, Release 12.4T.
• With Cisco IOS earlier than 12.3(14)T, configure Cisco Unified SRST as your MGCP gateway’s
fallback mode using the ccm-manager fallback-mgcp and call application alternate commands.
With Cisco IOS releases after 12.3(14)T, the ccm-manager fallback-mgcp and service commands
must be configured. Configuring these two commands allows Cisco Unified SRST to assume control
over the voice port and over call processing on the MGCP gateway. A complete configuration
describing setting up Cisco Unified SRST as your fallback mode is shown in Cisco Unified
Communications Manager Administration Guide, Release 5.1(3) Survivable Remote Site Telephony
Configuration.
Restrictions for Using Cisco Unified SRST Gateways as a

• Cisco Unified SRST multicast MOH does not support unicast MOH.
• Only a single Cisco Unified CM audio source can be used throughout the network. However, the
audio files on each Cisco Unified SRST gateway’s flash memory can be different.
• Cisco Unified SRST multicast MOH supports G.711 only.
• Unified SRST multicast MOH does not support co-location of tunnels on the same device.
• Multicast MOH support for H.323 is unavailable in all versions of Cisco Unified
Communications Manager 3.3.2. For more information, see CSCdz00697 using the Bug Toolkit.
• In the Cisco IOS Release 12.2(15)ZJ image for Cisco 1700 series gateways, Cisco Unified SRST
multicast MOH does not include support for H.323 mode.

B-12
Appendix B Integrating Cisco Unified Communications Manager and Cisco Unified SRST to Use Cisco Unified
Information About Using Cisco Unified SRST Gateways as a Multicast MOH Resource
Information About Using Cisco Unified SRST Gateways as a

To configure Cisco Unified SRST gateways as an MOH resource, you should understand the following
concepts:
• Cisco Unified SRST Gateways and Cisco Unified Communications Manager, page B-13
• Codecs, Port Numbers, and IP Addresses, page B-14
• Multicast MOH Transmission, page B-16
• MOH from a Live Feed, page B-16
• MOH from Flash Files, page B-17
Cisco Unified SRST Gateways and Cisco Unified Communications

Manager
Cisco Unified SRST gateways can be configured to multicast Real-Time Transport Protocol (RTP)
packets from flash memory during fallback and normal Cisco Unified CM operation. To make this
happen, Cisco Unified Communications Manager must be configured for multicast MOH so that the
audio packets do not cross the WAN. Instead, audio packets are broadcast from the flash memory of
Cisco Unified SRST gateways to the same multicast MOH IP address and port number configured for
Cisco Unified Communications Manager multicast MOH. IP phones at remote sites are able to pick up
RTP packets that are multicast from the local branch gateways instead of from the central Cisco Unified
CM.
Multicast MOH for PSTN callers is supported when the Cisco Unified SRST router is used as the
Cisco IOS voice gateway for Cisco Unified CM. In this state the Cisco Unified SRST function of the
router remains in standby mode (no phones registered) with call control of the phones and gateway
provided by Cisco Unified Communications Manager. This feature does not apply when the Cisco
Unified SRST router is in fallback mode (phones are registered to Cisco Unified SRST). Instead, MOH
is provided to PSTN callers via a direct internal path rather than through the multicast loopback
interface.
Figure 1 shows a sample configuration in which all phones are configured by Cisco Unified
Communications Manager to receive multicast MOH through port number 16384 and IP address
239.1.1.1. Cisco Unified CM is configured so that multicast MOH cannot reach the WAN, and local
Cisco Unified SRST gateways are configured to send audio packets from their flash files to port number
16384 and IP address 239.1.1.1. Cisco Unified CM and the IP phones are spoofed and behave as if Cisco
Unified CM were originating the multicast MOH.
Note Phone users at the central site would use multicast MOH from the central site.

B-13
Figure 1 Multicast MOH from Cisco Unified SRST Flash Memory
Cisco Unified
SRST gateway
V Cisco Unified
SRST gateway
Flash
packets
audio
Cisco Unified
Communications Manager
Cisco Unified
WAN
239.1.1.1 port 16384

SRST gateway
Multicast MOH
239.1.1.1 port 16384.
V
Cisco Unified
SRST gateway
116827
Codecs, Port Numbers, and IP Addresses
Cisco Unified SRST multicast MOH supports G.711 only. Figure 2 shows an example in which G.711
is the only codec used by a central Cisco Unified CM and three branches. In some cases, a Cisco Unified
CM system may use additional codecs. For example, for bandwidth savings, Cisco Unified CM may use
G.711 for multicast MOH and G.729 for phone conversations.
As shown in the example in Figure 2, IP address 10.1.1.1 and port 1000 are used during phone
conversations when G.729 is in use, and IP address 239.1.1.1 and port 16384 are used when a call is
placed on hold and G.711 is in use.

B-14
Figure 2 IP Address and Port Usage for G.711 and G.729 Configuration
Branch 1 calls Branch 2 (G.729 is used).
10.1.1.1 port 1000

V IP Branch 1
WAN
V IP Branch 2
Branch 2 places Branch 1 on hold (G.711 is used).
239.1.1.1 port 16384

Branch 1 V IP
WAN
Branch 2 V IP
Branch 2 takes Branch 1 off hold (G.729 is used).
10.1.1.1 port 1000

V IP Branch 1
WAN
116828
V IP Branch 2
Figure 1 and Figure 2 show all branches using Cisco Unified SRST multicasting MOH. Figure 3 shows
a case in which some gateways are configured with Cisco Unified SRST and other gateways are not.
When the central site and Branch 3 phone users are put on hold by other IP phones in the Cisco Unified
CM system, MOH is originated by Cisco Unified CM. When Branch 1 and Branch 2 phone users are put
on hold by other phone users in the Cisco Unified CM system, MOH is originated by the Cisco Unified
SRST gateways.

B-15
Figure 3 MOH Sources for Cisco Unified SRST and Other Unified SRST IP Phones Using MOH
Branch 1 with Cisco Unified SRST using G.711
239.1.1.1 port 16384

V
Branch 2 with Cisco Unified SRST using G.711
239.1.1.1 port 16384 239.1.1.1 port 16384

239.1.1.1 port 16386 V
WAN
Branch 3 without Cisco Unified SRST using G.729
239.1.1.1 port 16386
116829
V
To enable MOH audio packet transmission through two paths, the Cisco Unified CM MOH server must
be configured with either one IP address and two different port numbers or one port address and two
different IP multicast addresses so that one set of branches can use Cisco Unified SRST multicast MOH
and the other can use Cisco Unified CM multicast MOH.
Multicast MOH Transmission

If Cisco Unified SRST multicast MOH is supported by all branches in a system, such as in Figure 1,
Cisco Unified Communications Manager must be configured to keep all multicast MOH audio packets
from reaching the WAN. When there is a mix of Cisco Unified SRST branches, as shown in Figure 3,
one set of Cisco Unified Communications Manager MOH audio files must reach the WAN and another
set must not. Audio packets from the central Cisco Unified Communications Manager must cross the
WAN to reach branches running Cisco Unified Communications Manager. For branches running
Cisco Unified SRST, the packets must not reach the WAN. For more information about Multicast MOH,
see the “Configuring Cisco Unified SRST for Multicast MOH from an Audio File” section on page B-26.
MOH from a Live Feed

MOH live feed provides live feed MOH streams from an audio device connected to an E&M or FXO port
to Cisco IP phones in SRST mode. Music from a live feed is from a fixed source and is continuously fed
into the MOH playout buffer instead of being read from a flash file.
Cisco Unified SRST is enhanced with the moh-live command. The moh-live command provides live
feed MOH streams from an audio device connected to an E&M or FXO port to Cisco IP phones in SRST
mode. Live feed MOH can also be multicast to Cisco IP phones. For more information about
Cisco Unified SRST MOH live feed, see the “Configuring Cisco Unified SRST for MOH from a Live
Feed” section on page B-36.

B-16
MOH from Flash Files

The MOH Multicast from Flash Files feature facilitates the continuous multicast of MOH audio feed
from files in the flash memories of Cisco Unified SRST branch office routers during Cisco Unified
Communications fallback and normal Cisco Unified Communications service. Multicasting MOH from
individual branch routers saves WAN bandwidth by eliminating the need to stream MOH audio from
central offices to remote branches.
The MOH Multicast from Flash Files feature can act as a backup mechanism to the MOH live feed
feature. Using the Flash to backup the live-feed is the recommend method rather than using just the live
feed feature alone.
Cisco Unified Communications Manager MOH audio files must reach the WAN and another set must
not. Audio packets from the central Cisco Unified CM must cross the WAN to reach branches running
Cisco Unified CM. For branches running Cisco Unified SRST, the packets must not reach the WAN.
Table 1 provides a summary of options for MOH.
Table 1 Music-on-Hold (MOH)
Audio Source Description How to Configure

Flash memory No external audio input is required. Configuring Cisco Unified SRST for
Multicast MOH from an Audio File,
page B-26
Live feed The multicast audio stream has minimal Configuring Cisco Unified SRST for
delay for local IP phones. The MOH MOH from a Live Feed, page B-36
stream for PSTN callers is delayed by a
few seconds. If the live feed audio input
fails, callers on hold hear silence.
Live feed and flash The live feed stream has a few seconds Configuring Cisco Unified SRST for
memory of delay for both PSTN and local IP Multicast MOH from an Audio File,
phone callers. The flash MOH acts as page B-26
backup for the live-feed MoH. and
We recommend this option if you want
Configuring Cisco Unified SRST for
live-feed because it provides guaranteed MOH from a Live Feed, page B-36
MOH if the live-feed input is not found
or fails.

B-17
How to Use Cisco Unified SRST Gateways as a Multicast MOH Resource
How to Use Cisco Unified SRST Gateways as a Multicast

MOH Resource
To use Cisco Unified SRST gateways as a multicast MOH resource, perform the following tasks:
• Configuring Cisco Unified Communications Manager for Cisco Unified SRST Multicast MOH,
page B-18
• Configuring Cisco Unified SRST for Multicast MOH from an Audio File, page B-26
• Configuring Cisco Unified SRST for MOH from a Live Feed, page B-36
For Cisco Unified CM 8.0 or later, see the Configuring MOH-groups for Cisco Unified SRST (fallback)
section in the Cisco Unified Survivable Remote Site Telephony 8.0 Music On Hold Enhancement
document.
Configuring Cisco Unified Communications Manager for Cisco Unified

SRST Multicast MOH
The following sections describe the Cisco Unified CM configuration tasks for Cisco Unified SRST
multicast MOH:
• Configuring the MOH Audio Source to Enable Multicasting, page B-19
• Enabling Multicast on the Cisco Unified Communications Manager MOH Server and Configuring
Port Numbers and IP Addresses, page B-20
• Creating an MRG and an MRGL, Enabling MOH Multicast, and Configuring Gateways, page B-23
• Creating a Region for the MOH Server, page B-25
• Verifying Cisco Unified Communications Manager Multicast MOH, page B-26
To use Cisco Unified SRST gateways as multicast MOH resources, you must configure
Cisco Unified Communications Manager to multicast MOH to the required branch sites. To accomplish
this, you must configure IP addresses, port numbers, the MOH source, and the MOH audio server.
Even though the MOH routing is set up to prevent the Cisco Unified CM-sourced multicast MOH from
actually reaching the WAN and the remote phones, the configured Cisco Unified CM MOH IP port and
address information are still used by Cisco Unified CM to tell the phones which multicast IP address to
listen to for MOH (for the MOH sourced by SRST).
Configuring the MOH server involves designating a maximum number of hops for the audio source. A
configuration of one hop keeps Cisco Unified CM multicast MOH packets from reaching the WAN, thus
spoofing Cisco Unified CM and allowing Cisco Unified SRST multicast MOH packets to be sent from
Cisco Unified SRST gateways to their component phones. For cases in which Cisco Unified CM
multicast must reach gateways that do not run Cisco Unified SRST, use the Cisco IOS ip multicast
boundary command to control where multicast packets go.
After the MOH server is configured, the MOH server must be added to a Media Resource Group (MRG);
the MRG is added to a Media Resource Group List (MRGL); and the designated Cisco Unified CM
branch gateways are configured to use the MRGL.
Five Cisco Unified CM windows are used to configure the MOH server, audio source, MRG, MRGL,
and individual gateways. Figure 4 provides an overview of this process.
The last Cisco Unified CM configuration task involves creating an MOH region that assigns MOH G.711
codec usage for the central site or sites and branch office or offices.

B-18
Regions specify the codecs that are used for audio and video calls within a region and between existing
regions. For information about regions, see the “Region Configuration” section in the Cisco Unified
Communications Manager Administration Guide. From the Cisco Unified Communications
Manager documentation directory, click Maintain and Operate Guides and select the required
Cisco Unified Communications Manager version to locate the administration guide for your version.
Figure 4 Unified Communications Manager Screens for Configuring Multicast MOH
Configure MOH Server
Music On Hold (MOH) Audio Source Configuration Screen
Music On Hold (MOH) Server Configuration Screen
Add Server
Media Resource Group Configuration Screen
Add MRG
Media Resource Group Configuration Screen
Phone Configuration Screen
146319
Gateway Configuration Screen
Configuring the MOH Audio Source to Enable Multicasting

The MOH audio source is a file from which Cisco Unified CM transmits RTP packets. You can create
an audio file or use the default audio file. For Cisco Unified SRST multicast MOH, only one audio source
can be used, even if, for example, one out of 500 sites uses Cisco Unified SRST multicast MOH. In
addition, all Cisco Unified Communications Manager systems must use the same audio source for user
and network MOH because Cisco Unified SRST multicast MOH can stream audio only to a single
multicast IP address and port. For Cisco Unified SRST multicast MOH, the Cisco Unified
Communications Manager audio source file must be configured for G.711 bandwidth.

B-19
Tip The simplest way to create an audio source is to use the default audio source.
Whether you use a default Cisco Unified CM MOH audio source or you create one, the MOH audio
source must be configured for multicasting in the MOH Audio Source Configuration window.
Note that the MOH Audio Source File Status section shows that the MOH audio source file is configured
for four codec formats. If you are planning to use several codecs, ensure that the audio source file
accommodates them.
For further information about the creation of an MOH audio source, see the Cisco Unified
Communications Manager Administration Guide. From the Cisco Unified
Communications Manager documentation directory, click Maintain and Operate Guides and select the
required Cisco Unified CM version.
Use this procedure to configure the MOH audio source to enable multicasting and continuous play.
Note These instructions assume that an MOH audio source file was already created.
Step 1 To enable multicast MOH for the MOH audio source, choose Service > Media Resources > Music On
Hold Audio Source to display the MOH Audio Source Configuration window.
Step 2 Double-click the required audio source listed in the MOH Audio Sources column.
Step 3 In the MOH Audio Source Configuration window, check Allow Multicasting.
Step 4 Click Update.
Enabling Multicast on the Cisco Unified Communications Manager MOH Server and
Configuring Port Numbers and IP Addresses
Enter a base multicast IP address and port number in the Multicast Audio Source Information section of
the MOH Server Configuration window. If you are using Cisco Unified CM multicast MOH and
Cisco Unified SRST multicast MOH (see the “Codecs, Port Numbers, and IP Addresses” section on
page B-14 and the “Multicast MOH Transmission” section on page B-16), you must select a port and IP
address increment method to configure for two sets of port numbers and IP address.
If the Increment Multicast on radio button is set to IP address, each MOH audio source and codec
combination is multicast to different IP addresses but uses the same port number. If it is set to Port
Number, each MOH audio source and codec combination is multicast to the same IP address but uses
different destination port numbers.
Table 2 shows the difference between incrementing on an IP address and incrementing on a port number,
using the base IP address of 239.1.1.1 and the base port number of 16384. The table also matches
Cisco Unified Communications Manager audio sources and codecs to IP addresses and port numbers.

B-20
Table 2 Example of the Differences Between Incrementing Multicast on IP Address and Incrementing Multicast on Port Number
Increment Multicast on IP Address Increment Multicast on Port Number

Destination IP Destination Destination IP Destination
Audio Source Codec Address Port Address Port
1 G.711 mu-law 239.1.1.1 16384 239.1.1.1 16384
1 G.711 a-law 239.1.1.2 16384 239.1.1.1 16386
1 G.729 239.1.1.3 16384 239.1.1.1 16388
1 Wideband 239.1.1.4 16384 239.1.1.1 16390
2 G.711 mu-law 239.1.1.5 16384 239.1.1.1 16392
2 G.711 a-law 239.1.1.6 16384 239.1.1.1 16394
2 G.729 239.1.1.7 16384 239.1.1.1 16396
2 Wideband 239.1.1.8 16384 239.1.1.1 16398
Note The lower destination port 16384 is assigned to the first multicast-enabled audio source ID, and the
subsequent ports will be assigned to the subsequent multicast-enabled audio sources.
Incrementation is triggered by a change in codec usage. When codec usage changes, a new IP address or
port number (depending on the incrementation selected) is assigned to the new codec type and is put into
use. The original codec keeps its IP address and port number. For example, as seen in Table 2, if your
baseline IP address and port number are 239.1.1.1 and 16384 for a G.711 mu-law codec and the codec
usage changes to G.729 (triggering an increment on the port number), the IP address and port number in
use changes, or increment, to 239.1.1.1 and 16386. If G.711 usage resumes, the IP address and port
number returns to 239.1.1.1 and 16384. If G.729 is in use again, the IP address and port goes back to
239.1.1.1 and 16386, and so forth.
It is important to configure a Cisco Unified CM port number and IP address that use a G.711 audio source
for Cisco Unified SRST multicast MOH. If Cisco Unified CM multicast MOH is also being used on
gateways that do not have Cisco Unified SRST and use a different codec, such as G.729, ensure that the
additional or incremental port number or IP address uses the same audio source as the Cisco Unified
SRST gateways and the required codec.
The MOH Server Configuration window is also where the multicast audio source for the MOH server is
configured. For Cisco Unified SRST multicast MOH, the Cisco Unified CM MOH server can use only
one audio source. An audio source is selected by inputting the audio source’s maximum number of hops.
The Max Hops configuration sets the length of the transmission of the audio source packets. Limiting
the number of hops is one way to stop audio packets from reaching the WAN and thus spoofing
Cisco Unified Communications Manager so Cisco Unified SRST can multicast MOH. If all of your
branches run Cisco Unified SRST, use a low number of hops to prevent audio source packets from
crossing the WAN. If your system configuration includes routers that do not run Cisco Unified SRST,
enter a high number of hops to allow source packets to cross the WAN. Use the ip multicast bounder
and access-list commands to keep resource packets from specific IP addresses from reaching the WAN.

B-21
Use this procedure to enable multicast and configure port numbers and IP addresses.
Step 1 Enable multicast MOH for Cisco Unified CM.

a. Choose Service > Media Resource > Music On Hold Server.
b. The MOH Server Configuration window appears.
c. Call up an existing MOH server by clicking Find and double-clicking the required MOH server.
d. Whether you are updating an existing MOH server or creating a new one, click Enable Multicast
Audio Sources on this MOH Server.
Step 2 Set the base IP address and port number.
In the MOH Server Configuration window, enter an IP address in the Base Multicast IP Address field
and enter a port number in the Base Multicast Port Number field. Ensure that the IP address and port
number use the required audio source and codec. See Table 2.
Step 3 Select whether Cisco Unified CM increments port numbers or IP addresses.
In the MOH Server Configuration window, in the Increment Multicast on field, choose Port Number if
you want port numbers to be incremented and the IP address to remain unchanged. Choose IP Address
if you want IP addresses to be incremented and the port number to remain unchanged.
• If all of your branches run Cisco Unified SRST and thus use G.711 for MOH, use either setting
because incrementation does not take place and a selection does not matter.
• If your system configuration includes routers that do not run Cisco Unified SRST and use a different
codec, select an incrementation method.
Note If your branches include routers that do not run Cisco Unified SRST and do use G.711, configure
separate audio sources: one for the routers that run Cisco Unified SRST and one for the routers
that do not.
Step 4 Enter a maximum number of hops.

In the MOH Server Configuration window, next to the Audio Source Name field, enter 1 in the Max Hops
field if all of your branches run Cisco Unified SRST. If your system configuration includes routers that
do not run Cisco Unified SRST, enter 16 in the Max Hops field.
Step 5 Use Cisco IOS commands to stop Cisco Unified CM signals from crossing the WAN and reaching
Cisco Unified SRST gateways.
If all of your branches run Cisco Unified SRST, skip this step. If your system configuration includes
routers that do not run Cisco Unified SRST and use a different codec, enter the following Cisco IOS
commands starting from global configuration mode on the central site router:
a. interface {serial | fastethernet} slot/port
Enters interface configuration mode, where slot is the slot number and port is the port number.
b. ip multicast boundary access-list-number
Configures an administratively scoped boundary, where access-list-number is a number from 1 to 99
that identifies an access list that controls the range of group addresses affected by the boundary.
c. exit
Exits interface configuration mode.

B-22
d. access-list access-list-number deny ip-address

Configures the access list mechanism for filtering frames by IP address. For the ip-address
argument, enter the MOH IP address that you want to prevent from going over the WAN. Normally
this would be the base IP address entered in Step 2.
The following is an example configuration:
Router(config)# interface serial 0/0
Router(config-if)# ip multicast boundary 1
Router(config)# access-list 1 deny 239.1.1.1
Creating an MRG and an MRGL, Enabling MOH Multicast, and Configuring Gateways
The next task involves configuring individual gateways to use an MOH server that can transport the
required MOH audio source to their IP phones on hold. This is accomplished by creating a Media
Resource Group (MRG). An MRG references media resources, such as MOH servers. The MRG is then
added to a Media Resource Group List (MRGL), and the MRGL is added to the phone and gateway
configurations.
MRGs are created in the Media Resource Group Configuration window. MRGLs are created in the Media
Resource Group List Configuration window. Phones are configured in the Phone Configuration window.
Gateways are configured in the Gateway Configuration window.
Note The Gateway Configuration window for an H.323 gateway is similar for MGCP gateways.
Add MRGL to a gateway or IP phone configuration by adding the MRGL to a device pool configuration.
For further information about device pools, see Cisco Unified Communications Manager Administration
Guide. From the Cisco Unified Communications Manager documentation directory, click Maintain and
Operate Guides and select the required Cisco Unified CM version.

B-23
Use the following procedure to create an MRG and MRGL, to enable MOH multicast, and to configure
gateways.
Step 1 Create an MRG with a multicast MOH media resource.

a. Choose Service > Media Resource > Media Resource Group.
b. In the upper-right corner of the window, click the Add a New Media Resource Group link. The
Media Resource Group Configuration window appears.
c. Complete the Media Resource Group Name field.
d. Complete the Description field.
e. Select a media resource from the Available Media Resources pane.
This pane lists the media resources that can be chosen for an MRG and can include the following
media resource types:
– Conference bridges (CFB)
– Media termination points (MTP)
– Music-on-hold servers (MOH)
– Transcoders (XCODE)
– Annunciator (ANN)
Music-on-hold servers that are configured for multicast are labeled as (MOH) [Multicast].
f. Click the down arrow so that the selected media resource moves to the Selected Media Resources
pane.
g. Click Insert.
Step 2 Create an MRGL that contains the newly created MRG.
a. Choose Service > Media Resource > Media Resource Group List.
b. In the upper-right corner of the window, click the Add a New Media Resource Group List link.
The Media Resource Group List Configuration window appears.
c. Complete the Media Resource Group List Name field.
d. In the Available Media Resource Groups pane, select the MRG that you just created.
e. Add the MRG to the Selected Media Resource Groups pane by clicking the down arrow. After a
media resource group is added, its name moves to the Selected Media Resource Groups pane.
f. Click Insert.
Step 3 Add the MRGL to the required IP phones.
a. Choose Device > Phone to display the Find and List Phones window.
b. Click Find to display a list of phones.
c. Double-click the device name of the phone that you want to update.
d. Complete the Media Resource Group List field by choosing the required MRGL from the drop-down
menu.
e. Click Update.
Step 4 Add the MRGL to the required gateway.
a. Choose Device > Gateway to display the Find and List Gateways window.
b. Click Find to display a list of gateways.

B-24
c. Double-click the device name of the gateway that you want to update.
d. If the gateway is H.323, complete the Media Resource Group List field by choosing the required
MRGL from the drop-down menu.
e. Click Update.
Creating a Region for the MOH Server

To ensure that the MOH server uses G.711 for Cisco Unified SRST gateways, you must create a separate
region for the MOH server. For more information about codecs, see the “Codecs, Port Numbers, and IP
Addresses” section on page B-14. For information about regions, see Cisco Unified
Communications Manager Administration Guide. From the Cisco Unified Communications
Manager documentation directory, click Maintain and Operate Guides and select the required
Cisco Unified Communications Manager version.
Configure the Region Configuration window. If the Cisco Unified CM system uses G.711 only, all of the
central sites and their constituent branches for the MOH region must be set to G.711. If a Cisco Unified
CM system has a combination of branches that do and do not run Cisco Unified SRST multicast MOH
and the branches that do not run Cisco Unified SRST require a different codec for Cisco Unified
Communications Manager multicast MOH, they must be configured accordingly.
A Region Configuration window where the “MOH Server” region is configured to use the G.711 and
G.729 codecs might look like this:
• G.711 is used for Branch 1 because its gateway is configured to run Cisco Unified SRST multicast
MOH, which requires G.711.
• G.729 is used for Branch 2 because its gateway doe not run Cisco Unified SRST and it is configured
to use a port and IP address that use G.729.
• G.711 is configured for the central site and the MOH server region.
Use the following procedure to create a region for the MOH server.
Step 1 Create an MOH server region.

a. Choose System > Region.
b. In the upper-right corner of the window, click Add a New Region. The Region Configuration
window appears.
c. In the Region Name field, enter the name that you want to assign to the new region and click Insert.
d. If other regions were created, a list of regions appear. Use the drop-down list boxes to choose the
audio codec to use for calls between the new region and existing regions. The audio codec
determines the type of compression and the maximum amount of bandwidth that is allocated for
these calls.
e. In addition to other regions, the newly created region appears in the list. Use its drop-down box to
choose the codec for use within the new region.
f. Click Update.
Step 2 Create other regions as needed for different codecs.

B-25
Verifying Cisco Unified Communications Manager Multicast MOH

The Cisco Unified CM multicast MOH configuration must run correctly for Cisco Unified SRST
multicast MOH to work. Verification of Cisco Unified Communications Manager multicast MOH differs
for configurations using a WAN with multicast enabled and a WAN with multicast disabled.
You must verify that the Cisco Unified CM multicast MOH is provided through multicasting and not
unicasting. Because unicast MOH is enabled by default, it is easy to mistakenly conclude that multicast
MOH is working when it is not.
Step 1 Verify that Cisco Unified CM system’s multicast MOH is heard on a remote gateway.
a. If multicast is enabled on the WAN, make sure that the number of hops configured on the
Cisco Unified Communications Manager MOH server is sufficient to allow audio packets to reach
the remote site (see the “Enabling Multicast on the Cisco Unified Communications Manager MOH
Server and Configuring Port Numbers and IP Addresses” section on page B-20). Then call an IP
phone on a remote gateway, place the call on hold, and verify that MOH is heard.
b. If multicast is not enabled on the WAN, place an IP phone on the same subnet as the Cisco Unified
Communications Manager MOH server and verify that MOH can be heard. Because the IP phone
and the MOH server are on the same subnet, no multicast routing capabilities in the network are
required.
Step 2 Verify that the Cisco Unified CM system’s MOH is multicast, not unicast.
a. From Microsoft Windows, select Start > Programs > Administrative Tools > Performance.
b. In the Performance window, click the + (plus) icon located at the top of the right pane.
c. In the Add Counters window, select Cisco MOH Device.
d. In the Performance window, you can monitor the MOHMulticastResourceActive and
MOHUnicastResourceActive counters to check on multicast activity.
Configuring Cisco Unified SRST for Multicast MOH from an Audio File
Note Use the steps in this section only when you are using Microsoft Windows to run Cisco Unified
Communications Manager version 4.3 or below. Use the RTMT (Real-Time Monitoring Tool) in
Cisco Unified Communications Manager version 5.0 and later versions on the Linux operating system
to monitor MOH activity in Cisco Unified CM version. See Cisco Unified Communications
Serviceability System Guide, Release 4.0(1) for more information about RTMT.
Use the following procedures to configure Cisco Unified SRST for multicast MOH from an audio file.
• Enabling Multicast MOH on the Cisco Unified SRST Gateway, page B-27
• Verifying Basic Cisco Unified SRST Multicast MOH Streaming, page B-31
• Verifying Cisco Unified SRST MOH to PSTN, page B-32
• Verifying Cisco Unified SRST Multicast MOH to IP Phones, page B-36

B-26
Prerequisites
• The Cisco Unified SRST gateways must run Cisco IOS Release 12.2(15)ZJ2 or a later release.
• The flash memory in each of the Cisco Unified SRST gateways must have an MOH audio file. The
MOH file can be in .wav or .au file format, but must contain 8-bit 8-kHz data, such as an a-law or
mu-law data format. A known working MOH audio file (music-on-hold.au) is included in the
program .zip files that can be downloaded from http://www.cisco.com/cgi-bin/tablebuild.pl/ip-key.
Or the music-on-hold.au file can be downloaded from
http://www.cisco.com/cgi-bin/tablebuild.pl/ip-iostsp and copied to the flash memory on your
Cisco Unified SRST router.
Note The MOH file packaged with the SRST software is completely royalty free.
• For Cisco Unified CM versions 4.3 or earlier versions running on Windows, download MOH files
by copying one of the MOH files, such as SampleAudioSource.ULAW.wav, from C:\Program
Files\Cisco\MOH on Cisco Unified CM.
Note During the copying process, four files are added to each router’s flash automatically. One of
the files must use a mu-law format as indicated by the extension.ULAW.wav.
• You must configure a loopback interface and include its IP addresses in the Cisco Unified SRST
multicast MOH configuration. This configuration allows multicast MOH to be heard on POTS ports
on the gateway. The loopback interface does not have to bind to either H.323 or MGCP.
• Configure at least one ephone and directory number (DN), even if the gateway is not used for
Cisco Unified SRST. Cisco Unified SRST multicast MOH streaming never starts without an ephone
and directory number.
Enabling Multicast MOH on the Cisco Unified SRST Gateway

No multicast MOH routing configuration is required for Cisco Unified SRST gateways because each
Cisco Unified SRST gateway is configured to act as a host running an application that streams multicast
MOH packets from the network. The multicast moh command declares the Cisco Unified
Communications Manager multicast MOH address and port number and allows Cisco Unified SRST
gateways to route MOH from flash memory to up to four IP addresses. If no route IP addresses are
configured, the flash MOH is sent through the IP address configured in the Cisco Unified SRST
ip source-address command.
SUMMARY STEPS
1. ccm-manager music-on-hold
2. interface loopback number
3. ip address ip-address mask
4. exit
5. interface fastethernet slot/port
6. ip address ip-address mask
7. exit

B-27

11. max-dn max-directory-number
12. moh filename
13. multicasting-enabled
14. multicast moh multicast-address port port [route ip-address-list]
15. exit
DETAILED STEPS

Step 1 ccm-manager music-on-hold Enables the multicast MOH feature on a voice
gateway.
Example:
Router(config)# ccm-manager music-on-hold
Step 2 interface loopback number Configures an interface type and enters the
interface configuration mode.
Example: • number—Loopback interface number. The
Router(config)# interface loopback 1 range is from 0 to 2147483647.
Step 3 ip address ip-address mask Sets a primary IP address for an interface.
• ip-address—IP address.
Example: • mask—Mask for the associated IP subnet.
Router(config-if)# ip address 10.1.1.1 255.255.255.255
Step 4 exit Exits interface configuration mode.
Example:
Step 5 interface fastethernet slot/port (Optional if the route keyword is not used in the
multicast moh command. See Step 9 and
Step 13.) Configures an interface type and enters
Example:
Router(config)# interface fastethernet 0/0
interface configuration mode.
Step 6 ip address ip-address mask (Optional if the route keyword is not used in the
Step 13.) Sets a primary IP address for an
Example:
Router(config-if)# ip-address 172.21.51.143
interface.
255.255.255.192
Step 7 exit (Optional if the route keyword is not used in the
Step 13.) Exits interface configuration mode.
Example:
Example:

B-28

Step 9 ip source-address ip-address [port port] (Optional if the route keyword is not used in the
multicast moh command. See Step 13.) Enables a
router to receive messages from Cisco Unified IP
Example:
phones through the specified IP addresses and
172.21.51.143 port 2000 ports.
• ip-address—The pre-existing router IP
address, typically one of the addresses of the
Ethernet port of the router.
• port port—(Optional) The port to which the
gateway router connects to receive messages
from the Cisco Unified IP phones. The port
number range is from 2000 to 9999. The
default port number is 2000.
Step 10 max-ephones max-phones Configures the maximum number of
Cisco Unified IP phones that can be supported by
a router.
Example:
Router(config-cm-fallback)# max-ephones 1 • max-phones—Maximum number of Cisco IP
phones supported by the router. The
maximum number is platform-dependent. The
default is 0.
Step 11 max-dn max-directory-number Sets the maximum possible number of virtual
voice ports that can be supported by a router.
Example: • max-directory-number—Maximum number
Router(config-cm-fallback)# max-dn 1 of directory numbers or virtual voice ports
supported by the router. The maximum
possible number is platform-dependent. The
default is 0.
Step 12 moh filename Enables use of an MOH file.
• filename—Filename of the music file. The
Example: music file must reside in flash memory.
Router(config-cm-fallback)# moh music-on-hold.au
Step 13 multicasting-enabled Selects the multicast-enabled MOH audio source
in the User Hold MOH Audio Source field on the
Phone Configuration page in Cisco Unified CM
Administration GUI.

B-29

Step 14 multicast moh multicast-address port port [route Enables multicast of MOH from a branch office
ip-address-list] flash MOH file to IP phones in the branch office.
• multicast-address and port port—Declares
Example: the IP address and port number of MOH
Router(config-cm-fallback)# multicast moh 239.1.1.1 packets that are to be multicast. The multicast
port 16386 route 239.1.1.2 239.1.1.3 239.1.1.4
IP address and port must match the IP address
239.1.1.5
and the port number that Cisco Unified CM is
configured to use for multicast MOH. If you
are using different codecs for MOH, these
might not be the base IP address and port,
instead an incremented IP address or port
number. See the “Configuring the MOH
Audio Source to Enable Multicasting” section
on page B-19. If you have multiple audio
sources configured on Cisco Unified CM,
ensure that you are using the audio sources’s
correct IP address and port number.
• route—(Optional) List of explicit router
interfaces for the IP multicast packets.
• ip-address-list—(Optional) List of up to four
explicit routes for multicast MOH. The
default is that the MOH multicast stream is
automatically output on the interfaces that
correspond to the address that was configured
with the ip source-address command.
Example:

B-30
Verifying Basic Cisco Unified SRST Multicast MOH Streaming

Use the following procedure to verify that multicast MOH packets are configured with the multicast
moh command.
SUMMARY STEPS
1. debug ephone moh

2. show interfaces fastethernet
3. show ephone summary
DETALED STEPS

Step 1 debug ephone moh This command sets debugging for MOH. You can
use this command to show that the
Cisco Unified SRST gateway is multicasting MOH
Example:
Router# debug ephone moh
out of Loopback 0 and Fast Ethernet 0/0.
!
MOH route If FastEthernet0/0 ETHERNET 172.21.51.143
via ARP
MOH route If Loopback0 46 172.21.51.98 via
172.21.51.98
!
Step 2 show interfaces fastethernet Use this command to confirm that the interface
output rates match one G.711 stream, which the
show interfaces fastethernet output displays as
Example:
Router# show interfaces fastethernet 0/0
50 packets/sec and 80 kbps or more.
!
30 second output rate 86000 bits/sec, 50 packets/sec
!
Step 3 show ephone summary Use this command to verify that the Cisco IOS
software was able to read the MOH audio file
successfully.
Example:
Router# show ephone summary
!
File music-on-hold.au type AU
Media_Payload_G.711Ulaw64k 160 bytes
!
The show ephone summary output should show a file type as either .au or .wav. If INVALID appears,
an error exists.
Router# show ephone summary
!
File music-on-hold.au type INVALID Media_Payload_G.711Ulaw64k 160 bytes651-
!

B-31
An invalid output might be caused by the order in which the Cisco Unified SRST configuration
commands are entered. Use the no call-manager-fallback command and reenter the multicast MOH
commands. Rebooting may also clear the error.
Verifying Cisco Unified SRST MOH to PSTN

Use the following procedure to verify Cisco Unified CM control of MOH (the WAN link is up) and that
multicast MOH packets transmit over a public switched telephone network (PSTN).
Note This feature does not apply when the Cisco Unified SRST router is in fallback mode.
SUMMARY STEPS
1. Verify that a PSTN caller hears MOH when placed on hold by an IP phone caller.
2. show ccm-manager music-on-hold
3. debug h245 asn
4. show call active voice
DETAILED STEPS

Step 1 Verify that a PSTN caller hears MOH when placed on hold by an
IP phone caller.
Use a Cisco Unified SRST gateway IP phone to call a PSTN
phone, and put the PSTN caller on hold. The PSTN caller should
hear MOH.
Step 2 show ccm-manager music-on-hold Use this command to verify that the MOH is
multicast if you are using Windows and Cisco
Unified CM version 4.3 or an earlier version.
Example:
Router# show ccm-manager music-on-hold
Note that the show ccm-manager
Current active multicast sessions : 1 music-on-hold command displays information
Multicast RTP port Packets Call Codec Incoming about PSTN connections on hold only. It does
Address number in/out id Interface not display information about multicast streams
======================================================= going to IP phones on hold. The following is an
239.1.1.1 16384 326/326 42 G.711ulaw Lo0
example of show ccm-manager music-on-hold
command output.
If the PSTN caller hears MOH, and the show
ccm-manager music-on-hold command
displays no active multicast streams, the MOH is
unicast. Confirm this by checking the MOH
performance counters as discussed in the
“Verifying Cisco Unified Communications
Manager Multicast MOH” section on
page B-26.

B-32

Step 3 debug h245 asn Use this command if H.323 is being used and no
multicast address appears in the show
ccm-manager music-on-hold command output
Example:
Router# debug h245 asn
to verify the H.323 handshaking between Cisco
*Mar 1 04:20:19.227: H245 MSC INCOMING PDU ::= Unified Communications Manager and the
value MultimediaSystemControlMessage ::= response : Cisco Unified SRST gateway. When a PSTN
openLogicalChannelAck : caller is placed on hold, Cisco Unified
{ Communications Manager sends an H.245
forwardLogicalChannelNumber 6
forwardMultiplexAckParameters
closeLogicalChannel, followed by an
h2250LogicalChannelAckParameters : openLogicalChannel. Verify that the final
{ openLogicalChannelAck from Cisco Unified
sessionID 1 Communications Manager to the Cisco Unified
mediaChannel unicastAddress : iPAddress : SRST gateway contains the expected multicast
{
network 'EF010101'H
IP address and port number. In the following
tsapIdentifier 16384 example, the IP address is EF010101 (239.1.1.1)
} and the port number is 16384.
mediaControlChannel unicastAddress : iPAddress :
{
network 'EF010101'H
tsapIdentifier 16385
}
}
}
Step 4 show call active voice Use this command with the debug h245 asn
command to further verify the H.323
handshaking between Cisco Unified
Example:
Router# show call active voice | include RemoteMedia
Communications Manager and the Cisco
RemoteMediaIPAddress=239.1.1.1 Unified SRST gateway.
RemoteMediaPort=16384
The IP address and port number displayed must
match the IP address and port number displayed
by the debug h245 asn command. If the
RemoteMediaIPAddress field displays 0.0.0.0,
you probably have encountered caveat
CSCdz00697. For more information, see the
Cisco Bug ToolKit and the “Restrictions for
Using Cisco Unified SRST Gateways as a
Multicast MOH Resource” section on
page B-12.
• If the PSTN caller hears tone on hold (TOH) instead of MOH, two problems are probable:
– Cisco Unified CM has failed to activate MOH and has used TOH as a fallback. To verify that
this is the case, see the “Verifying Cisco Unified Communications Manager Multicast MOH”
section on page B-26.
– Cisco Unified CM does not have the appropriate MOH resource available. Use the show
ccm-manager music-on-hold command to find out if the MOH resource is the problem.

B-33
Note The show ccm-manager music-on-hold command displays information about PSTN
connections on hold only. It does not display information about multicast streams going to
IP phones on hold.
Current active multicast sessions : 1

Multicast RTP port Packets Call Codec Incoming
Address number in/out id Interface
===================================================================
239.1.1.1 16384 326/326 42 G.711ulaw Lo0*
If no MOH streams are shown (that is, there are no rows of data beneath the columns),
Cisco Unified Communications Manager was not correctly configured to provide the
Cisco Unified SRST gateway with MOH. Configuration errors might include that the required
codec has not been enabled on Cisco Unified Communications Manager (check the service
parameters) and that no MRGL was assigned to the gateway, or, if one was assigned, it has
insufficient resources. Check Cisco Intrusion Detection System (Cisco IDS) Event Viewer for
error messages.
• If the POTS caller on hold does not hear a sound, Cisco Unified CM has successfully completed the
multicast MOH handshaking with the Cisco Unified SRST gateway, and the gateway is failing to
pick up the locally generated multicast RTP stream.
Use the show ccm-manager music-on-hold command to investigate.
Current active multicast sessions : 1

Multicast RTP port Packets Call Codec Incoming
Address number in/out id Interface
===================================================================
239.1.1.1 16384 326/326 42 G.711ulaw Lo0 *
– If no MOH streams are shown, Cisco Unified CM was not correctly set up to provide the
Cisco Unified SRST gateway with MOH. A typical error is that Cisco Unified Communications
Manager was not configured with an appropriate MOH resource. The configuration error might
be that the required codec has not been enabled on Cisco Unified CM (check the service
parameters) or that no MRGL was assigned to the gateway, or, if one is assigned, it has
insufficient resources. Check the IDS Event Viewer for error messages.
– Verify that the multicast address and RTP port number shown in the show ccm-manager
music-on-hold command output match the multicast-address and port arguments in the
moh multicast command configuration.
– Verify that the Packets in/out field shows a count that is incrementing. Repeat the show
ccm-manager music-on-hold command to verify that the Packets in/out counters are
incrementing.
– Verify that the codec field matches the codec type of the audio file stored in the
Cisco Unified SRST gateway’s flash memory. If another codec value besides G.711 mu-law or
G.711 a-law appears in the show ccm-manager music-on-hold command output, review the
Cisco Unified CM region for incorrect codec configuration. See the “Creating a Region for the
MOH Server” section on page B-25.

B-34
– The Incoming Interface field shows where the Cisco Unified SRST gateway is to receive the
multicast MOH packets. An interface must be listed and it must be one of the interfaces included
in the multicast moh command or the default IP source address, which is configured with the
ip source-address command.
For more information, see Step 9 in the “Enabling Multicast MOH on the Cisco Unified SRST
Gateway” section on page B-27.

B-35
Verifying Cisco Unified SRST Multicast MOH to IP Phones

To verify that Cisco Unified CM is signaling the IP phone to receive Cisco Unified SRST multicast
MOH correctly, perform the following steps.
Step 1 Verify that an IP phone caller hears MOH when placed on hold by an IP phone caller.
Use an IP phone to call a second IP phone, and put the second caller on hold. The second caller should
hear MOH.
Step 2 Check the MOHMulticastResourceActive and MOHUnicastResourceActive counters.
Use the Performance window to check the MOHMulticastResourceActive and
MOHUnicastResourceActive counters under the Cisco MOH Device performance object. See Step 2 in
the “Verifying Cisco Unified Communications Manager Multicast MOH” section on page B-26. For
Cisco Unified SRST multicasting MOH to work, the multicast counter must increment.
If no MOH is heard and the Cisco Unified SRST MOH signaling is multicasting, connect a sniffer to the
PC port on the back of IP phone. If the IP phone and Cisco Unified SRST gateway are connected to the
same subnet, multicast RTP packets must be detected at all times, even when the IP phone was not placed
on hold. If the IP phone and the Cisco Unified SRST gateway are not connected to the same subnet,
multicast RTP packets are detected only when the IP phone is placed on hold and sends an Internet Group
Management Protocol (IGMP) Join to the closest router.
Configuring Cisco Unified SRST for MOH from a Live Feed

The following sections describe the configuration tasks for Cisco Unified SRST MOH live feed:
• Prerequisites, page B-37
• Restrictions, page B-37
• Setting Up the Voice Port on the Cisco Unified SRST Gateway, page B-37
• Setting Up the Directory Numbers on the Cisco Unified SRST Gateway, page B-39
• Establishing the MOH Feed, page B-39
• Verifying Cisco Unified SRST MOH Live Feed, page B-41
To configure MOH from a live feed, establish a voice port and dial peer for the call and then create a
“dummy” phone or directory number. The dummy number allows for making and receiving calls, and
the number is not assigned to a physical phone. It is that number that the MOH system autodials to
establish the MOH feed.
The moh-live command allocates one of the virtual voice ports from the pool of virtual voice ports
created by the max-dn command. The virtual voice port places an outgoing call to the dummy number;
that is, the directory number specified in the moh-live command. The audio stream obtained from the
MOH call provides the music-on-hold audio stream.
We recommend that the interface for live-feed MOH is an analog E&M port because it requires the
minimum number of external components. Connect a line-level audio feed (standard audio jack) directly
to pins 3 and 6 of an E&M RJ-45 connector. The E&M WAN interface card (WIC) has a built-in audio
transformer that provides appropriate electrical isolation for the external audio source. (An audio

B-36
connection on an E&M port does not require loop current.) The signal immediate and auto-cut-through
commands disable E&M signaling on this voice port. A G.711 audio packet stream is generated by a
digital signal processor (DSP) on the E&M port.
In Cisco IOS Release 12.4(15)T and later releases, you can directly connect a live-feed source to an FXO
port if the signal loop-start live-feed command is configured on the voice port; otherwise, the port must
connect through an external third-party adapter to provide a battery feed. An external adapter must
supply normal telephone company (telco) battery voltage with the correct polarity to the tip and ring
leads of the FXO port and it must provide transformer-based isolation between the external audio source
and the tip and ring leads of the FXO port.
Music from a live feed is continuously fed into the MOH playout buffer instead of being read from a
flash file, so there is typically a 2-second delay. An outbound call to an MOH live-feed source is
attempted (or reattempted) every 30 seconds until the connection is made by the directory number that
was configured for MOH. If the live-feed source is shut down for any reason, the flash memory source
automatically activates.
A live-feed MOH connection is established as an automatically connected voice call that is made by the
Cisco Unified SRST MOH system itself or by an external source directly calling in to the live-feed MOH
port. An MOH call can be from or to the PSTN or can proceed via VoIP with voice activity detection
(VAD) disabled. The call is assumed to be an incoming call unless the out-call keyword is used with the
moh-live command during configuration.
The Cisco Unified SRST router uses the audio stream from the call as the source for the MOH stream,
displacing any audio stream that is available from a flash file. An example of an MOH stream received
over an incoming call is an external H.323-based server device that calls the directory number to deliver
an audio stream to the Cisco Unified SRST router.
Prerequisites
Cisco Unified SRST for multicast MOH, as described in the “Configuring Cisco Unified SRST for
Multicast MOH from an Audio File” section on page B-26, is not required for the MOH live-feed
configuration. However, MOH live feed is designed to work in conjunction with multicast MOH.
Restrictions
• An FXO port can be used for a live feed if the port is supplied with an external third-party adapter
to provide a battery feed.
• An FXS port cannot be used for a live feed.
• For a live feed from VoIP, VAD must be disabled.
• MOH is supplied to PSTN and VoIP G.711 calls. Some versions of Cisco Unified SRST provide
MOH to local phones. On Cisco Unified SRST that do not support MOH for local IP phones, callers
hear a repeating tone on hold for reassurance that they are still connected.
• Conditions may occur within your network that is caused by brief spikes of a higher CPU usage.
Small spikes in CPU usage can temporarily affect the quality of the MOH heard by parties connected
via TDM (FXO / PRI / S) interfaces.
Setting Up the Voice Port on the Cisco Unified SRST Gateway

Use the following procedure to activate MOH from a live feed and to set up and connect the physical
voice port.

B-37
SUMMARY STEPS
1. voice-port port
2. input gain decibels
3. auto-cut-through (E&M only)
4. operation 4-wire (E&M only)
5. signal immediate (E&M only)
6. no shutdown
7. exit
DETAILED STEPS

Step 1 voice-port port Enters voice-port configuration mode to set up the physical
voice port. To find the correct definition of the port
argument for your router, see Cisco IOS Survivable Remote
Example:
Router(config)# voice-port 1/1/0
Site Telephony Version 3.2 Command Reference.
Step 2 input gain decibels Specifies, in decibels, the amount of gain to be inserted at
the receiver side of the interface. Acceptable values are
integers from –6 to 14.
Example:
Router(config-voice-port)# input gain 0
Step 3 auto-cut-through (E&M ports only) Enables call completion when a PBX
does not provide an M-lead response. MOH requires that
you use this command with E&M ports.
Example:
Router(config-voiceport)# auto-cut-through
Step 4 operation 4-wire (E&M ports only) Selects the 4-wire cabling scheme. MOH
requires that you specify 4-wire operation with this
command for E&M ports.
Example:
Router(config-voiceport)# operation 4-wire
Step 5 signal immediate (E&M ports only) For E&M tie trunk interfaces, directs the
calling side to seize a line by going off-hook on its E-lead
and to send address information as DTMF digits.
Example:
Router(config-voiceport)# signal immediate
Step 6 no shutdown Activates the voice port.
Example:
Router(config-voiceport)# no shutdown
Step 7 exit Exits voice-port configuration mode.
Example:
Router(config-voiceport)# exit

B-38
Setting Up the Directory Numbers on the Cisco Unified SRST Gateway

After setting up the voice port, create a dial peer and give the voice port a directory number with the
destination-pattern command. The directory number is the number that the system uses to access the
MOH.
SUMMARY STEPS
1. dial-peer voice tag pots

2. destination-pattern string
3. port port
4. exit
DETAILED STEPS

Step 1 dial-peer voice tag pots Enters dial-peer configuration mode.
Example:
Router(config)# dial-peer voice 7777 pots
Step 2 destination-pattern string Specifies the directory number that the system uses to create
MOH. This command specifies either the prefix or the full
E.164 telephone number to be used for a dial peer.
Example:
Router(config-dial-peer)# destination-pattern
7777
Step 3 port port Associates the dial peer with the voice port that was
specified in the “Setting Up the Voice Port on the Cisco
Unified SRST Gateway” section on page B-37.
Example:
Router(config-dial-peer)# port 1/1/0
Example:
Establishing the MOH Feed

Use the following procedure to establish the MOH feed and connect the music source, such as a CD
player, to autodial the directory number.
SUMMARY STEPS
2. max-dn max-directory-number
3. multicast moh multicast-address port port [route ip-address-list]
4. moh-live dn-number calling-number out-call outcall-number
5. exit

B-39
DETAILED STEPS

Example:
Step 2 max-dn max-directory-number Sets the maximum possible number of virtual voice ports
that can be supported by a router.
Example: • max-directory-number—Maximum number of
Router(config-cm-fallback)# max-dn 1 directory numbers or virtual voice ports supported by
the router. The maximum possible number is
platform-dependent. The default is 0.
Step 3 multicast moh multicast-address port port Enables multicast of MOH from a branch office flash MOH
[route ip-address-list] file to IP phones in the branch office.
Note This command must be used to source live feed
Example: MOH to multicast Cisco Unified CM mode. It is not
Router(config-cm-fallback)# multicast moh required in strict SRST mode.
239.1.1.1 port 16386 route 239.1.1.2 239.1.1.3
239.1.1.4 239.1.1.5
• multicast-address and port port—Declares the IP
address and port number of MOH packets that are to be
multicast. The multicast IP address and port must
match the IP address and the port number that
Cisco Unified Communications Manager is configured
to use for multicast MOH. If you are using different
codecs for MOH, these might not be the base IP address
and port, but an incremented IP address or port number.
See the “Configuring the MOH Audio Source to Enable
Multicasting” section on page B-19. If you have
multiple audio sources configured on Cisco Unified
CM, ensure that you are using the audio sources’
correct IP address and port number.
• route ip-address-list—(Optional) Declares the IP
address or addresses from which the flash MOH
packets can be transmitted. A maximum of four IP
address entries are allowed. If a route keyword is not
configured, the Cisco Unified SRST system uses the
ip source-address command value configured for
Cisco Unified SRST.

B-40
Configurations Examples for Cisco Unified SRST Gateways

Step 4 moh-live dn-number calling-number out-call Specifies that this telephone number is to be used for an
outcall-number outgoing call that is to be the source for an MOH stream.
• dn-number calling-number—Sets the MOH telephone
Example: number. The calling-number argument is a sequence of
Router(config-cm-fallback)# moh-live dn-number digits that represent a telephone number.
3333 out-call 7777
• out-call outcall-number—Indicates that the router is
calling out for a live feed that is to be used for MOH and
specifies the number to be called. The outcall-number
argument is a sequence of digits that represent a
telephone number, typically of an E&M port.
The outcall keyword makes a connection to the local
router voice port that was specified in the “Setting Up
the Voice Port on the Cisco Unified SRST Gateway”
section on page B-37.
Example:
Verifying Cisco Unified SRST MOH Live Feed

To verify MOH live feed, use the debug ephone moh command and the other commands described in
the “Verifying Basic Cisco Unified SRST Multicast MOH Streaming” section on page B-31.
Configurations Examples for Cisco Unified SRST Gateways

This section provides the following configuration examples for Cisco Unified SRST gateways:
• MOH Routed to Two IP Addresses: Example, page B-41
• MOH Live Feed: Example, page B-42
MOH Routed to Two IP Addresses: Example

The following example declares the Cisco Unified CM multicast MOH IP address 239.1.1.1 and port
number 16384 and streams music-on-hold.au audio file packets out the interfaces that are configured
with the IP addresses 10.1.1.1 and 172.21.51.143:
ccm-manager music-on-hold
interface Loopback0
ip address 10.1.1.1. 255.255.255.255
ip address 172.21.51.143 255.255.255.192
max-ephones 1
max-dn 1

B-41
Feature Information for Cisco Unified SRST as a Multicast MOH Resource
multicast moh 239.1.1.1 port 16384 route 172.21.51.143 10.1.1.1
Note The multicast IP address and port must match the IP address and the port number that Cisco Unified CM
is configured to use for multicast MOH. If you are using different codecs for MOH, these might not be
the base IP address and port, but an incremented IP address or port number. See the “Configuring the
MOH Audio Source to Enable Multicasting” section on page B-19. If you have multiple audio sources
configured on Cisco Unified CM, ensure that you are using the audio source’s correct IP address and port
number.
MOH Live Feed: Example

The following example configures MOH from a live feed. Note that the dial peer references the E&M
port that was set with the voice-port command and that the dial peer number (7777) matches the outcall
number configured with the out-call keyword of the moh-live command.
voice-port 1/0/0
input gain 3
auto-cut-through
operation 4-wire
signal immediate
!
port 2/0/0
!
!
max-conferences 8
max-dn 1
moh-live dn-number 3333 out-call 7777
!
.
.
.
Feature Information for Cisco Unified SRST as a Multicast

MOH Resource
Table 3 lists the enhancements to the Cisco Unified SRST as a Mulitcast MOH Resource feature by
version.
To determine hardware and software compatibility, see the Cisco Unified CM Compatibility Information
page at the following URL:
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_device_support_tables_list.html
See also the Cisco Unified CM Documentation Roadmaps at the following URL:
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_documentation_roadmaps_list.htm.
Cisco Feature Navigator enables you to determine which Cisco IOS software images support a specific
software release, feature set, or platform. To access Cisco Feature Navigator, go to

B-42
Where to Go Next
Note Table 3 lists the Cisco Unified SRST version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified SRST software also support that feature.
Table 3 Feature Information for Cisco Unified SRST as a Multicast MOH Resource

Cisco Unified SRST as a Multicast MOH 3.0 The MOH-live feature was added.
Resource
Where to Go Next
For additional information, see the “Additional References” section on page 1-30 in the “Cisco Unified
SRST Feature Overview” section on page 1-1 chapter.

B-43
Where to Go Next

B-44

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What is Cisco Unified SRST?

What is Cisco Unified SRST?

What are the main features of Cisco Unified SRST?

What are the main features of Cisco Unified SRST?

Cisco Unified SCCP and SIP SRST

System Administrator Guide

November 30, 2018

Cisco Systems, Inc.

Cisco has more than 200 offices worldwide.

Obtain Documentation and Submit a Service Request ii-ii

Cisco Unified Survivable Remote Site Telephony Feature Roadmap iii-iii

Feature Roadmap iii-v

Information About New Features in Cisco Unified SRST iii-xi

New Features for Unified SRST Version 12.3 iii-xi

New Features for Unified SRST Version 12.2 iii-xi

New Features for Unified SRST Version 12.1 iii-xii

New Feature for Unified SRST Version 12.0 iii-xii

New Features for Cisco Unified SRST Version 11.0 iii-xii

New Features for Cisco Unified SRST Version 10.5 iii-xii

Cisco Unified SIP SRST System Administrator Guide

conference-pattern blocked iii-xx

New Features in Cisco Unified SRST Version 8.0 iii-xxix

New Features in Cisco Unified SRST Version 7.0/4.3 iii-xxix

New Features in Cisco Unified SRST Version 4.1 iii-xxx

Cisco Unified SIP SRST System Administrator Guide

New Features in Cisco SRST Version 3.2 iii-xxxiii

New Features in Cisco SRST Version 3.0 iii-xxxvi

New Features in Cisco SRST Version 2.1 iii-xl

Cisco Unified SIP SRST System Administrator Guide

Cisco Unified IP Phone Conference Station 7935 Support iii-xliv

Cisco Unified SRST Feature Overview 1-1

Cisco Unified SCCP SRST 1-1

MGCP Gateways and SRST 1-17

IPv6 Support for Unified SRST SIP IP Phones 1-18

Cisco Unified SIP SRST System Administrator Guide

Availability of Cisco IOS Software Images 1-26

Where to Go Next 1-30

Additional References 1-30

Platform and Memory Support 2-36

Cisco IOS Software Releases that Support Unified SRST 2-36

Cisco Unified SIP SRST System Administrator Guide

Configure Voice Hunt Groups on Unified SRST 2-69

Cisco Unified Enhanced Survivable Remote Site Telephony 3-87

Cisco Unified SIP SRST System Administrator Guide

Migration from Unified SRST Manager to Unified E-SRST 3-87

Setting Up the Network 4-115

Information About Setting Up the Network 4-116

How to Set Up the Network 4-116

Cisco Unified SIP SRST System Administrator Guide

Cisco Unified SIP SRST 4.1 5-127

Prerequisites for Cisco Unified SIP SRST 4.1 5-127

Restrictions for Cisco Unified SIP SRST 4.1 5-128

Information About Cisco Unified SIP SRST 4.1 5-128

Setting Up Cisco Unified IP Phones using SCCP 6-137

Information About Setting Up Cisco Unified IP Phones 6-137

Cisco Unified SIP SRST System Administrator Guide

How to Set Up Cisco Unified IP Phones 6-138

Setting Up Cisco Unified IP Phones using SIP 7-157

Prerequisites for Configuring the SIP Registrar 7-157