Cisco Unified SCCP and SIP SRST System Administrator Guide (All

Versions)
Last Modified: 2022-12-14

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 CONTENTS

CHAPTER 1 Cisco Unified Survivable Remote Site Telephony Feature Roadmap 1

Documentation Organization 1
Feature Roadmap 2
Information About New Features in Cisco Unified SRST 11
New Features for Cisco Unified SRST Version 14.2 11

New Features for Unified SRST Version 14.1 11

New Features for Unified SRST Version 12.7 11

New Features for Cisco Unified SRST Version 12.6 11

New Features for Cisco Unified SRST Version 12.3 11

New Features for Cisco Unified SRST Version 12.2 12

New Features for Cisco Unified SRST Version 12.1 12

New Feature for Cisco Unified SRST Version 12.0 12

New Features for Cisco Unified SRST Version 11.0 12

New Features for Cisco Unified SRST Version 10.5 12

Support for Cisco Unified DX650 SIP IP Phones 13

Support for Cisco Unified 78xx SIP IP Phones 13
New Features in Cisco Unified SRST Version 10.0 13

Cisco Jabber for Windows 13

Version Negotiation for Cisco Unified SIP IP Phones 14
New Features in Cisco Unified SRST Version 9.5 14

After-hour Pattern Blocking Support for Regular Expressions 14

Call Park Recall Enhancement 15
Park Monitor 15
Transfer-Pattern 18
Backward Compatibility 18
Transfer Max-Length 18

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Transfer-Pattern Blocked 19
Conference-Pattern Blocked 19
Configuring the Maximum Number of Digits for a Conference Call 20
Configuring Conference Blocking Options for Phones 21
Transfer-Pattern Blocked 23
Conference Transfer-Pattern 23
New Features in Cisco Unified SRST Version 9.1 23

Key Expansion Module Support for Cisco Unified SIP IP Phones 24

Enhancement in Speed-Dial Support 24
Voice Hunt Group Support 25
New Features in Cisco Unified SRST Version 9.0 25

Support for Cisco Unified 6901 and 6911 SIP IP Phones 25

Support for Cisco Unified 6921, 6941, 6945, and 6961 SIP IP Phones 26
Support for Cisco Unified 8941 and 8945 SIP IP Phones 26
Multiple Calls Per Line 26
Cisco Unified 8941 and 8945 SCCP IP Phones 26
Cisco Unified 6921, 6941, 6945, 6961, 8941, and 8945 SIP IP Phones 26
Voice and Fax Support on Cisco ATA-187 27
New Features in Cisco Unified SRST Version 8.8 27

Support for Cisco Unified 6945, 8941, and 8945 SCCP IP Phones 27
New Features in Cisco Unified SRST Version 8.0 27

New Features in Cisco Unified SRST Version 7.0/4.3 27

New Features in Cisco Unified SRST Version 4.2(1) 28
New Features in Cisco Unified SRST Version 4.1 28

New Features in Cisco Unified SRST Version 4.0 28

Additional Cisco Unified IP Phone Support 28

Cisco IP Communicator Support 28
Fax Pass-through using SCCP and ATAs Support 29
H.323 VoIP Call Preservation Enhancements for WAN Link Failures for SCCP Phones 29
Video Support 29
New Features in Cisco Unified SRST Version 3.4 29

Cisco SIP SRST 3.4 29

New Features in Cisco SRST Version 3.3 30

Secure SRST 30

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Cisco Unified IP Phone 7970G and Cisco Unified 7971G-GE Support 30

Enhancement to the show ephone Command 30
New Features in Cisco SRST Version 3.2 31

Enhancement to the alias Command 31

Enhancement to the cor Command 31
Enhancement to the pickup Command 31
Enhancement to the user-locale Command 31
Increased the Number of Cisco Unified IP Phones Supported on the Cisco 3845 31

MOH Live-Feed Support 31

No Timeout for Call Preservation 32
RFC 2833 DTMF Relay Support 32
Translation Profile Support 32
New Features in Cisco Unified SRST Version 3.1 32

Cisco Unified IP Phone 7920 Support 33

Cisco Unified IP Phone 7936 Support 33
New Features in Cisco SRST Version 3.0 33

Additional Language Options for IP Phone Display 33

Consultative Call Transfer and Forward Using H.450.2 and H.450.3 for SCCP Phones 33
Customized System Message for Cisco Unified IP Phones 34
Dual-Line Mode 34
E1 R2 Signaling Support 34
European Date Formats 35
Huntstop for Dual-Line Mode 36
Music On Hold for Multicast from Flash Files 36
Ringing Timeout Default 36
Secondary Dial Tone 36
Enhancement to the Show ephone Command 36
System Log Messages for Phone Registrations 36
Three-Party G.711 Ad Hoc Conferencing 37
Support for Cisco VG248 Analog Phone Gateway 1.2(1) and Higher Versions 37
New Features in Cisco SRST Version 2.1 37

Additional Language Options for IP Phone Display 37

Cisco Unified SRST Aggregation 38
Cisco ATA 186 and ATA 188 Support 38

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Cisco Unified IP Phone 7902G Support 38

Cisco Unified IP Phone 7905G Support 38
Cisco Unified IP Phone 7912G Support 39
Cisco Unified IP Phone Expansion Module 7914 Support 39
Enhancement to the Dial Plan-Pattern Command 39
New Features in Cisco SRST Version 2.02 39

Cisco Unified IP Phone Conference Station 7935 Support 39

Increase in Directory Numbers 39
Cisco Unity Voicemail Integration Using In-Band DTMF Signaling Across the PSTN and
BRI/PRI 40

CHAPTER 2 Cisco Unified SCCP and SIP SRST Feature Overview 41

Cisco Unified SRST Feature Overview 41

Cisco Unified SCCP SRST 42
Information About SCCP SRST 42
Prerequisites for Configuring Cisco Unified SCCP SRST 44
Installing Cisco Unified Communications Manager 44
Installing Cisco Unified SCCP SRST 45
Integrating Cisco Unified SCCP SRST with Cisco Unified Communications Manager 45
If You Have Cisco Communications Manager V3.3 or Later Versions 45
If You Have Cisco Unified Communications Manager Version Prior to V3.3 46
Restrictions for Configuring Cisco Unified SCCP SRST 46
Cisco Unified SIP SRST 49
Information About SIP SRST 49
Prerequisites for Configuring Cisco Unified SIP SRST 50
Restrictions for Configuring Cisco Unified SIP SRST 50
Interface Support for Cisco Unified Communications Manager Express and Cisco Unified SRST 55
MGCP Gateways and SRST 55
IPv6 Support for Unified SRST SIP IP Phones 56
IPv6 Support for Unified SRST SIP IP Phones 56
Feature Support for IPv6 in Unified SRST SIP IP Phones 56
Restrictions 56
Configure IPv6 Pools for SIP IP Phones 57
Examples for Configuring IPv6 Pools for SIP IP Phones 60

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Support for Cisco Unified IP Phones and Platforms 61

Support for Cisco Unified IP Phones and Platforms 61
Finding Cisco IOS Software Releases That Support Cisco Unified SRST 62
Cisco Unified IP Phone Support 62
Platform and Memory Support 62
Platform and Memory Support 62
Determining Platform Support Through Cisco Feature Navigator 62
Availability of Cisco IOS Software Images 62
Cisco Unified Communications Manager Compatibility 63
Signal Support 63
Language Support 63
Switch Support 63
Multicast Music On Hold 64
Multicast Music On Hold 64
Configure Multicast Music On Hold for Unified SRST 64
Where to Go Next 66
Additional References 67
Related Documents 67
Standards 68
MIBs 69
RFCs 69
Technical Assistance 69
Obtaining Documentation, Obtaining Support, and Security Guidelines 69

CHAPTER 3 Cisco Unified SIP SRST on Cisco 4000 Series Integrated Services Router 71
Overview 71
Platform and Memory Support 72
Cisco IOS Software Releases that Support Unified SRST 72
Install Cisco IOS XE Software 72
Feature Support 74
Restrictions of Unified SRST on Cisco 4000 Series Integrated Services Routers 75
Unified IP Phone Support 76
Cisco Jabber with Unified SRST 76
Cisco Unified Communications Manager Compatibility 76

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Installing Cisco Unified Communications Manager 76

Integrating Cisco Unified SIP SRST with Cisco Unified Communications Manager 76
Supported PSTN Trunk Connectivity 77
Language Support 77
Switch Support 77
Interface Support for Unified SRST 78
Simple Network Management Protocol (SNMP) Support for Unified SRST 78
Licensing 78
Cisco Smart Licensing for Unified SRST 78
Smart License Operation 79
Cisco IOS XE Everest 16.5.1 Release to Cisco IOS XE Fuji 16.9.1 Release 79
Cisco IOS XE Gibraltar 16.10.1 Release Onwards 79
Cisco IOS XE Everest 16.5.1 Release to Cisco IOS XE Amsterdam 17.3.1a Release 79
Cisco IOS XE Gibraltar 16.12.1 Release to Cisco IOS XE Amsterdam 17.3.1a Release 80
Cisco IOS XE Amsterdam 17.3.2 and Cisco IOS XE Bengaluru 17.4.1a Release Onwards 80
Configure SIP Registrar Functionality for SIP Phones on Unified SRST 81
Configure Backup Registrar Service to SIP Phones 83
Configure Backup Registrar Service to SIP Phones (Using Optional Commands) 86
Verify SIP Registrar Configuration 89
Verify Proxy Dial-Peer Configuration 90
Unified SRST, Unified E-SRST, and Unified Secure SRST Password Policy 93
Guidelines for Password Configuration and Encryption 93
Deprecation of CLI commands 94
Removal of Passwords and Keys from Logs 94
Toll Fraud Prevention for SIP Line Side on Unified SRST 96
Configuration Recommendations for Toll Fraud Prevention on Unified SRST 96
Upgrade Considerations 98
Configure Toll Fraud Prevention 98
Configure IP Address Trusted Authentication for Incoming VoIP Calls 98
Add Valid IP Addresses For Incoming VoIP Calls 100
Troubleshooting Tips for Toll Fraud Prevention 101
VRF Support for Unified SRST 102
Information About VRF Support 102
Design Recommendations for VRF 102

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Configuration Examples for VRF 103

Configure Virtual Routing and Forwarding (VRF) for Unified SRST 104
IPv6 Support for Unified SRST SIP IP Phones 106
Feature Support for IPv6 in Unified SRST SIP IP Phones 106
Restrictions 106
Configure IPv6 Pools for SIP IP Phones 107
Configure Unified SRST on Cisco 4000 Series Integrated Services Platform 110
Configure Voice Hunt Groups on Unified SRST 114
Configure Feature Support on Unified SIP SRST 117
Configure SIP-to-SIP Call Forwarding 117
Configure Call Blocking Based on Time of Day, Day of Week, or Date 119
Verification 122
SIP Call Hold and Resume 122
Configure Music On Hold for Unified SRST 122
Enabling KPML for SIP Phones 123
Disabling SIP Supplementary Services for Call Forward and Call Transfer 124
Configuring idle Prompt Status for SIP Phones 126
Examples 127
Example for Configuring Unified SIP SRST on Cisco 4000 Series Integrated Services Routers 127
Example for Configuring Voice Hunt Groups in Unified SIP SRST 127
Examples for Configuring IPv6 Pools for SIP IP Phones 128
Example for Configuring Call Blocking Based on Time of Day, Day of Week, or Date 128
Example for Configuring Music On Hold for Unified SIP SRST 129
Example for Configuring SIP-to-SIP Call Forwarding on Unified SRST 129
Example for Configuring idle Prompt Status for SIP Phones 129
Example for Disabling SIP Supplementary Services for Call Forward and Call Transfer 129

CHAPTER 4 Enhanced SRST 131

Migration from Cisco Unified SRST Manager to Unified E-SRST 131

Benefits 131
Restrictions 132
Restrictions for Unified E-SRST, Release 12.2 132

Support for Cisco Unified IP Phones and Platforms 132

Licensing 133

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Cisco Smart Licensing for Unified E-SRST 133

Smart License Operation 134
Cisco IOS XE Everest 16.5.1 Release to Cisco IOS XE Fuji 16.9.1 Release 134
Cisco IOS XE Gibraltar 16.10.1 Release Onwards 134
Cisco IOS XE Everest 16.5.1 Release to Cisco IOS XE Amsterdam 17.3.1a Release 134
Cisco IOS XE Gibraltar 16.12.1 Release to Cisco IOS XE Amsterdam 17.3.1a Release 135
Cisco IOS XE Amsterdam 17.3.2 and Cisco IOS XE Bengaluru 17.4.1a Release Onwards 135
Toll Fraud Prevention for SIP Line Side on Unified E-SRST 136
Unified E-SRST with Support for Voice Hunt Group 136
Support for B-ACD in Unified E-SRST 137
Recommendations for Configuring Voice Hunt Group on Unified E-SRST 137
SIP: Configure Unified E-SRST 138
Restrictions 139
Enable the E-SRST Mode 139
Configure SIP shared-line 140
Configure BLF 141
Enable a SIP Directory Number to Be Watched 141
Enable BLF on a Voice Register Pool 142
Example: ESRST Mode 143
Example: Configuring Shared Line 143
Example: Configuring BLF 143
Configure Unified E-SRST 143
Configure Voice Hunt Groups on Unified E-SRST 147
Example for Configuring Unified E-SRST with Voice Hunt Group Enhancements 149
Example for Configuring B-ACD with Unified E-SRST 150
Example for Configuring Shared Line with Voice Hunt Group on Unified E-SRST 151
SCCP: Configure Unified E-SRST 153
Configure Mixed Shared Lines with SCCP Phones 157
Configure BLF for SCCP Phones 158
Enable an SCCP Directory Number to Be Watched 158
Enable BLF on an Ephone 159
Configure Digest Credentials on Cisco Unified Communications Manager 159
Configure Digest Credentials on Unified E-SRST for SIP 160
Example: Configuring Digest Credentials on ESRST 161

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Configure Digest Credentials on Unified E-SRST for SCCP 161

CHAPTER 5 Setting Up the Network 163

Information About Setting Up the Network 163

How to Set Up the Network 163
Enabling Cisco Unified SRST on an MGCP Gateway 163
Configuring Cisco Unified SRST on an MGCP Gateway Before Cisco IOS Release 12.3(14)T 164
Configuring SRST on an MGCP Gateway Using Cisco IOS Release 12.3(14)T or Later
Releases 165
Configuration Example of Enabling SRST on a MGCP Gateway using Cisco IOS Release
12.3(14)T 167
Configuring DHCP for Cisco Unified SRST Phones 168
Defining a Single DHCP IP Address Pool 169
Defining a Separate DHCP IP Address Pool for Each Cisco Unified IP Phone 169
Defining the DHCP Relay Server 170
Specifying Keepalive Intervals 171

CHAPTER 6 Cisco Unified SIP SRST 173

Prerequisites for Cisco Unified SIP SRST 4.1 173

Restrictions for Cisco Unified SIP SRST 4.1 174

Information About Cisco Unified SIP SRST 4.1 174

Out-of-Dialog REFER 174

Digit Collection on SIP Phones 174
KPML Digit Collection 175
SIP Dial Plans 175
Caller ID Display 175
Disabling SIP Supplementary Services for Call Forward and Call Transfer 176
Idle Prompt Status 177
Enhanced 911 Services 177
How to Configure Cisco Unified SIP SRST 4.1 Features 178
Enabling KPML for SIP Phones 178
Disabling SIP Supplementary Services for Call Forward and Call Transfer 179
Configuring Idle Prompt Status for SIP Phones 180

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CHAPTER 7 Setting Up Cisco Unified IP Phones using SCCP 183

Information About Setting Up Cisco Unified IP Phones 183
How to Set Up Cisco Unified IP Phones 183
Configuring Cisco Unified SRST to Support Phone Functions 184
Configuring Cisco Unified 8941 and 8945 SCCP IP Phones 186
Verifying That Cisco Unified SRST Is Enabled 187
Configuring IP Phone Clock, Date, and Time Formats 187
Configuring IP Phone Language Display 189
Configuring Customized System Messages for Cisco Unified IP Phones 190
Configuring a Secondary Dial Tone 192
Configuring Dual-Line Phones 192
Configuring Eight Calls per Button (Octo-Line) 194
Configuring the Maximum Number of Calls 196
Troubleshooting 197
How to Set Up Cisco IP Communicator for Cisco Unified SRST 198
Prerequisites 198
Verifying Cisco IP Communicator 199
Troubleshooting Cisco IP Communicator 199
Where to Go Next 199

CHAPTER 8 Setting Up Cisco Unified IP Phones using SIP 201

Prerequisites for Configuring the SIP Registrar 201

Restrictions for Configuring the SIP Registrar 201
Information About Configuring the SIP Registrar 201
How to Configure the SIP Registrar 202
Configuring the SIP Registrar 202
Configuring Backup Registrar Service to SIP Phones 203
Configuring Backup Registrar Service to SIP Phone (Using Optional Commands) 207
Verifying SIP Registrar Configuration 210
Verifying Proxy Dial-Peer Configuration 211

CHAPTER 9 Configuring Call Handling 215

Prerequisites for Configuring SIP SRST Features Using Back-to-Back User Agent Mode 215

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Restrictions for Configuring SIP SRST Features Using Back-to-Back User Agent Mode 216
Information About Configuring SCCP SRST Call Handling 216
H.323 VoIP Call Preservation Enhancements for WAN Link Failures 216
Toll Fraud Prevention 216
Information About Configuring SIP SRST Features Using Back-to-Back User Agent Mode 217
Cisco Unified SIP SRST and Cisco SIP Cisco Unified Communications Manager Express Feature
Crossover 217
How to Configure Cisco Unified SCCP SRST 220
Configuring Incoming Calls 220
Configuring Call Forwarding During a Busy Signal or No Answer 220
Configuring Call Rerouting 222
Configuring Call Pickup 222
Configuring Consultative Transfer 224
Configuring Transfer Digit Collection Method 225
Configuring Global Prefixes 226
Enabling Digit Translation Rules 228
Enabling Translation Profiles 229
Verifying Translation Profiles 232
Configuring Dial-Peer and Channel Hunting 233
Configuring Busy Timeout 234
Configuring the Ringing Timeout Default 234
Configuring Outgoing Calls 235
Configuring Local and Remote Call Transfer 235
Enabling Consultative Call Transfer and Forward Using H.450.2 and H.450.3 with Cisco Unified
SRST 3.0 236
Enabling Analog Transfer Using Hookflash and the H.450.2 Standard with Cisco Unified SRST 3.0
or Earlier 239
Configuring Trunk Access Codes 243
Configuring Interdigit Timeout Values 244
Configuring Class of Restriction 245
Call Blocking (Toll Bar) Based on Time of Day and Day of Week or Date 248
How to Configure Cisco Unified SIP SRST 250
Configuring SIP Phone Features 250
Configuring SIP-to-SIP Call Forwarding 252
Configuring Call Blocking Based on Time of Day, Day of Week, or Date 255

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Verification 258
SIP Call Hold and Resume 258
How to Configure Optional Features 260
Enabling Three-Party G.711 Ad Hoc Conferencing 260
Defining XML API Schema 261
Configuration Examples for Call Handling 262
Example: Monitoring the Status of Key Expansion Modules 262
Example: Configuring Voice Hunt Groups in Cisco Unified SIP SRST 263
Where to Go Next 263

CHAPTER 10 Configuring Secure SRST for SCCP and SIP 265

Prerequisites for Configuring Secure SRST 265

Restrictions for Configuring Secure SRST 266
Information About Configuring Secure SRST 268
Benefits of Secure SRST 268
Secure SIP SRST Support on Cisco 4000 Series Integrated Services Router 268
Secure Music On Hold for Unified Secure SRST (SIP) 269
Secure SCCP SRST on Cisco 4000 Series Integrated Services Router 269
Secure SCCP SRST for Analog Voice Gateways 269
Secure Music On Hold for Secure Unified SRST (SCCP) 270
Three-way Software Conferencing for Secure SCCP, Unified SRST Release 12.3 270

Feature Support for Secure SRST (SCCP), Unified SRST Release 12.3 271

Cisco IP Phones Clear-Text Fallback During Non-Secure SRST 272

Signaling Security on Unified SRST - TLS 272
SRST Routers and the TLS Protocol 272
TLS Cipher Support for Secure SRST 12.6 and Later Releases 274
SIP OAuth Client Registration for Unified Secure SRST 276
Feature Characteristics 277
Restrictions 277
Configure SIP OAuth-based Listener Port 277
Retrieve SIP OAuth Keys from CUCM 278
Enable SIP OAuth-based Registration 279
Verify SIP OAuth for Secure SRST 280
SHA2-Cipher-Only Mode for Unified Secure SRST 284

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Benefits 285
Configure SHA2 Cipher Suite with TLS 286
Media Security on Unified SRST - SRTP 286
Establishment of Secure Cisco Unified SRST to the Cisco Unified IP Phone 287
Secure SRST Authentication and Encryption 289
How to Configure Secure Unified SRST 290
Preparing the Cisco Unified SRST Router for Secure Communication 290
Configuring Cisco Unified Communications Manager to the Secure Cisco Unified SRST Router 306
Enabling SRST Mode on the Secure Cisco Unified SRST Router 310
Configuring Secure SCCP SRST 311
Configuring Secure SIP Call Signaling and SRTP Media with Cisco SRST 324
Configuration Example for SIP OAuth 336
Configuration Examples for SHA-2 Cipher Suites 338
Additional References 338
Related Documents 338
Standards 338
MIBs 339
RFCs 339
Technical Assistance 339
Command Reference 339
Feature Information for Secure SCCP and SIP SRST 340

CHAPTER 11 Configuring SIP Trunking on Unified SRST 341

Unified SRST and Unified Border Element Co-location 341
Configuration Recommendations for Unified SRST and Unified Border Element Co-location 343
Restrictions 346
Examples 346
Feature Information for Configuring SIP Trunking on Cisco Unified SRST 356

CHAPTER 12 Integrating Voice Mail with Cisco Unified SRST 359

Information About Integrating Voicemail with Cisco Unified SRST 359
How to Integrate Voicemail with Cisco Unified SCCP and SIP SRST 360
Configuring Direct Access to Voicemail 361
Examples 364

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Configuring Message Buttons 364

Examples 366
Redirecting to Cisco Unified Communications Manager Gateway 366
Configuring Call Forwarding to Voicemail 367
Call Routing Instructions Using DTMF Digit Patterns 367
Prerequisites 369
Configuring Call Forwarding to Voicemail 369
Examples 370
Configuring Message Waiting Indication (Cisco Unified SRST Routers) 371
Configuring Message Waiting Indication (SIP Phones in SRST Mode) 373
Configuration Examples for Unified SRST 374
Configuring Local Voicemail System (FXO and FXS): Example 374
Configuring Central Location Voicemail System (FXO and FXS): Example 375
Configuring Voicemail Access over FXO and FXS: Example 376
Configuring Voicemail Access over BRI and PRI: Example 376
Message Waiting Indication for SIP SRST: Example 377
How to Configure DTMF Relay for SIP Applications and Voicemail 377
DTMF Relay Using SIP RFC 2833 377

Troubleshooting Tips 379

DTMF Relay Using SIP Notify (Nonstandard) 379
Troubleshooting Tips 380

CHAPTER 13 Setting Video Parameters 383

Prerequisites for Setting Video Parameters 383

Restrictions for Setting Video Parameters 384
Information About Setting Video Parameters 384
Matching Endpoint Capabilities 385
Retrieving Video Codec Information 385
Call Fallback to and Audio-Only Endpoint 385
Call Setup for Video Endpoints 385
Call Setup Between Two Local SCCP Endpoints 386
Call Setup Between SCCP and H.323 Endpoints 386
Call Setup Between Two SCCP Endpoints Across an H.323 Network 386
Flow of the RTP Video Stream 386

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How to Set Video Parameters for Cisco Unified SRST 387

Configuring Slow Connect Procedures 387
Verifying Cisco Unified SRST 388
Setting Video Parameters for Cisco Unified SRST 393
Troubleshooting Video for Cisco Unified SRST 394

CHAPTER 14 Monitoring and Maintaining Cisco Unified SRST 397

Monitoring and Maintaining Cisco Unified SRST 397

APPENDIX A Appendix A: Configuring Cisco Unified SIP SRST Features Using Redirect Mode 399
Prerequisites for Cisco Unified SIP SRST Features Using Redirect Mode 399
Restrictions for Cisco Unified SIP SRST Features Using Redirect Mode 399
Information About Cisco Unified SIP SRST Features Using Redirect Mode 400
How to Configure Cisco Unified SIP SRST Features Using Redirect Mode 400
Configuring Call Redirect Enhancements to Support Calls Between SIP IP Phones for Cisco Unified
SIP SRST 400
Configuring Audio and Video Codecs at the Dial Peer Level 400
Configuring Call Redirect Enhancements to Support Calls On a Specific VoIP Dial Peer 401
Configuring Sending 300 Multiple Choice Support 402
Configuration Examples for Cisco Unified SIP SRST Features Using Redirect Mode 404
Cisco Unified SIP SRST: Example 404

APPENDIX B Appendix B: Integrating Cisco Unified Communications Manager and Cisco Unified SRST to Use Cisco
Unified SRST as a Multicast MOH Resource 407
Prerequisites for Using Cisco Unified SRST Gateways as a Multicast MOH Resource 407
Restrictions for Using Cisco Unified SRST Gateways as a Multicast MOH Resource 408
Information About Using Cisco Unified SRST Gateways as a Multicast MOH Resource 408
Cisco Unified SRST Gateways and Cisco Unified Communications Manager 408
Codecs, Port Numbers, and IP Addresses 410
Multicast MOH Transmission 411
MOH from a Live Feed 411
MOH from Flash Files 411
How to Use Cisco Unified SRST Gateways as a Multicast MOH Resource 412
Configuring Cisco Unified Communications Manager for Cisco Unified SRST Multicast MOH 412

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Configuring the MOH Audio Source to Enable Multicasting 414

Enabling Multicast on the Cisco Unified Communications Manager MOH Server and Configuring
Port Numbers and IP Addresses 415
Creating an MRG and an MRGL, Enabling MOH Multicast, and Configuring Gateways 418
Creating a Region for the MOH Server 418
Verifying Cisco Unified Communications Manager Multicast MOH 419
Configuring Cisco Unified SRST for Multicast MOH from an Audio File 420
Prerequisites 420
Enabling Multicast MOH on the Cisco Unified SRST Gateway 421
Verifying Basic Cisco Unified SRST Multicast MOH Streaming 423
Verifying Cisco Unified SRST MOH to PSTN 424
Verifying Cisco Unified SRST Multicast MOH to IP Phones 426
Troubleshooting Tips 426
Configuring Cisco Unified SRST for MOH from a Live Feed 426
Prerequisites 427
Restrictions 427
Setting Up the Voice Port on the Cisco Unified SRST Gateway 428
Setting Up the Directory Numbers on the Cisco Unified SRST Gateway 429
Establishing the MOH Feed 429
Verifying Cisco Unified SRST MOH Live Feed 431
Configurations Examples for Cisco Unified SRST Gateways 431
MOH Routed to Two IP Addresses: Example 431
MOH Live Feed: Example 432
Feature Information for Cisco Unified SRST as a Multicast MOH Resource 432
Where to Go Next 433

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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.
• Documentation Organization, on page 1
• Feature Roadmap, on page 2
• Information About New Features in Cisco Unified SRST, on page 11

Documentation Organization
This document consists of the following chapters or appendixes as shown in the following table .

Chapter or Appendix Description

Cisco Unified SCCP and SIP SRST Feature Gives a brief description of Cisco Unified SRST and provides
Overview, on page 41 information on the 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, on page 163 Describes how to set up a Cisco Unified SRST system to
communicate with your network.

Enhanced SRST, on page 131 Describes how to configure the Cisco Unified Enhanced SRST
feature in your network.

Cisco Unified SIP SRST, on page 173 Describes the features for Cisco Unified SIP SRST Version
4.1 and provides the associated configuration procedures.

Setting Up Cisco Unified IP Phones using Describes how to set up the basic Cisco Unified SRST phone
SCCP, on page 183 configuration.

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Feature Roadmap

Chapter or Appendix Description

Setting Up Cisco Unified IP Phones using Describes features available in Version 3.0 that are also
SIP, on page 201 necessary for Version 3.4. 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, on page 215 Describes how to configure incoming and outgoing calls.

Configuring Secure SRST for SCCP and SIP, Describes the Secure SRST security functionality to the Cisco
on page 265 Unified SRST.

Integrating Voice Mail with Cisco Unified Describes how to set up voicemail.
SRST, on page 359

Setting Video Parameters, on page 383 Describes how to set up video parameters.

Monitoring and Maintaining Cisco Unified Provides a list of useful show commands for monitoring and
SRST, on page 397 maintaining Cisco Unified SRST.

Appendix A: Configuring Cisco Unified SIP Describes features using redirect mode, which applies to
SRST Features Using Redirect Mode, on page version 3.0 only.
399

Appendix B: Integrating Cisco Unified Describes how to configure Cisco Unified CM and Cisco
Communications Manager and Cisco Unified Unified SRST to enable multicast music-on-hold (MOH).
SRST to Use Cisco Unified SRST as a
Multicast MOH Resource, on page 407

Feature Roadmap
The following table provides a feature history summary of Cisco Unified SRST features.

Cisco Unified Cisco IOS Release Enhancements or Modifications

SRST

Version 14.3 Cisco IOS XE YANG model enhancements for Unified SRST:
Dublin 17.10.1a
Programmability Guide for Cisco IOS XE Unified Communications
VoIP Products
https://www.cisco.com/c/en/us/td/docs/routers/sdwan/command/
sdwan-cr-book.html

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Feature Roadmap

Cisco Unified Cisco IOS Release Enhancements or Modifications

SRST

Version 14.3 Cisco IOS XE Webex Managed Gateway Command Reference

Cupertino 17.9.1a
Introduced the following commands as part of Cisco Webex Calling
Branch Survivability:
• mode webex-sgw
• voice register webex-sgw sync
• show voice register webex-sgw users

Modified the following command as part of Cisco Webex Calling

Branch Survivability:
• id (voice register pool) modified to include phone-number
e164-number and extension-number extension-number

Version 14.3 Cisco IOS XE CDR Accounting Overview

Cupertino 17.9.1a
Configuring File Accounting
gw-accounting

Version 14.2 Cisco IOS XE SHA2-Cipher-Only Mode for Unified Secure SRST, on page 284
Cupertino 17.8.1a

Version 14.2 Cisco IOS XE SIP OAuth Client Registration for Unified Secure SRST, on page 276
Cupertino 17.8.1a
Version 14.1 Cisco IOS XE Programmability Guide for Cisco IOS XE Unified Communications
Bengaluru 17.6.1a VoIP Products

Version 14.1 Cisco IOS XE • Support for Unified SRST on Cisco 1100 Integrated Services
Bengaluru 17.5.1a Router
Release
• Support for Unified SRST and E-SRST on Cisco 8200L Catalyst
Series Edge Platforms

Version 14.1 Cisco IOS XE • Support for Unified SRST and E-SRST on Cisco 8200 Catalyst
Bengaluru 17.4.1a Series Edge Platforms
Release
• Smart Licensing Using Policy—Licensing
• Smart Licensing Using Policy—Licensing

Version 14.1 Cisco IOS XE • Support for Unified SRST and E-SRST on Cisco 8300 Catalyst
Amsterdam 17.3.2 Series Edge Platforms
Release
• Smart Licensing Using Policy—Licensing
• Smart Licensing Using Policy—Licensing

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Feature Roadmap

Cisco Unified Cisco IOS Release Enhancements or Modifications

SRST

Version 12.8 Cisco IOS XE • Cisco Jabber with Unified SRST

Amsterdam 17.2.1r
• VRF Support for Unified SRST
• Support for YANG Models in Unified SRST

Version 12.7 Cisco IOS XE Support for maximum number of devices in Cisco 4451 and 4461
Amsterdam Integrated Services Routers was increased from 1500 to 2000
17.1.1

Version 12.6 Cisco IOS XE • Simple Network Management Protocol (SNMP) Support for
Gibraltar 16.11.1a Unified SRST
• Toll Fraud Prevention for SIP Line Side on Unified SRST
• Unified SRST, Unified E-SRST, and Unified Secure SRST
Password Policy

Version 12.5 Cisco IOS XE Support for Unified SRST on Cisco 4461 Integrated Services Routers
Gibraltar 16.10.1a

Version 12.3 Cisco IOS XE Fuji Secure SCCP SRST on Cisco 4000 Series Integrated Services Router
16.9.1

Version 12.2 Cisco IOS XE Fuji Unified E-SRST with Support for Voice Hunt Group
16.8.1

Version 12.1 Cisco IOS XE Fuji • Licensing

16.7.1
• Secure SCCP SRST on Cisco 4000 Series Integrated Services
Router
• Unified SRST and Unified Border Element Co-location

Version 12.0 Cisco IOS XE IPv6 Support for Unified SRST SIP IP Phones
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 • Setting Up the Network

• Support for Cisco Unified DX650 SIP IP Phones
• Support for Cisco Unified 78xx SIP IP Phones
• Support for Cisco IP Phones 88xx, 8941, 8945, and 8961

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Feature Roadmap

Cisco Unified Cisco IOS Release Enhancements or Modifications

SRST

Version 10.0 15.3(3)M • Cisco Jabber for Windows

• SIP: Configure Unified E-SRST

Version 9.5 15.3(2)T • After-hour Pattern Blocking Support for Regular Expressions
• Call Park Recall Enhancement
• Display Support for Name of Called Voice Hunt Groups
• Preventing Local-Call Forwarding to Final Agent in Voice Hunt
Groups
• Trunk-to-Trunk Transfer Blocking for Toll Fraud Prevention on
Cisco Unified SIP IP Phones

Version 9.1 15.2(4)M • Key Expansion Module Support for Cisco Unified SIP IP Phones
• Enhancement in Speed-Dial Support
• Voice Hunt Group Support

Version 9.0 15.2(2)T • Support for Cisco Unified 6901 and 6911 SIP IP Phones
• Support for Cisco Unified 6921, 6941, 6945, and 6961 SIP IP
Phones
• Support for Cisco Unified 8941 and 8945 SIP IP Phones
• Multiple Calls Per Line
• Voice and Fax Support on Cisco ATA-187

Version 8.8 15.2(1)T Support for Cisco Unified 6945, 8941, and 8945 SCCP IP Phones

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.

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:
• SRST Routers and the TLS Protocol
• Media Security on Unified SRST - SRTP
• Configuring Secure SIP Call Signaling and SRTP Media with
Cisco SRST

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Feature Roadmap

Cisco Unified Cisco IOS Release Enhancements or Modifications

SRST

Version 7.0/4.3 See Cisco Feature • Configuring Eight Calls per Button (Octo-Line)
Navigator for
compatibility. • Configuring Consultative Transfer

Version 4.2(1) See Cisco Feature Enhanced 911 Services.

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

• Disabling SIP Supplementary Services for Call Forward and Call
Transfer
• Configuring Idle Prompt Status for SIP Phones
• Enhanced 911 Services

Version 4.0 12.4(4)XC • Cisco IP Communicator Support

• Fax Pass-through using SCCP and ATAs Support
• H.323 VoIP Call Preservation Enhancements for WAN Link
Failures for SCCP Phones
• Video Support

Version 3.4 12.4(4)T • Cisco SIP SRST 3.4

• Appendix A: Configuring Cisco Unified SIP SRST Features
Using Redirect Mode
• Configuring Call Handling (see Back-to-Back User Agent Mode)

Version 3.3 • Secure SRST

• Cisco Unified IP Phone 7970G and Cisco Unified 7971G-GE
Support
• Enhancement to the show ephone Command

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Feature Roadmap

Cisco Unified Cisco IOS Release Enhancements or Modifications

SRST

Version 3.2 12.3(11)T • Enhancement to the alias Command

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

Version 3.1 12.3(7)T • Cisco Unified IP Phone 7920 Support

• Cisco Unified IP Phone 7936 Support

Version 3.0 12.2(15)ZJ • Additional Language Options for IP Phone Display

12.3(4)T
• Consultative Call Transfer and Forward Using H.450.2 and
H.450.3 for SCCP Phones
• Customized System Message for Cisco Unified IP Phones
• Dual-Line Mode
• E1 R2 Signaling Support
• European Date Formats
• Huntstop for Dual-Line Mode
• Music On Hold for Multicast from Flash Files
• Ringing Timeout Default
• Secondary Dial Tone
• Enhancement to the Show ephone Command
• System Log Messages for Phone Registrations
• Three-Party G.711 Ad Hoc Conferencing
• Support for Cisco VG248 Analog Phone Gateway 1.2(1) and
Higher Versions

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Feature Roadmap

Cisco Unified Cisco IOS Release Enhancements or Modifications

SRST

Version 2.1 • Cisco Unified IP Phone 7902G Support

• Cisco Unified IP Phone 7912G Support
• Additional Language Options for IP Phone Display
• Cisco Unified SRST Aggregation
• Cisco ATA 186 and ATA 188 Support
• Cisco Unified IP Phone 7905G Support
• Cisco Unified IP Phone Expansion Module 7914 Support
• Enhancement to the Dial Plan-Pattern Command

Version 2.02 • Cisco Unified IP Phone Conference Station 7935 Support

• Increase in Directory Numbers
• Cisco Unity Voicemail Integration Using In-Band DTMF
Signaling Across the PSTN and BRI/PRI, on page 40
• 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.

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Feature Roadmap

Cisco Unified Cisco IOS Release Enhancements or Modifications

SRST

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.

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Feature Roadmap

Cisco Unified Cisco IOS Release Enhancements or Modifications

SRST

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.

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Information About New Features in Cisco Unified SRST

Information About New Features in Cisco Unified SRST

New Features for Cisco Unified SRST Version 14.2
Cisco Unified SRST 14.2 Release introduces support for the following new features:
• Restrict Secure SIP SRST and Secure SCCP SRST to only using TLS 1.2 SHA2 Cipher
Suites—SHA2-Cipher-Only Mode for Unified Secure SRST, on page 284
• SIP OAuth Support for Secure SRST—SIP OAuth Client Registration for Unified Secure SRST, on page
276

New Features for Unified SRST Version 14.1

Unified SRST 14.1 Release introduces support for the following new features:
• Voice: Class of Restriction YANG Configuration Model—Programmability Guide for Cisco IOS XE
Unified Communications VoIP Products
• Smart Licensing Using Policy—Cisco Smart Licensing for Unified SRST
• Smart Licensing Using Policy—Cisco Smart Licensing for Unified E-SRST

New Features for Unified SRST Version 12.7

Unified SRST 12.7 Release introduces support for the following new feature:
• Support for maximum number of devices in Cisco 4451 and 4461 Integrated Services Routers was
increased from 1500 to 2000.

New Features for Cisco Unified SRST Version 12.6

Cisco Unified SRST 12.6 Release introduces support for the following new features:
• Simple Network Management Protocol (SNMP) Support for Unified SRST
• Toll Fraud Prevention for SIP Line Side on Unified SRST
• Unified SRST, Unified E-SRST, and Unified Secure SRST Password Policy

New Features for Cisco Unified SRST Version 12.3

Cisco Unified SRST 12.3 Release introduces support for Secure SCCP SRST on Cisco 4000 Series Integrated
Services Router.

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New Features for Cisco Unified SRST Version 12.2

New Features for Cisco Unified SRST Version 12.2

Cisco Unified SRST 12.2 Release introduces support for Unified E-SRST with Support for Voice Hunt Group.

New Features for Cisco Unified SRST Version 12.1

Cisco Unified SRST 12.1 introduces support for the following new features:
• Licensing
• Secure SCCP SRST on Cisco 4000 Series Integrated Services Router
• Unified SRST and Unified Border Element Co-location

New Feature for Cisco 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 Cisco Unified SRST, see IPv6 Support for Unified SRST SIP IP Phones.

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
Cisco Unified Communications Manager Express, Cisco 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:
• Where to Go Next, Setting Up the Network, on page 163

For more information on the Cisco Unified SRST 10.5 supported feature, see the SCCP: Configure Unified
E-SRST.
Cisco Unified SRST 10.5 supports the following new Cisco Unified SIP IP phones:
• Support for Cisco Unified DX650 SIP IP Phones
• Support for Cisco Unified 78xx SIP IP Phones

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Support for Cisco Unified DX650 SIP IP Phones

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.

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.

New Features in Cisco Unified SRST Version 10.0

Cisco Unified SRST 10.0 supports the following new features:
• Cisco Jabber for Windows
• SIP: Configure Unified E-SRST

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.

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Version Negotiation for Cisco Unified SIP IP Phones

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.

New Features in Cisco Unified SRST Version 9.5

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*
• 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.

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Call Park Recall Enhancement

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.

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.
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.

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Display Support for Name of Called Voice Hunt Groups

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 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.

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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Preventing Local-Call Forwarding to Final Agent in Voice Hunt Groups

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.
The following table lists the transfer-blocking commands and the appropriate configuration modes for Cisco
Unified CME and Cisco Unified SRST.

Commands Cisco Unified SRST

transfer-pattern call-manager-fallback

transfer max-length voice register pool

transfer-pattern blocked voice register pool

conference call-manager-fallback
transfer-pattern

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.

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Transfer-Pattern

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 the trunk) are allowed when no transfer patterns are configured through the transfer-pattern
, transfer-pattern blocked, or transfer max-length commands.
For Cisco Unified SCCP IP phones, if you do not configure transfer patterns, Call Transfers over the trunk
are blocked.

Dial Plans
Whatever dial plan is used for external calls, the same numbers should be configured as specific numbers
using thetransfer-pattern command.
If a dial plan requires “9” to be dialed before making an external call, then prefix “9” to the transfer-pattern
number. For example, if 12345678 is an external number that requires “9” to be dialed before making the
external call, then the transfer-pattern number is 912345678.

Transfer Max-Length
The transfer max-length command is used to indicate the maximum length of the number being dialed for
Call Transfer. When only a specific number of digits are allowed during a Call Transfer, value from 3 through
16 is configured. When the number dialed exceeds the maximum length, then the Call Transfer is blocked.
For example, if you configure 5 as the maximum length, Call Transfers from Cisco Unified SIP IP phones
allows up to a five-digit directory number. All Call Transfers to directory numbers with more than five digits
are blocked.

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Transfer-Pattern 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 nonlocal
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.
The following table compares the behaviors of Cisco Unified SCCP and SIP IP phones for specific
configurations.

Configuration Cisco Unified SCCP IP Phones Cisco Unified SIP IP Phones

No transfer patterns are Blocks all nonlocal Call Transfers. Allows all nonlocal Call Transfers for
configured. backward compatibility.

Specific transfer patterns Allows Call Transfers to specific Allows Call Transfers to specific external
are configured. external entities. entities.

The transfer-pattern Blocks all nonlocal Call Transfers All nonlocal Call Transfers are blocked.
blocked command is are blocked.
Note The configuration
configured.
Note The configuration reverts unconditionally blocks all
to the default, where no nonlocal Call Transfers. It does
transfer patterns are not return to the default, where
configured. all nonlocal 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

No conference-pattern Allowing/Blocking of conference call Allowing/Blocking of conference call

blocked (default case) depends on configured conference depends on configured transfer
max-length. max-length.

Conference-pattern blocked Conference calls are not allowed on SIP and SCCP phones.

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Configuring the Maximum Number of Digits for a Conference Call

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 + Y Y Y N
Conference max-length
(conference max-length has
precedence over transfer
max-length for conference)

No transfer max-length + Y Y N N
Conference max-length
(conference max-length has
precedence over transfer
max-length for conference)

No transfer max-length + No All transfer and conference calls are allowed.

conference max-length

Configuring the Maximum Number of Digits for a Conference Call

Before you begin

Cisco Unified SRST 10.5 or a later version.

SUMMARY STEPS
1. enable
2. configure terminal
3. voice register pool pool-tag ORephonephone-tag
4. conference max-length value
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
Example: • Enter your password if prompted.
Router# enable

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Configuring Conference Blocking Options for Phones

Command or Action Purpose

Step 2 configure terminal Enters global configuration mode.
Example:
Router# configure terminal

Step 3 voice register pool pool-tag ORephonephone-tag Enters voice register Pool configuration mode and creates
a Pool configuration for a Cisco Unified SIP IP phone in
Example:
Cisco Unified Communications Manager Express or for a
Router(config)# voice register pool 25 set of Cisco Unified SIP IP phones in Cisco Unified SIP
SRST.
• pool-tag : Unique number assigned to the Pool. Range
is 1–100.

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 1–10.

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 max-length value 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
max-lenght 6 conference max-length Allows conference call depending
on the configured conference max-length. Range is 3–16.

Step 5 end Exits telephony-service configuration mode and enter

privileged EXEC mode.
Example:
Router(config-telephony)# end

Configuring Conference Blocking Options for Phones

Before you begin

• Use Cisco Unified SRST 10.5 or a later version.
• Configure the transfer-pattern command.
• Configure the conference transfer-pattern command.

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Configuring Conference Blocking Options for Phones

SUMMARY STEPS
1. enable
2. configure terminal
3. voice register pool pool-tag ORephonephone-tag
4. conference-pattern blocked
5. exit

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
Example: • Enter your password if prompted.
Router# enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 voice register pool pool-tag ORephonephone-tag Enters voice register Pool configuration mode and creates
a Pool configuration for a Cisco Unified SIP IP phone in
Example:
Cisco Unified Communications Manager Express or for a
Router(config)# voice register pool 25 set of Cisco Unified SIP IP phones in Cisco Unified SIP
SRST.
• pool-tag : Unique number assigned to the Pool. Range
is 1–100.

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 1–10.

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.

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Transfer-Pattern Blocked

Command or Action Purpose

Step 5 exit Exits telephony-service configuration mode and enter 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 nonlocal
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.
The following table compares the behaviors of Cisco Unified SCCP and SIP IP phones for specific
configurations.

Configuration Cisco Unified SCCP IP Phones Cisco Unified SIP IP Phones

No transfer patterns are Blocks all nonlocal Call Transfers. Allows all nonlocal Call Transfers for
configured. backward compatibility.

Specific transfer patterns Allows Call Transfers to specific Allows Call Transfers to specific external
are configured. external entities. entities.

The transfer-pattern Blocks all nonlocal Call Transfers All nonlocal Call Transfers are blocked.
blocked command is are blocked.
Note The configuration
configured.
Note The configuration reverts unconditionally blocks all
to the default, where no nonlocal Call Transfers. It does
transfer patterns are not return to the default, where
configured. all nonlocal 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 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 Communications Manager
Express: Example section of Cisco Unified Communications Manager Administration Guide.

New Features in Cisco Unified SRST Version 9.1

Cisco Unified SRST 9.1 supports the following new features:
• Key Expansion Module Support for Cisco Unified SIP IP Phones

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Key Expansion Module Support for Cisco Unified SIP IP Phones

• Enhancement in Speed-Dial Support

• Voice Hunt Group Support

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 must upgrade to
Cisco ISR 881, 886VA, 887VA, 888, 888E, 1861E, 2900, 3900, and 3900E Series platforms to utilize
these new features.

Key Expansion Module 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 KEMs support for Cisco Unified 8851/51NR, 8861, 8961, 9951, and 971 SIP IP phones,
see Phone Feature Support Guide for Cisco Unified Communications Manager Express, Cisco 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 UDP port for registration.
• KEMs 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

Cisco Unified SRST 9.1 ignores the “,” or comma (pause indicator) to avoid break-in speed-dial support.
Because the pause speed-dial feature (supported in Cisco Unified Communications Manager or Cisco Unified
Communications Manager) is not supported in Cisco Unified SRST, Cisco Unified Communications Manager
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

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Voice Hunt Group Support

Unified SRST router during WAN outages and Cisco Unified Communications Manager fails, 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 call placed to a single
(pilot) number to contact multiple destinations.
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 while defining the hunt group. 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, reregistered, or went on-hook.

Cisco Unified SCCP IP phones support only ephone hunt groups whereas a voice hunt group supports Cisco
Unified SCCP IP phones, Cisco Unified SIP IP phones. In addition, it also supports 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 Communications Manager 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 groups as the pilot of another hunt group is
not supported.

New Features in Cisco Unified SRST Version 9.0

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 Cisco Unified Communications Manager Express, Cisco Unified SRST,
Unified E-SRST, and Unified Secure SRST.

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 Cisco Unified Survivable Remote Site Telephony Feature Roadmap
Support for Cisco Unified 6921, 6941, 6945, and 6961 SIP IP Phones

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 Cisco Unified Communications Manager Express, Cisco
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 Cisco Unified Communications Manager Express, Cisco Unified SRST,
Unified E-SRST, and Unified Secure SRST.

Multiple Calls Per Line

Cisco Unified SRST 9.0 supports the Multiple Calls Per Line (MCPL) feature on Cisco Unified 6921, 6941,
6945, and 6961 SIP IP phones. In addition, it supports Cisco Unified 8941, 8945 SCCP, and SIP IP phones.
Before Cisco Unified SRST 9.0, supports only two calls for every directory number (DN) on Cisco Unified
8941 and 8945 SCCP IP phones.
With Cisco Unified SRST 9.0, the MCPL feature overcomes the limitation on the maximum number of calls
per line.
Cisco Unified SRST 9.0 does not support the MCPL feature on Cisco Unified 6921, 6941, 6945, and 6961
SCCP IP phones. Allows only two calls on these phones whereas allows only one call on octo-line directory
numbers on these phones before activating Call Forward Busy or busy tone.

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 andtimeouts 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 Configuring the Maximum Number of Calls.

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:
• 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.

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Voice and Fax Support on Cisco ATA-187

Voice and Fax Support on Cisco ATA-187

Cisco ATA-187 is a SIP-based analog phone adapter that turns traditional phone 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 Cisco Unified Communications Manager Express, Cisco 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.

New Features in Cisco Unified SRST Version 8.8

Cisco Unified SRST 8.8 supports the following new Cisco Unified SCCP IP phones:
• Cisco Unified 6945 SCCP IP Phones
• Cisco Unified 8941 SCCP IP Phones
• Cisco Unified 8945 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 Cisco Unified Communications Manager Express, Cisco
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).

New Features in Cisco Unified SRST Version 8.0

Beginning with Cisco IP Phone firmware 8.5(3) and Cisco IOS Release 15.1(1)T, Cisco Unified 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:
• Configuring Eight Calls per Button (Octo-Line)
• Configuring Consultative Transfer

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New Features in Cisco Unified SRST Version 4.2(1)

New Features in Cisco Unified SRST Version 4.2(1)

Cisco Unified SRST Version 4.2(1) introduces the new feature enhancements for Enhanced 911 Services.

New Features in Cisco Unified SRST Version 4.1

Cisco Unified SRST Version 4.1 introduces the following new feature:
• Enhanced 911 Services

New Features in Cisco Unified SRST Version 4.0

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).
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.

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Fax Pass-through using SCCP and ATAs Support

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.
For more information, see Setting Video Parameters.

New Features in Cisco Unified SRST Version 3.4

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.

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New Features in Cisco SRST Version 3.3

New Features in Cisco SRST Version 3.3

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
Configuring Secure SRST for SCCP and SIP, on page 265.

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 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).

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New Features in Cisco SRST Version 3.2

New Features in Cisco SRST Version 3.2

Enhancement to the alias Command
The alias command is enhanced as follows:
• The cfw keyword was added, providing call forward no-answer/busy capabilities.
• 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 Command
Reference (All Versions).

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
Command Reference (All Versions).

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 Appendix B: Integrating Cisco Unified Communications Manager and Cisco Unified SRST to Use
Cisco Unified SRST as a Multicast MOH Resource section for configuration instructions.

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No Timeout for Call Preservation

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 SCCP and SIP SRST Feature Overview section 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 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 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).

New Features in Cisco Unified SRST Version 3.1

Cisco Unified SRST V3.1 introduced the new features described in the following sections:
• Cisco Unified IP Phone 7920 Support
• Cisco Unified IP Phone 7936 Support

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

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.
No configuration is necessary.

New Features in Cisco SRST Version 3.0

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
extra ISO-3166 codes for German, Danish, Spanish, French, Italian, Japanese, Dutch, Norwegian, Portuguese,
Russian, Swedish, United States.

Note This feature is available only for Cisco Unified SRST running under Cisco Unified Communications
Manager V3.2.

Consultative Call Transfer and Forward Using H.450.2 and H.450.3 for SCCP Phones
Cisco Unified SRST V1.0, Cisco Unified SRST V2.0, and Cisco Unified 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 Unified SRST use a Cisco Unified SRST
proprietary mechanism to perform blind transfers. Cisco Unified 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 Unified 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.

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Customized System Message for Cisco Unified IP Phones

Note All voice gateway routers in the VoIP network must support H.450. For H.450 support, routers with
Cisco Unified SRST must run either Cisco Unified SRST V3.0 and higher versions or Cisco IOS Release
12.2(15)ZJ and later releases. Routers without Cisco Unified SRST must run either Cisco Unified SRST
V2.1 and higher versions or Cisco IOS Release 12.2(11)YT and later releases. SIP phones do not support
this feature.

For more information about the default session application, see the Default Session Application Enhancements
Guide.
For configuration information, see the Enabling Consultative Call Transfer and Forward Using H.450.2 and
H.450.3 with Cisco Unified SRST 3.0 section.

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.

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 Unified SRST fallback. The max-dn command
affects all IP phones on a Cisco Unified SRST router.
For configuration information, see the Configuring Dual-Line Phones section.

E1 R2 Signaling Support
Cisco Unified 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 several countries and geographic regions implement R2 in
entirely different ways. Cisco addresses this challenge by supporting many localized implementations of R2
signaling in its Cisco IOS Software.
The Cisco 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 that there are multiple R2 signaling types in the specified country, but Cisco supports the ITU variant.
Cisco also supports specific local variants of E1 R2 signaling in the following regions, countries, and
corporations:
• Argentina
• Australia
• Bolivia
• Brazil

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European Date Formats

• 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)
• Telmex Corporation (Mexico)
• Telnor Corporation (Mexico)
• Thailand
• Uruguay
• Venezuela
• Vietnam

European Date Formats

The date format on a Cisco IP phone display can be configured with the following two extra formats:

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Huntstop for Dual-Line Mode

• 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.

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.

Music On Hold for Multicast from Flash Files

You can configure Cisco Unified SRST 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.

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.

Secondary Dial Tone

Secondary dial tone is available for Cisco Unified IP Phones running Cisco Unified SRST. The secondary
dial tone is generated when you dial a predefined PSTN access prefix. For example, you would hear different
dial tone when a designated number is pressed to reach an outside line.
The secondary dial tone is created through the secondary dial tone command. For more information, see the
Configuring a Secondary Dial Tone section.

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.

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Three-Party G.711 Ad Hoc Conferencing

Three-Party G.711 Ad Hoc Conferencing

Cisco Unified SRST supports three-party instant meeting conferencing using the G.711 coding technique.
For conferencing to be available, connect two lines to one or more buttons of an IP phone.
For more information, see the Enabling Three-Party G.711 Ad Hoc Conferencing section.

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 Unified
Communications system. It allows organizations to support their legacy analog devices while taking advantage
of the new opportunities afforded by using IP telephony. The Cisco VG248 is a high-density gateway for
using analog phones, fax machines, modems, voicemail systems, and speakerphones within an enterprise
voice system based on Cisco Unified Communications Manager.
During Cisco Unified Communications Manager 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 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.

New Features in Cisco SRST Version 2.1

Cisco SRST V2.1 introduced the new features described in the following sections:
• Additional Language Options for IP Phone Display
• Cisco Unified SRST Aggregation
• Cisco ATA 186 and ATA 188 Support
• Cisco Unified IP Phone 7902G Support
• Cisco Unified IP Phone 7905G Support
• Cisco Unified IP Phone 7912G Support
• Cisco Unified IP Phone Expansion Module 7914 Support
• Enhancement to the Dial Plan-Pattern Command

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

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Cisco Unified SRST Aggregation

• Italy
• Portugal
• Spain
• United States

Note This feature is available only in Cisco Unified SRST running under Cisco Unified Communications
Manager V3.2.

For configuration information, see the Configuring IP Phone Language Display section.

Cisco Unified SRST Aggregation

For systems running Cisco Unified Communications Manager 3.3(2) and later versions, the restriction of
running Cisco Unified SRST on a default gateway was removed. Multiple SRST routers can be used to support
more phones. Carefully plan and configure the dial peers and dial plans for Call Transfer and forwarding to
work properly.

Cisco ATA 186 and ATA 188 Support

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

Cisco Unified IP Phone 7905G Support

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 softkeys that guide a user through call features and functions via the
pixel-based 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.
No configuration is necessary.

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Cisco Unified IP Phone 7912G Support

Cisco Unified IP Phone 7912G Support

The Cisco Unified IP Phone 7912G provides core business features and addresses the communication needs
of a cubicle worker who conducts low to medium phone traffic. Four dynamic softkeys 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 from 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 Dial Plan-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 phone 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).

New Features in Cisco SRST Version 2.02

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

The following table shows the increases in directory numbers.

Cisco Router Maximum Phones Increase in Maximum Directory Number

From To

Cisco 1751 24 96 120

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Cisco Unity Voicemail Integration Using In-Band DTMF Signaling Across the PSTN and BRI/PRI

Cisco Router Maximum Phones Increase in Maximum Directory Number

From To

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

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 Cisco 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

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 CHAPTER 2
Cisco Unified SCCP and SIP 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.
• Cisco Unified SRST Feature Overview, on page 41
• Cisco Unified SCCP SRST, on page 42
• Cisco Unified SIP SRST, on page 49
• Interface Support for Cisco Unified Communications Manager Express and Cisco Unified SRST, on
page 55
• MGCP Gateways and SRST, on page 55
• IPv6 Support for Unified SRST SIP IP Phones, on page 56
• Support for Cisco Unified IP Phones and Platforms, on page 61
• Multicast Music On Hold, on page 64
• Where to Go Next, on page 66
• Additional References, on page 67
• Obtaining Documentation, Obtaining Support, and Security Guidelines, on page 69

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.

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Cisco Unified SCCP SRST

Cisco Unified SCCP SRST

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.

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Information About SCCP SRST

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.

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 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

The figure 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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Prerequisites for Configuring Cisco Unified SCCP SRST

Figure 1: Branch Office Cisco Unifed IP Phones Connected to a Remote Central Cisco Unified Communications Manage Operating in
SRST Mode

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 Licensing.
• You have an account on Cisco.com to download software.
To obtain an account on Cisco.com, go to http://www.cisco.com and clickRegister at the top of the
screen.

Installing Cisco Unified Communications Manager

When installing Cisco Unified Communications Manager, consider the following:

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Installing Cisco Unified SCCP SRST

• 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 Integrating Cisco Unified SIP SRST with Cisco
Unified Communications Manager section.

Installing Cisco Unified SCCP SRST

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. Cisco IOS software can be downloaded from the Cisco
Software Center at http://www.cisco.com/public/sw-center/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. 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.
1. Create an SRST Reference
• From any page in Cisco Unified Communications Manager, click System and SRST.
• On the Find and List SRST References page, click Add a New SRST Reference.
• 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.
• Click Insert.

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If You Have Cisco Unified Communications Manager Version Prior to V3.3

2. Apply the SRST reference or the default gateway to one or more device pools.
• From any page in Cisco Unified Communications Manager, click System and Device Pool.
• On the Device Pool Configuration page, click on the required device pool icon.
• 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.
1. Go to the Cisco Unified Communications Manager Phone Configuration page.
• From any page in Cisco Unified Communications Manager, click Device and Phone.
• In the Find and List Phones page, click Find.
• After a list of phones appears, click on the required device name.
• The Phone Configuration appears.

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.
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.

Restrictions for Configuring Cisco Unified SCCP SRST

The following table provides a history of restrictions from Cisco SCCP SRST Version 1.0 to the present
version of Cisco Unified SCCP SRST.

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Cisco Unified Cisco IOS Restrictions

SRST Version Release

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.

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Cisco Unified Cisco IOS Restrictions

SRST Version Release

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
Version 3.2 12.3(11)T in fallback mode displays the name and number of the caller. SCCP
phones in fallback mode display only the caller-id number assigned to
Version 3.1 12.3(7)T the line; the caller-ID name configuration for SCCP phones is not
Version 3.0 12.2(15)ZJ preserved during SRST fallback.

Version 2.1 12.3(4)T Call transfer is supported only on the following:

Version 2.02 12.2(15)T • VoIP H.323, VoFR, and VoATM between Cisco gateways running
Version 2.01 12.2(13)T Cisco IOS Release 12.2(11)T and using the H.323 nonstandard
information element
Version 2.0 12.2(11)T
• FXO and FXS loop-start (analog)
12.2(8)T1
• FXO and FXS ground-start (analog)
12.2(8)T
• Ear and mouth (E&M) (analog) and DID (analog)
12.2(2)XT
• T1 channel-associated signaling (CAS) with FXO and FXS ground-start
signaling
• 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.

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Cisco Unified SIP SRST

Cisco Unified Cisco IOS Restrictions

SRST Version Release

Version 1.0 12.2(2)XB • Does not support first generation Cisco Unified IP phones, such as
Cisco IP Phone 30 VIP and Cisco IP Phone 12 SP+.
12.2(2)XG
12.1(5)YD • Does not support other Cisco Unified Communications Manager
applications or 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.

Note Voice VRF is not supported for SCCP SRST on Cisco Integrated Services Router Generation 2 (ISR
G2).

Cisco Unified SIP SRST

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 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

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

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Prerequisites for Configuring Cisco Unified SIP SRST

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 Branch Office Cisco Unifed IP Phones Connected to a Remote Central Cisco
Unified Communications Manage Operating in SRST Mode.

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
to Licensing section in Cisco Unified SIP SRST on Cisco 4000 Series Integrated Services Router chapter.

Restrictions for Configuring Cisco Unified SIP SRST

The following table provides a history of restrictions from Cisco SIP SRST Version 3.0 to the present version
of Cisco Unified SIP SRST.

Cisco Cisco IOS Restrictions

Unified Release
SRST
Version

Version 15.1(1)T SIP phones may be configured on the Cisco Unified CM with an Authenticated
8.0 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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Cisco Cisco IOS Restrictions

Unified Release
SRST
Version

Version 12.4.(15)T
4.1

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Cisco Cisco IOS Restrictions

Unified Release
SRST
Version
• 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

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Cisco Cisco IOS Restrictions

Unified Release
SRST
Version
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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Restrictions for Configuring Cisco Unified SIP SRST

Cisco Cisco IOS Restrictions

Unified Release
SRST
Version

Version 12.4(4)XC Not Supported

4.0
12.4(4)T • MOH is not supported for a call hold invoked from a SIP phone. A caller hears
Version only silence when placed on hold by a SIP phone.
12.3(11)T
3.4
12.3(7)T • As of Cisco IOS Release 12.4(4)T, bridged call appearance, find-me, incoming
Version call screening, paging, SIP presence, call park, call pickup, and SIP location
3.2 12.2(15)ZJ are not supported.
12.3(4)T
Version • SIP-NAT is not supported.
3.1
• Cisco Unity Express is not supported.
Version
3.0 • 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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Interface Support for Cisco Unified Communications Manager Express and Cisco Unified SRST

Interface Support for Cisco Unified Communications Manager

Express and Cisco Unified SRST
Cisco Unified Communications Manager Express and Cisco Unified SRST routers have multiple interfaces
and is 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
depend 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, use the Loopback Interface for Cisco Unified Communications Manager Express and Cisco
Unified SRST.
Cisco Unified Communications Manager Express and Cisco Unified SRST supports the following interfaces:
• 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.
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 alternatecommands. 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.

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 Configuring MGCP Gateway Support
for Cisco Unified Communications Manager document or the MGCP Gateway Fallback Transition to Default
H.323 Session Application document.

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IPv6 Support for Unified SRST SIP IP Phones

IPv6 Support for Unified SRST SIP IP Phones

IPv6 Support for Unified SRST SIP IP Phones
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, on page 60.
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, on page 57.
For an example of configuring IPv6 Support on Unified SRST, see Examples for Configuring IPv6 Pools for
SIP IP Phones, on page 60.
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 basic feature supported for a IPv6 WAN down scenario is:
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.

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Configure IPv6 Pools for SIP IP Phones

• 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.
• 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 Survivable Remote Site Telephony Configurationin Cisco Unified
Communications Manager Administration Guide.

SUMMARY STEPS
1. enable
2. configure terminal
3. ipv6 unicast-routing
4. voice service voip
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
17. voice register global
18. default mode

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19. max-dnmax-directory-numbers
20. max-poolmax-voice-register-pools
21. exit
22. voice register poolpool-tag
23. id{networkaddressmaskmask|ip address maskmask|macaddress}
24. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
Example: • Enter your password if prompted.
Router> enable

Step 2 configure terminal Enters global configuration mode.

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:
Router (config)# voice service voip • voip — Specifies Voice over IP (VoIP) parameters.

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

Step 9 exit Exits voice service voip configuration mode.

Example:
Router(config-voi-sip)# exit

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Command or Action Purpose

Step 10 sip-ua Enters SIP user-agent configuration mode.
Example:
Router(config)# sip-ua

Step 11 protocol Allows phones to interact with phones on IPv6 voice

mode{ipv4|ipv6|dual-stack[preference{ipv4|ipv6}]} gateways. You can configure phones for IPv4 addresses,
IPv6 address es, or for a dual-stack mode.
Example:
Router(config-sip-ua)# protocol mode dual-stack • ipv4—Allows you to set the protocol mode as an IPv4
preference ipv6 address.
• 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.

Step 12 exit Exits SIP configuration mode.

Example:
Router(config-sip-ua)# exit

Step 13 voice service{voip} Enters voice-service configuration mode to specify a voice

encapsulation type.
Example:
Router (config)# voice service voip • voip — Specifies Voice over IP (VoIP) parameters.

Step 14 sip Enters SIP configuration mode.

Example:
Router(config-voi-serv)# sip

Step 15 no call service stop Activates SIP call service.

Example:
Router(config-serv-sip)# call service stop

Step 16 exit Exits SIP configuration mode.

Example:
Router(config-serv-sip)# exit

Step 17 voice register global Enters voice register global configuration mode to set
parameters for all supported SIP phones in Cisco Unified
Example:
CME.
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

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Command or Action Purpose

Step 19 max-dnmax-directory-numbers Limits number of directory numbers to be supported by
this router.
Example:
Router(config-register-global)# max-dn 50 Maximum number is platform and version-specific. Type
? for value.

Step 20 max-poolmax-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 poolpool-tag Enters voice register pool configuration mode to set
phone-specific parameters for a SIP phone.
Example:
Router(config)# voice register pool 1

Step 23 id{networkaddressmaskmask|ip address Explicitly identifies a locally available individual SIP

maskmask|macaddress} phone to support a degree of authentication.
Example:
Router(config-register-pool)# id network
2001:420:54FF:13::901:0/117
Router(config-register-pool)# id network
10.64.88.0 mask 255.255.255.0

Step 24 end Exits to privileged EXEC mode.

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

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Support for Cisco Unified IP Phones and Platforms

max-pool 40
exit
voice register pool 1
id network 2001:420:54FF:13::901:0/117
end

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
under voice service voip, sip submode

Support for Cisco Unified IP Phones and Platforms

Support for Cisco Unified IP Phones and Platforms
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.
Unified SRST is supported on Cisco 1100 Series Integrated Services Router (ISR) platforms with Cisco IOS
XE Amsterdam 17.3.2 and later releases.
Unified SRST is supported on Cisco 4000 ( 4321, 4331, 4351, 4431, 4451, and 4461) ISR Series Platforms
with Cisco IOS XE Amsterdam 17.3.2 and later releases:
Unified SRST is supported on Cisco Catalyst 8000 Series Edge Platforms as following:
• Cisco Catalyst 8300 Series Edge Platforms—Cisco IOS XE Amsterdam 17.3.2 and later releases
• Cisco Catalyst 8200 Series Edge Platforms—Cisco IOS XE Bengaluru 17.4.1a and later releases
• Cisco Catalyst 8200L Series Edge Platforms—Cisco IOS XE Bengaluru 17.5.1a and later releases

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Finding Cisco IOS Software Releases That Support Cisco Unified SRST

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

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

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.

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Cisco Unified Communications Manager Compatibility

For the most up-to-date information about Cisco IOS software images, see Compatibility Information.

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-5ess Lucent 5ESS 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)

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Multicast Music On Hold

Multicast Music On Hold

Multicast Music On Hold
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.
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-g711filename
(for example, mohenable-g711flash: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 mohfilename. 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.

Configure Multicast Music On Hold for Unified SRST

Before you begin
To configure multicast MOH for Unified SRST, perform the following steps:
• Unified SRST 3.0 or later versions.
• IP phones do not support multicast at 224.x.x.x addresses.

SUMMARY STEPS
1. enable
2. configure terminal
3. call-manager-fallback
4. mohfilename
5. multicast mohip-addressportport number[routeip-address-list]
6. exit

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
Example: • Enter your password if prompted.

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Command or Action Purpose

Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 call-manager-fallback Enters call-manager-fallback configuration mode.

Example:
Router(config)# call-manager-fallback

Step 4 mohfilename Enables music on hold using the specified file.

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

Step 5 multicast mohip-addressportport Specifies that this audio stream is to be used for multicast
number[routeip-address-list] and also for MOH.
Example: Note This command is required to use MOH for
Router(config-cm-fallback)# multicast moh 239.1.1.1 internal calls and it must be configured after
port 2000 MOH is enabled with the moh command.

• 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.

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Where to Go Next

Command or Action Purpose

Step 6 exit Exits call-manager-fallback configuration mode.
Example:
Router(config-cm-fallback)# exit

Where to Go Next
The next chapters of this book describe how to configure Cisco Unified SIP SRST. As shown in the following
table, 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.

Task Where Task Is Described

7. Setting up a Cisco Unified SRST system to communicate Setting Up the Network

with your network

8. Configuring Version 4.1 features Cisco Unified SIP SRST 4.1

9. Setting up the basic Cisco Unified SRST phone Setting Up Cisco Unified IP Phones using SCCP
configuration using SCCP
10. Providing a backup to an external SIP call control Setting Up Cisco Unified IP Phones using SIP
(IP-PBX) by supplying basic registrar services
11. Configuring incoming and outgoing calls Configuring Call Handling

12. Configuring optional security for SRST Configuring Secure SRST for SCCP and SIP

13. Setting up voicemail Integrating Voicemail with Cisco Unified SRST

14. Setting up video parameters Setting Video Parameters

15. Monitoring and maintaining Cisco Unified Survivable Monitoring and Maintaining Cisco Unified SRST
Remote Site Telephony (SRST)

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Additional References

Additional References
Related Documents
Related Topic Documents

Cisco IOS voice product • Cisco IOS Voice Configuration Library

configuration
• 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 • Configuring MGCP Gateway Support for Cisco Unified
Fallback Communications Manager
• MGCP Gateway Fallback Transition to Default H.323 Session
Application
• Configuring SRS Telephony and MGCP Fallback

Cisco Unified • Cisco Unified Communications Manager

Communications Manager user
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 • Cisco Unified SRST and Cisco Unified SIP SRST Command
specifications Reference (All Versions)
• Cisco Unified SRST 8.0 Supported Firmware, Platforms,
Memory, and Voice Products
• Cisco Unified SRST 4.3 Supported Firmware, Platforms,
Memory, and Voice Products

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 Cisco Unified SCCP and SIP SRST Feature Overview
Standards

Related Topic Documents

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 • Cisco IOS SIP SRST Feature Roadmap
Survivable Remote Site Telephony
Feature Roadmap

Cisco SIP functionality • Cisco IOS SIP Configuration Guide

Cisco SRST command reference • Cisco IOS Survivable Remote Site Telephony Version 3.2
Command Reference

Command reference information for • Cisco IOS Voice Command Reference

voice and 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 IP Phones 7900 Series

Cisco Unified SRST
• Survivable Remote Site Telephony

Standard Glossary • Cisco IOS Voice Configuration Library Glossary

Standard Preface • Cisco IOS Voice Configuration Library Preface

Standards
Standard Title

ITU X. 509 Version Public-Key and Attribute Certificate Frameworks

3

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MIBs

MIBs
MIB MIBs Link

No new or modified MIBs are supported by this To locate and download MIBs for selected platforms,
feature, and support for existing MIBs has not Cisco IOS releases, and feature sets, use Cisco MIB
been modified by this feature. Locator found at the following URL:
http://www.cisco.com/go/mibs

RFCs
RFC Title

RFC2246 The Transport Layer Security (TLS) Protocol Version 1.0

RFC SIP: Session Initiation Protocol

2543

RFC SIP: Session Initiation Protocol

3261

RFC3711 The Secure Real-Time Transport Protocol (SRTP)

Technical Assistance
Description Link

The Cisco Technical Support & Documentation website contains thousands http://www.cisco.com/techsupport
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.

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Obtaining Documentation, Obtaining Support, and Security Guidelines

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 CHAPTER 3
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.

Note Unified SRST 12.6 on Cisco IOS XE Gibraltar 16.11.1a Release is not a recommended release version
for call flows that include Multicast Music On Hold.

• Overview, on page 71
• Platform and Memory Support, on page 72
• Cisco IOS Software Releases that Support Unified SRST, on page 72
• Feature Support, on page 74
• Unified IP Phone Support, on page 76
• Cisco Unified Communications Manager Compatibility, on page 76
• Interface Support for Unified SRST, on page 78
• Simple Network Management Protocol (SNMP) Support for Unified SRST, on page 78
• Licensing, on page 78
• Configure SIP Registrar Functionality for SIP Phones on Unified SRST, on page 81
• Unified SRST, Unified E-SRST, and Unified Secure SRST Password Policy, on page 93
• Toll Fraud Prevention for SIP Line Side on Unified SRST, on page 96
• Configure Toll Fraud Prevention, on page 98
• VRF Support for Unified SRST, on page 102
• IPv6 Support for Unified SRST SIP IP Phones, on page 106
• Configure Unified SRST on Cisco 4000 Series Integrated Services Platform, on page 110
• Configure Voice Hunt Groups on Unified SRST, on page 114
• Examples, on page 127

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.

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Platform and Memory Support

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
• 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

Platform and Memory Support

From Unified SRST Release 10.0 (Cisco IOS XE Release 3.10S), Unified SIP SRST is supported on the Cisco
4000 Series Integrated Services platform. As part of the Cisco IOS XE Release 3.10S Release, support was
introduced on the Cisco 4451-X Integrated Services Router. From Unified SRST Release 10.5 (Cisco IOS
XE Release 3.13S), 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
• Cisco ISR 4331 Integrated Services Routers
• Cisco ISR 4351 Integrated Services Routers
• Cisco ISR 4431 Integrated Services Routers
• Cisco ISR 4451 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 XE Software

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

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Install Cisco IOS XE Software

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 XE software release is installed on router. Log in to the router and use the show
version EXEC command.
2. Compare the Cisco IOS XE 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.
3. If necessary, download and extract the recommended Cisco IOS XE image to flash memory in the router.
4. To reload the Unified SRST router with the new software after replacing or upgrading the Cisco IOS XE
release, use the reload privileged EXEC command.

DETAILED STEPS

Command or Action Purpose

Step 1 Identify which Cisco IOS XE software release is installed
on router. Log in to the router and use the show version
EXEC command.
Example:
Router> show version
Cisco IOS XE Software, Version
BLD_POLARIS_DEV_LATEST_20200621_053200
Cisco IOS Software [Amsterdam], ISR Software
(X86_64_LINUX_IOSD-UNIVERSALK9-M),
Version 17.3.1
[S2C-build-polaris_dev-116144-/nobackup/mcpre/BLD-BLD_POLARIS_DEV_LATEST_20200621_0532
00 259]
Copyright (c) 1986-2020 by Cisco Systems, Inc.
Compiled Sun 21-Jun-20 07:03 by mcpre

Step 2 Compare the Cisco IOS XE 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 To find software installation information, access information
IOS XE image to flash memory in the router. located at www.cisco.com > Support > Products &
Downloads > Networking Software > {Choose release} >
Configuration Guides / System Management / Configuration
fundamentals.

Step 4 To reload the Unified SRST router with the new software
after replacing or upgrading the Cisco IOS XE release, use
the reload privileged EXEC command.
Example:
Router# reload
System configuration has been modified. Save?
[yes/no]: yes

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Feature Support

Command or Action Purpose

Building configuration...
[OK]
Proceed with reload? [confirm]
Jun 24 00:45:13.827: %PMAN-5-EXITACTION: R0/0: pvp:
Process manager is exiting:
process exit with reload chassis code
Initializing Hardware ...
Checking for PCIe device presence...done
System integrity status: 0x610
Rom image verified correctly
System Bootstrap, Version 16.12(2r), RELEASE
SOFTWARE
Copyright (c) 1994-2019 by cisco Systems, Inc.
Current image running: Boot ROM0
Last reset cause: LocalSoft
ISR4331/K9 platform with 4194304 Kbytes of main
memory
........
Located
isr4300-universalk9.BLD_POLARIS_DEV_LATEST_20200621_053200.SSA.bin
Router>

Feature Support
The following features are 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

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Restrictions of Unified SRST on Cisco 4000 Series Integrated Services Routers

• 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
• 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.
• Voice VRF is not supported for SCCP SRST on Cisco Integrated Services Router 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.

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Unified IP Phone Support

• The CLI command max-redirect is not supported for SIP on Unified SRST.
• 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 all the SIP phones,
including Cisco IP Phone 7800 Series and Cisco IP Phone 8800 Series.

Cisco Jabber with Unified SRST

Unified SRST 12.8 (Cisco IOS XE Amsterdam 17.2.1r) and later releases support the following Cisco Jabber
clients:
• Cisco Jabber for Windows,12.9
• Cisco Jabber for Mac, 12.9

Cisco Unified Communications Manager Compatibility

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.
• 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 section.

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.

SUMMARY STEPS
1. Create an SRST reference.

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Supported PSTN Trunk Connectivity

2. Apply the SRST reference or the default gateway to one or more device pools.

DETAILED STEPS

Command or Action Purpose

Step 1 Create an SRST reference.
Step 2 Apply the SRST reference or the default gateway to one or
more device pools.

Supported PSTN Trunk Connectivity

Unified SRST is supported with 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
• 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-5ess Lucent 5ESS 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

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Interface Support for Unified SRST

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)

Simple Network Management Protocol (SNMP) Support for

Unified SRST
Unified SRST supports Simple Network Management Protocol (SNMP) Management Information Base
(MIBs) for monitoring the product status. Unified SRST Release 12.6 and later versions is SNMP Version 3
(SNMPv3) compliant. The following is the main SNMP MIB supported by Unified SRST:
• CISCO-SRST-MIB

For information on configuration of SNMP version 3 on Unified SRST router, see SNMP Configuration
Guide.

Licensing
This section provides information on licensing of Cisco Unified Survivable Remote Site Telephony (Unified
SRST).

Cisco Smart Licensing for Unified SRST

Cisco Smart Licensing is a software licensing model that provides visibility of ownership and usage through
the Cisco Smart Software Manager (CSSM) portal. CSSM is a central license repository that manages licenses
across all Cisco products that you own, including Unified SRST. Devices send license usage to CSSM either
directly or use an on-premises satellite. Your Smart Account Administrator controls your access to CSSM.
Use your Cisco credentials to access the CSSM portal using http://software.cisco.com.
Smart Licensing applies to all platform technology (UCK9, Security) and Unified SRST feature licenses that
the router uses. Unified SRST requires one license entitlement (SRST_EP) for each configured SIP or SCCP
phone.

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Smart License Operation

CSSM shows license usage across all devices that are registered to a virtual account. A Virtual Account
License Inventory displays the quantity of licenses that are purchased, those licenses in use, and a balance.
An Insufficient Licenses alert is displayed if the license balance is below 0.
For example, consider a smart account in CSSM with 50 SRST_EP licenses. If you have a single registered
Unified SRST router with 20 phones configured, the CSSM licenses page shows Purchased as 50, In Use as
20 and Balance as 30.
For more information on Smart Software Manager, see the Cisco Smart Software Manager User Guide.

Note The SRST_EP license count reflects the total phone count for both the ephones and voice register pools
that are configured in the Unified SRST irrespective of whether the phones are registered or not. To
avoid unnecessary reporting while Unified SRST is being configured, license usage is reported three
minutes after the last configuration change.

Note Unified SRST Smart Licenses also provide RTU entitlement for routers that are not configured for Smart
Licensing.

Smart License Operation

Cisco IOS XE Everest 16.5.1 Release to Cisco IOS XE Fuji 16.9.1 Release
Cisco 4000 Series Integrated Services Routers support Smart Licensing as an alternative to Cisco Software
RTU Licensing. Use the license smart enable command to enable Smart Licensing. To disable Smart Licensing,
use the no form of the command and re-accept the EULA using the license accept end user agreement
command.

Cisco IOS XE Gibraltar 16.10.1 Release Onwards

The Cisco RTU Licensing and the CLI license smart enable command are deprecated. Smart Licensing is
mandatory from this release.

Cisco IOS XE Everest 16.5.1 Release to Cisco IOS XE Amsterdam 17.3.1a Release
Routers configured to use Smart Licensing offer a 90-day evaluation period, during which you can use all the
features without registering to CSSM. A Unified SRST device is associated with CSSM using a registration
token. You can obtain the registration token from the virtual CSSM account or from an on-premises satellite.
Once registered, the evaluation period pauses and you can use the balance license later. You cannot renew
the evaluation period on its expiry.

Warning Unified SRST shuts down when the router is unregistered and allowed to pass in to the Evaluation
Expired state.

To register the Unified SRST router with CSSM, use license smart register idtoken command. For information
on registering the device with CSSM, see Software Activation Configuration Guide.

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Cisco IOS XE Gibraltar 16.12.1 Release to Cisco IOS XE Amsterdam 17.3.1a Release

Upon successful registration, the device sends an authorization request to CSSM for the licenses in use. For
each license type requested, if the Smart Account has sufficient licenses, CSSM responds with Authorized
. If the Smart Account does not have sufficient licenses, CSSM responds with Out of Compliance .
Post successful authorization of the request, licenses are bound to the requesting device until the next
authorization request submission. An authorization request is sent every 30 days or when there is any change
in license consumption, to maintain the registration with CSSM. The authorization expires if you do not update
the license request for the router within 90 days. The certificate issued to identify the router at the time of
registration is valid for one year and renewed every six months. The router displays the License authorization
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
-----------------------------------------------------------------------------
ISR_4351_UnifiedCommun.. (ISR_4351_UnifiedCommun..) 1 AUTHORIZED
SRST v12 Endpoint Li... (SRST_EP) 4 AUTHORIZED

Cisco IOS XE Gibraltar 16.12.1 Release to Cisco IOS XE Amsterdam 17.3.1a Release
Specific License Reservation (SLR) is supported on Cisco 4000 Series Integrated Services Routers. SLR
allows reservation and utilization of Cisco Smart Licenses without communicating the license information to
CSSM. To reserve specific licenses for a device, generate request code from the device. Enter the request
code in CSSM along with the required licenses and their quantity, and generate authorization code. Enter the
authorization code on the device to map the license to the Unique Device identifier (UDI).

Note If upgrading to IOS XE Amsterdam 17.3.1a with a license reservation in place, update the reservation
to include version 14, rather than version 12 SRST licenses. The reservation may be updated before or
after the software upgrade.

Cisco IOS XE Amsterdam 17.3.2 and Cisco IOS XE Bengaluru 17.4.1a Release Onwards
This release introduces a new paradigm for tracking license usage across your business. In earlier releases,
license authorization was forward looking, binding licenses to a device until the next authorization request.
Actual license usage during the proceeding reporting period is now sent to CSSM, allowing you to plan
ongoing license requirements based on historical usage data. Initial device registration is no longer required
to use most platform functionality and the evaluation period is deprecated.
License usage reports are submitted periodically according to a minimum reporting policy set for your account.
Typically, this period could be once per year. However, you can generate reports more frequently if the use
of licensed features varies over time. CSSM acknowledges each Resource Utilization Monitoring (RUM)
report to ensure that the usage is recorded reliably. If the router does not receive an acknowledgment within

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the minimum reporting period, call processing is disabled. Call processing is resumed when a valid
acknowledgment is received.
Reports can be submitted to CSSM directly or through a satellite. Cisco Smart Licensing Utility (CSLU)
applications can also receive usage reports, providing you with more flexibility in managing your license
usage. Also, when a device is not able to communicate directly with a licensing server, a signed usage report
can be generated and manually uploaded to CSSM. The acknowledgment that is generated by CSSM must
be uploaded to the device within the license reporting policy period to ensure continued use.
As license reporting is now based on historical usage, the registration process that is used previously has been
replaced with a trust association that also defines the reporting policy set in your account. Establishing trust
with CSSM or Cisco Smart Software Manager Satellite uses an identity token similar to earlier registrations.
Use the license smart trust idtoken token command to establish the trust relationship within the initial
reporting period set for the device. The CLI license smart register command is deprecated from this release.
Current license usage for Cisco Unified SRST is displayed using the show license summary command:

Warning When using any of the following releases, Unified SRST shuts down if the router does not receive a
report acknowledgment from CSSM before the acknowledgment deadline set by the account policy:
17.3.2, 17.3.3, 17.3.4a, 17.6.1a, or any 17.4 or 17.5 release. Unified SRST does not shut down in this
way with later releases.

Note Smart License Reservation (SLR) for SRST licenses is not compatible with IOS XE Amsterdam 17.3.2
and later releases. Even if a reservation is in place when upgrading to one of these releases, license use
reporting will still be required in accordance with the device policy.

Router#show license summary

License Usage:
License Entitlement tag Count Status
-----------------------------------------------------------------------------
appxk9...................(ISR_4400_Application) ......1...... IN USE
uck9.................... (ISR_4400_UnifiedCommun..)...1.......IN USE
securityk9.............. (ISR_4400_Security)......... 1.......IN USE
SRST_E_EP............... (SRST_E_EP)..................2.......IN USE
SRST_EP..................(SRST_EP)...................18...... IN USE

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.

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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
2. configure terminal
3. voice service voip
4. allow-connections sip to sip
5. sip
6. registrar server [expires [max sec] [min sec]]
7. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
Example: • Enter your password if prompted.
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

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

Step 5 sip Enters SIP configuration mode.

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:
Router(config)# call-manager-fallback • expires : (Optional) Sets the active time for an
incoming registration.
• max sec : (Optional) Maximum expiration time for a
registration, in seconds. The range is from 600 to
86400. The default is 3600.

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Command or Action Purpose

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 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 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 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.
Restrictions
• The id command identifies the individual SIP IP phone or sets of SIP IP phones that are to be configured.
Thus, theidcommand 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 To monitor SIP proxies, the call fallback active command must be configured, as described in Step 3.

Note The command proxy described in Step 7 is an optional configuration.

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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.

Before you begin

The SIP registrar must be configured before a voice register pool is set up.

SUMMARY STEPS
1. enable
2. configure terminal
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]]
8. voice-class codec tag
9. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
Example: • Enter your password if prompted.
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 call fallback active (Optional) Enables a call request to fall back to alternate
dial peers if there is network congestion.
Example:
Router(config)# call fallback active • This command is used if you want to monitor the proxy
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:
Router(config)# voice register pool 12 Use this command to control which registrations are
accepted or rejected by a Cisco Unified SIP SRST device.

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Command or Action Purpose

Step 5 id [{network address mask mask |ip address mask Explicitly identifies a locally available individual or set of
mask |mac address }] [device-id-name devicename ] SIP IP phones. The keywords and arguments are defined
as follows:
Example:
Router(config-register-pool)# id network • network address mask mask : The network address
172.16.0.0 mask 255.255.0.0 mask mask keyword/argument combination is used
to accept SIP Register messages for the indicated
phone 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
Example:
highest preference.
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.

Step 7 proxy ip-address [preference value] [monitor probe (Optional) Autogenerates additional VoIP dial peers to reach
{icmp-ping | rtr} [alternate-ip-address]] the main SIP proxy whenever a Cisco Unified SIP IP Phone
registers with a Cisco Unified SIP SRST gateway. The
Example:
keywords and arguments are defined as follows:
Router(config-register-pool)# proxy
10.2.161.187 preference 1 • ip-address : IP address of the SIP proxy.
• 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 value 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.

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Command or Action Purpose

• 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 end Returns to privileged EXEC mode.

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.
• 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.

Before you begin

Before configuring the alias command, translation rules must be set using the translation-profile outgoing
(voice register pool) command.

SUMMARY STEPS
1. enable
2. configure terminal
3. voice register pooltag
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 {preferencevalue} [huntstop]
9. dtmf-relay [cisco-rtp] [rtp-nte] [sip-notify]

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10. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
Example: • Enter your password if prompted.
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 voice register pooltag Enters voice register pool configuration mode.
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 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
Example:
a SIP phone.
Router(config-register-pool)#
voice translation-rule 1 • Profile-tag : Translation profile name to handle
rule 1 /1000/ /1006/ translation to outgoing calls.
!
!
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
Example:
the main proxy is not available. The keywords and
Router(config-register-pool)# alias 1 94... to arguments 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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Command or Action Purpose

Step 6 cor {incoming | outgoing} cor-list-name {cor-list-number Configures a class of restriction (COR) on the VoIP dial
starting-number [- ending-number] | default } peers associated with directory numbers. COR specifies
which incoming dial peers can use which outgoing dial
Example:
peers to make a call. Each dial peer can be provisioned
Router(config-register-pool)# cor incoming with an incoming and outgoing COR list. The keywords
call91 1 91011
and arguments are defined as follows:
• 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
Example:
a sequence of digits that represent a phone number prefix.
Router(config-register-pool)# incoming
called-number 308

Step 8 number tag number-pattern {preferencevalue} Indicates the E.164 phone numbers that the registrar
[huntstop] permits to handle the Register message from the
Cisco Unified SIP IP Phone. The keywords and arguments
Example:
are defined as follows:
Router(config-register-pool)# number 1 50..
preference 2 • tag : Number from 1 to 10 and the distinguishing
factor 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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Command or Action Purpose

Step 9 dtmf-relay [cisco-rtp] [rtp-nte] [sip-notify] Specifies how a SIP gateway relays dual tone
multifrequency (DTMF) tones between telephony
Example:
interfaces and an IP network. The keywords are defined
Router(config-register-pool)# dtmf-relay as 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.

Step 10 end Returns to privileged EXEC mode.

Example:
Router(config-register-pool)# end

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

DETAILED STEPS

Command or Action Purpose

Step 1 debug voice register errors Use this command to debug errors that happen during
registration.
Example:
Router# debug voice register errors If there are no voice register pools configured for a
*Apr 22 11:52:54.523 PDT: VOICE_REG_POOL: Contact particular registration request, the message “Contact doesn’t
doesn't match any pools match any pools” is displayed.
*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.
*Apr 22 11:52:54.559 PDT: VOICE_REG_POOL: Register
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
Example:
includes matching of pools, registration creation, and

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Command or Action Purpose

Router# debug voice register events automatic creation of dial peers. For more details and error
Apr 22 10:50:21.731 PDT: VOICE_REG_POOL: Contact
conditions, you can use the debug voice register errors
matches pool 1
Apr 22 10:50:21.731 PDT: VOICE_REG_POOL: key(91011) command.
contact(192.168.0.2) add to contact table
Apr 22 10:50:21.731 PDT: VOICE_REG_POOL: key(91011)
The phone number 91011 registered successfully, and type
exists in contact table 1 is reported, which means there is a pre-existing VoIP dial
Apr 22 10:50:21.731 PDT: VOICE_REG_POOL: 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
Apr 22 10:50:21.731 PDT: VOICE_REG_POOL:
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

Verify Proxy Dial-Peer Configuration

To use the icmp-ping keyword with the proxy command to assist in troubleshooting proxy dial peers,
perform the following steps.

SUMMARY STEPS
1. configure terminal
2. voice register pool tag
3. proxy ip-address [preference value] [monitor probe {icmp-ping | rtr} [alternate-ip-address]]
4. end
5. show voice register dial-peers
6. show dial-peer voice

DETAILED STEPS

Command or Action Purpose

Step 1 configure terminal Use this command to enter global configuration mode.
Example:
Router# configure terminal

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Command or Action Purpose

Step 2 voice register pool tag Use this command to enter voice register pool configuration
mode.
Example:
Router(config)# voice register pool 1

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

Step 4 end Returns to privileged EXEC mode.

Example:
Router(config-register-pool)# end

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
dial-peer voice 40036 voip
preference 1
destination-pattern 91011
session target ipv4:10.2.161.187
session protocol sipv2
voice-class codec 1
monitor probe icmp-ping 10.2.161.187

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 Note Also highlighted is the output of the cor and
VoiceOverIpPeer40036 incoming called-number commands.
peer type = voice, information type = voice,
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 = `unknown'
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,

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Command or Action Purpose

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
! Default output for cor command
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 = `ipv4:10.2.161.187',
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,
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,

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Unified SRST, Unified E-SRST, and Unified Secure SRST Password Policy

Command or Action Purpose

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.

Unified SRST, Unified E-SRST, and Unified Secure SRST

Password Policy
From Unified SRST 12.6 Release (Cisco IOS XE Gibraltar 16.11.1a) onwards, all configurations on Unified
SRST, Unified E-SRST, and Unified Secure SRST must meet the password policy.
General Password Policy Guidelines:
• Passwords must have a minimum of 6 alphanumeric characters, and a maximum of 15 alphanumeric
characters.
• Passwords must not contain symbols or special characters.
• Passwords must contain at least one numeral, one uppercase alphabet, and one lowercase alphabet.

If the password is not configured as per the policy, the Unified SRST router displays an error message:
Error: The password you have entered is incorrect.
Your password must contain:
1. A minimum of 6 and a maximum of 15 alphanumeric characters, excluding symbols and
special characters.
2. A minimum of one numeral, one uppercase alphabet, and one lowercase alphabet.

The Unified CME password policy is applicable for Unified SRST configurations on Cisco IOS XE 16.11.1a
and later. Unified SRST password policy is not applicable in the following scenarios:
• Upgrade from an older IOS version to Cisco IOS XE 16.11.1a
• Downgrade from Cisco IOS XE 16.11.1a to an older version

Guidelines for Password Configuration and Encryption

Configure the passwords relevant to Unified SRST, Unified E-SRST, and Unified Secure SRST using the
CLI commands as follows:
• call-manager-fallback configuration mode
• xml user username password [0|6]password privilege-level

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Deprecation of CLI commands

Note The 0 in the parameter [0|6] mentioned in the CLI command represents plain, unencrypted text and 6
represents level 6 password encryption.

• Apart from the parameter configurations ([0|6]) at the command level, configure the Unified SRST router
to support encryption.
• Configure the CLI command encrypt password under call-manager-fallback configuration mode to
support type 6 encryption on the Unified SRST router.
• Also, it is mandatory to configure key config-key password-encrypt[key]password encryption aes to
support encryption on the Unified SRST router.
• If the key used to encrypt the password is replaced with a new key (replace key or re-key), then the
password is re-encrypted with the new key.
• You must adhere to SRST Password Policy for both type 0 and type 6 parameters that you configure on
Unified SRST.
• Configure no encrypt password for type 0 password on the Unified SRST router. A type 0 password is
displayed as unencrypted plain text.
• If you are performing a downgrade from Unified SRST 12.6 to an earlier version, then you must execute
the CLI command no encrypt password. If the CLI command no encrypt password is configured, the
password is presented as plain text.

The following is a sample configuration on Unified SRST router to support password encryption:
Router(config)#key config-key password-encrypt <cisco123>
Router(config)#password encryption aes
Router(config)#call-manager-fallback
Router(config-cm-fallback)encrypt password

Deprecation of CLI commands

From Unified SRST Release 12.6 onwards, the following CLI commands that are configured under
call-manager-fallback configuration mode are deprecated to enhance product security:
• log passwordpassword-string
• xmltest
• xmlschemaschema-url
• xmlthread number

Removal of Passwords and Keys from Logs

From Unified SRST Release 12.6 onwards, passwords and sRTP keys are not printed to logs to enhance
security of Unified SRST. The information about keys is available only in the show commands from Unified
SRST 12.6 release onwards. The CLI command show ephone offhook for SCCP and show sip-ua calls for
SIP are enhanced to display the keys that are in use per media stream, along with the sRTP Ciphers.

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The following is a sample output for the show command, show sip-ua calls. The lines that are added to the
show command output as part of the Unified SRST 12.6 enhancement are the local crypto key and the remote
crypto key:
SIP UAC CALL INFO
Number of SIP User Agent Client(UAC) calls: 0
SIP UAS CALL INFO
Call 1
SIP Call ID : [email protected]
State of the call : STATE_ACTIVE (7)
Substate of the call : SUBSTATE_NONE (0)
Calling Number : 1001
Called Number : 6901%23
Called URI : sip:6901%[email protected];user=phone
Bit Flags : 0x10C0401C 0x10000100 0x4
CC Call ID : 196
Local UUID : 61488a9100105000a000007278df12e0
Remote UUID : c4b7f9475629538096ef61699b96746f
Source IP Address (Sig ): 8.39.25.11
Destn SIP Req Addr:Port : [8.55.0.195]:52704
Destn SIP Resp Addr:Port: [8.55.0.195]:52704
Destination Name : 8.55.0.195
Number of Media Streams : 1
Number of Active Streams: 1
RTP Fork Object : 0x0
Media Mode : flow-through
Media Stream 1
State of the stream : STREAM_ACTIVE
Stream Call ID : 196
Stream Type : voice+dtmf (1)
Stream Media Addr Type : 1
Negotiated Codec : g711ulaw (160 bytes)
Codec Payload Type : 0
Negotiated Dtmf-relay : rtp-nte
Dtmf-relay Payload Type : 101
QoS ID : -1
Local QoS Strength : BestEffort
Negotiated QoS Strength : BestEffort
Negotiated QoS Direction : None
Local QoS Status : None
Media Source IP Addr:Port: [8.39.25.11]:8080
Media Dest IP Addr:Port : [8.55.0.195]:23022
Local Crypto Suite : AEAD_AES_256_GCM
Remote Crypto Suite : AEAD_AES_256_GCM (
AEAD_AES_256_GCM
AEAD_AES_128_GCM
AES_CM_128_HMAC_SHA1_80
AES_CM_128_HMAC_SHA1_32 )
Local Crypto Key : 3taqc13ClF6BBpvd65WTMPrad/i0uyQ6iNouh+jYHxbf48d4TFmsOGyh4Vs=
Remote Crypto Key : 2/TNTV+Rc1Nh/wbGj0MGwIsLrJ4l+N2jKWGczolEnf7sgsA0Q9AEIz0a4eg=
Mid-Call Re-Assocation Count: 0
SRTP-RTP Re-Assocation DSP Query Count: 0

The following is a sample output for the show command, show ephone offhook . The lines that are added to
the show command output as part of the Unified SRST 12.6 enhancement are local key and remote key.
ephone-1[0] Mac:549A.EBB5.8000 TCP socket:[1] activeLine:1 whisperLine:0 REGISTERED in
SCCP
ver 21/17 max_streams=1 + Authentication + Encryption with TLS connection
mediaActive:1 whisper_mediaActive:0 startMedia:1 offhook:1 ringing:0 reset:0 reset_sent:0
paging 0 debug:0 caps:8
IP:8.44.22.63 * 17872 SCCP Gateway (AN) keepalive 28 max_line 1 available_line 1
port 0/0/0
button 1: cw:1 ccw:(0 0)

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dn 1 number 6901 CM Fallback CH1 CONNECTED CH2 IDLE

Preferred Codec: g711ulaw
Lpcor Type: none Active Secure Call on DN 1 chan 1 :6901 8.44.22.63 18116
to 8.39.25.11 8066 via 8.39.0.1
G711Ulaw64k 160 bytes no vad
SRTP cipher: AES_CM_128_HMAC_SHA1_32
local key: 0OPV0yxvcnRLPMzHfmYbwgHfdxcuS1uPbp5j/Tjk
remote key: e8DQl3Kvk7LjZlipaCoMg9TMreBmiPsFmNiVHwIA
Tx Pkts 0 bytes 0 Rx Pkts 0 bytes 0 Lost 0
Jitter 0 Latency 0 callingDn -1 calledDn -1

Toll Fraud Prevention for SIP Line Side on Unified SRST

Unified SRST Release 12.6 enhances the existing Toll Fraud Prevention feature by enforcing security on the
SIP line side of Unified SRST. The feature enhancement secures the Unified SRST system against potential
toll fraud exploitation by unauthorized users from the SIP line side.

Note Unified SRST 8.1 to 12.5 Releases restricts toll fraud prevention only to securing calls over the SIP
trunk. For more information about Toll Fraud Prevention over a SIP trunk, see Configuring a Trusted
IP Address List for Toll-Fraud Prevention.

Some of the key features of Toll Fraud Prevention on Unified SRST for secure calls over SIP lines are:
• Authenticates all the SIP line messages that are triggered from the endpoints to Unified SRST.
• If the IP address of the endpoint is not part of the IP address trusted list, the call is rejected by Unified
SRST.
• Unified SRST authenticates both IPv4 an IPv6 addresses as part of the toll fraud prevention mechanism.

Prerequisites for Configuring Toll Fraud Prevention for SIP Line Side
• Unified SRST 12.6 or a later version.
• Cisco IOS XE Gibraltar Release 16.11.1a or later.

Configuration Recommendations for Toll Fraud Prevention on Unified SRST

Unified SRST 12.6 enforces security and toll fraud prevention for SIP line side on Unified SRST. The ip
address trusted authentication configuration blocks unauthorized calls from the line side. Hence, the toll
fraud prevention feature secures Unified SRST 12.6 and later from unauthorized users on the line side.
The IP addresses of SRST endpoints are available before registration with Unified SRST, as they are configured
(under voice register pool) for fallback from Unified CM. Hence, it is not mandatory that the endpoints are
registered to Unified SRST for configuring toll fraud prevention.
The IP trust list for Unified SRST is populated based on the IP address information available under voice
register pool configuration mode. You can find the IP address of the SIP endpoints on Unified SRST under
the following commands in voice register pool configuration mode:
• id ip (For example, id ip192.168.0.0 )
• id network (For example, id network192.168.25.0mask255.255.255.0 )

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Configuration Recommendations for Toll Fraud Prevention on Unified SRST

Sometimes, IP addresses of endpoints are not available to Unified SRST before registration. Consider a
scenario where id device-id is the CLI command configured under voice register pool configuration mode to
define the device name. Then, the IP address of the device or endpoint is available to Unified SRST only
during registration.
The following are the configurations of Toll Fraud Prevention in Unified SRST, 12.6:
• The CLI command ip address trusted authentication is enabled by default in Unified SRST. The
command ip address trusted authentication ensures that security is enabled on the Unified SRST
system.
• You can manually configure your Unified SRST endpoints as trusted by entering the IP address or subnet
of the trusted phone under theiptrust-list configuration mode, as follows:

Router#config t
Router(config)#voice service voip
Router(conf-voi-serv)#ip address trusted list
Router(cfg-iptrust-list)#ipv4 192.168.10.0 /16
OR
Router(cfg-iptrust-list)#ipv4 192.168.12.0 255.255.255.0

• You can verify the manually added IP address of the Unified SRST endpoint, as follows:

Router#show running-config | section voice service voip

voice service voip
ip address trusted list
ipv4 192.168.10.1
ipv4 192.168.10.2 255.255.0.0
ipv4 192.168.10.3 255.255.0.0
ipv4 192.168.10.4 255.255.255.0

• The CLI command ip address trusted list under voice service voip configuration mode supports manual
configuration of trusted IP addresses.
• The CLI command show ip address trusted check provides information on whether a particular IP
address is trusted or not.
• The CLI command silent-discard untrustedsip in configuration mode silently discards SIP requests
from untrusted sources. This command is enabled by default on Unified SRST.
• The show ip address trusted list CLI command displays a list of trusted IP addresses. The trusted IP
addresses are displayed under the following lists:
• Dial Peer (only applicable for trunk side): Provides details on the IP address of the trunk that is configured
under the dial-peer configuration mode.
• Configured IP Address Trusted List: Provides details on the manually configured IP addresses that are
trusted.
• Dynamic IP Address Trusted List: Provides details on the IP address of all the phones that are configured
for fallback from Unified CM. This list is introduced in Unified CME 12.6 Release.
• Server Group: Provides details on the IP address of the phones that are configured under server-groups
configuration mode.

Router>enable
Router#show ip address trusted list
IP Address Trusted Authentication
Administration State: UP
Operation State: UP

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Upgrade Considerations

IP Address Trusted Call Block Cause: call-reject (21)

VoIP Dial-peer IPv4 and IPv6 Session Targets:
Peer Tag Oper State Session Target
-------- ---------- --------------
4 UP ipv4:10.65.125.155
Configured IP Address Trusted List:
ipv4 192.168.20.1
ipv4 192.168.20.2 255.255.0.0
ipv4 192.168.20.3 255.255.0.0
ipv4 192.168.20.4 255.255.255.0
Dynamic IP Address Trusted List:
IP Address Subnet Mask Count Reason
-------------------------------------------- --------------- ----- ----------------
ipv4:8.55.0.0 255.255.0.0 1 Pool Configured
ipv4:192.168.0.1 255.255.0.0 2 Pool Configured
ipv6:2001:420:54FF:13::312:0 119 1 Pool Configured
ipv4:8.55.22.15 1 Phone Registered

Note The column Count in Dynamic IP Address Trusted List displays the number of directory numbers (DNs)
sharing the same IP address. For example, ipv4 192.168.0.1 with count 2 represents two DNs sharing
the IP address 192.168.0.1.

Note The output of show ip address trusted list command displays the entry in column Type as ‘Phone
Registered’ if id device-id is configured.

Upgrade Considerations
When you upgrade to Unified SRST 12.6 version, you need not perform extra configurations for supporting
toll fraud prevention. All the endpoints that are manually configured or auto-registered on Unified SRST are
added to the Unified SRST IP Address Trust List. You can view the list of trusted IP addresses under the
output of the CLI command show ip address trusted list.

Configure Toll Fraud Prevention

Configure IP Address Trusted Authentication for Incoming VoIP Calls
Before you begin
• Unified SRST 8.1 or a later version for secure trunk calls.
• Unified SRST 12.6 or a later version for secure line and trunk calls.
• The CLI command silent-discard untrusted needs to be configured for the feature to work

Restrictions
For an incoming VoIP call, IP trusted authentication must be invoked when the IP address trusted authentication
is in “UP” operational state.

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SUMMARY STEPS
1. enable
2. configure terminal
3. voice service voip
4. ip address trusted authenticate
5. ip-address trusted call-block cause
6. end
7. show ip address trusted list

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
Example: • Enter your password if prompted.
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 voice service voip Enters voice service voip configuration mode.
Example:
Router(config)# voice service voip

Step 4 ip address trusted authenticate Enables IP address authentication on incoming H.323 or

SIP trunk calls for toll fraud prevention support.
Example:
Router(conf-voi-serv)# ip address trusted IP address trusted list authenticate is enabled by default.
authenticate Use the no ip address trusted list authenticate command
to disable the IP address trusted list authentication.

Step 5 ip-address trusted call-block cause Issues a cause-code when the incoming call is rejected to
the IP address trusted authentication. This command is
Example:
enabled by default.
Router(conf-voi-serv)#ip address trusted
call-block cause call-reject Note If the IP address trusted authentication fails, a
call-reject (21) cause-code is issued to disconnect
the incoming VoIP call.

Step 6 end Returns to privileged EXEC mode.

Example:
Router()# end

Step 7 show ip address trusted list Verifies a list of valid IP addresses.

Example:
Router# #show ip address trusted list
IP Address Trusted Authentication
Administration State: UP

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Command or Action Purpose

Operation State: UP
IP Address Trusted Call Block Cause:
call-reject (21)

Example
Router>enable
Router#show ip address trusted list
IP Address Trusted Authentication
Administration State: UP
Operation State: UP
IP Address Trusted Call Block Cause: call-reject (21)
VoIP Dial-peer IPv4 and IPv6 Session Targets:
Peer Tag Oper State Session Target
-------- ---------- --------------
Configured IP Address Trusted List:
ipv4 192.168.20.1
ipv4 192.168.20.2 255.255.0.0
ipv4 192.168.20.3 255.255.0.0
ipv4 192.168.20.4 255.255.255.0
Dynamic IP Address Trusted List:
IP Address Subnet Mask Count Type
-------------------------------------------- --------------- ----- ----------------
ipv4:8.55.0.0 255.255.0.0 1 Pool Configured
ipv4:192.168.0.1 255.255.0.0 1 Pool Configured

Add Valid IP Addresses For Incoming VoIP Calls

Before you begin
Cisco Unified CME 8.1 or a later version.

SUMMARY STEPS
1. enable
2. configure terminal
3. voice service voip
4. ip address trusted list
5. ipv4 ipv4 address network mask { <ipv4 address>[ <network mask> ] }
6. end
7. show ip address trusted list

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
Example: • Enter your password if prompted.
Router> enable

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Command or Action Purpose

Step 2 configure terminal Enters global configuration mode.
Example:
Router# configure terminal

Step 3 voice service voip Enters voice service voip configuration mode.
Example:
Router(config)# voice service voip

Step 4 ip address trusted list Enters ip address trusted list mode and allows to manually
add additional valid IP addresses.
Example:
Router(conf-voi-serv)# ip address trusted list
Router(cfg-iptrust-list)#

Step 5 ipv4 ipv4 address network mask { <ipv4 address>[ Allows you to add up to 100 IPv4 addresses in ip address
<network mask> ] } trusted list. Duplicate IP addresses are not allowed in the
ip address trusted list.
Example:
Router(cfg-iptrust-list)#ipv4 172.19.245.1 • network mask — allows to define a subnet IP address.
Router(cfg-iptrust-list)#ipv4 172.19.243.1

Step 6 end Returns to privileged EXEC mode.

Example:
Router(config-register-pool)# end

Step 7 show ip address trusted list Displays a list of valid IP addresses for incoming H.323 or
SIP trunk calls.
Example:
Router# show shared-line

Example
The following example shows three IP addresses configured as trusted IP addresses:
Router#show ip address trusted list
IP Address Trusted Authentication
Administration State: UP
Operation State: UP
IP Address Trusted Call Block Cause: call-reject (21)
Configured IP Address Trusted List:
ipv4 192.168.20.1
ipv4 192.168.20.2 255.255.0.0
ipv4 192.168.20.3 255.255.0.0
ipv4 192.168.20.4 255.255.255.0

Troubleshooting Tips for Toll Fraud Prevention

For troubleshooting toll fraud mechanism supported on Unified SRST, you can enable the CLI commands
debug voip iptrust debug and debug voip iptrust detail, as follows:

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Router#debug voip iptrust

voip iptrust debugging is on
Router#debug voip iptrust detail
voip iptrust detail debugging is on

VRF Support for Unified SRST

Virtual Routing and Forwarding (VRF) for Unified SRST divides a physical router into multiple logical
routers. Each of these logical routers has its own set of interfaces and routing and forwarding tables. VRF
support allows you to bind the Unified SRST feature to a specific VRF. Previously with the Cisco 4000 Series
Integrated Services Routers, Unified SRST was always associated with the global or default routing instance.
From Unified SRST Release 12.8 (Cisco IOS XE 17.2.1r), support is introduced for VRF functionality on
Cisco 4000 Series Integrated Services Router. Before Unified SRST Release 12.8 (Cisco IOS XE 17.2.1r),
support for VRF was available only on Cisco Integrated Services Router Generation 2 platform.
From Unified SRST Release 12.8, the following support is available for VRF:
• VRF for line side on Cisco 4000 Series Integrated Services Routers– Introduced in Unified SRST 12.8
• VRF support for Unified SRST 12.8 and later releases is compatible with SIP trunks that are configured
to use a VRF. However, you can configure different VRFs for the trunk and Unified SRST.

Information About VRF Support

Typically, service providers use a VRF between Provider Edge (PE) and Customer Edge (CE) routers to
provide VPN support for customers. VRF is also used to segment data and voice traffic for improved traffic
management. VRF can be configured on an interface to process incoming packets according to the assigned
VRF.
By configuring VRF-awareness on voice gateways, you can specify a VRF for the voice traffic that is generated
from within the gateway. Voice VRF is added to the VoIP service provider interface (SPI) of the gateway to
send and receive signaling and media packets in the configured VRF. The SPI can send and receive signaling
and media packets only in the configured VRF.

Note We recommend that you configure voice vrf for Unified SRST. For more information, see Design
Recommendations for VRF.

Design Recommendations for VRF

• SIP endpoints supported by Unified SRST, including Cisco IP Phone 7800 Series, Cisco IP Phone 8800
Series, and Cisco Jabber support VRF for Unified SRST.
• VRF support is offered for both secure and nonsecure deployments of Unified SRST.
• Configuring SRST to use a VRF is compatible with both SIP and TDM trunk configurations.
• If Global Bind and voice vrf are configured on the Unified SRST, then preference is given to the Global
Bind.

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• We recommend that
• For SRST line side, configure VRF using voice vrf command.
• For SIP trunk side, configure VRF using bind command configured under voice class tenant
configuration mode and attach the tenant to the required SIP trunk dial-peer.

• VRF Preference Order—The following is the binding preference order for call processing on the trunk
side and line side for SRST:

Preference Bind Configuration

Order

1 Dial-peer bind command is configured under dial-peer configuration mode

Bind
Note This configuration is only for trunk side.

2 Tenant Bind bind command is configured under voice class tenant configuration mode
Note This configuration is only for trunk side.

3 Global Bind bind command is configured under sip in voice service voip configuration
mode.
Note This configuration is both for trunk side and Unified SRST line
side.

4 Voice VRF voice vrf command configuration

Note This configuration is both for trunk side and Unified SRST line
side.

Configuration Examples for VRF

The following is a sample configuration for voice vrf in Unified SRST line side:
vrf definition vrf1
rd 100:101
!
address-family ipv4
exit-address-family

voice vrf vrf1

interface GigabitEthernet0/0/0
vrf forwarding vrf1
ip address 8.44.22.77 255.255.0.0
ip route vrf vrf1 8.0.0.0 255.0.0.0 8.44.0.1

The following is a sample configuration of Global bind (voice service voip). In this case, both Unified SRST
line side and SIP trunks without an explicit binding use the same VRF configuration.

voice service voip

no ip address trusted authenticate
media statistics
media bulk-stats
media disable-detailed-stats

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allow-connections sip to sip

no supplementary-service sip moved-temporarily
no supplementary-service sip refer
supplementary-service media-renegotiate
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
bind all source-interface GigabitEthernet 0/0/0
session transport tcp
min-se 90
session refresh
registrar server expires max 120 min 60
!

Configure Virtual Routing and Forwarding (VRF) for Unified SRST

Before you begin
• Unified SRST 12.8 or a later version.
• For design recommendations, see Design Recommendations for VRF.

SUMMARY STEPS
1. enable
2. configure terminal
3. vrf definition vrf-name
4. rd route-distinguisher
5. address-family ipv4
6. exit-address-family
7. voice vrf vrf-name
8. interface interface-name
9. vrf forwarding customer-vrf-name
10. ip address <ip address> <network mask>
11. ip route vrf vrf-name <ip address> <networkmask> <ip address>
12. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
Example: • Enter your password if prompted.
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 vrf definition vrf-name Creates a VRF with the specified name. In the example,
VRF name is vrf1.
Example:

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Command or Action Purpose

Router(config)# vrf definition vrf1 Note Space is not allowed in VRF name.

Step 4 rd route-distinguisher Creates a VRF table by specifying a route

distinguisher.Enter either an AS number and an arbitrary
Example:
number (xxx:y) or an IP address and arbitrary number
Router (config)# rd 100:101 (A.B.C.D:y).

Step 5 address-family ipv4 Configures IPv4 or IPv6 address-family sessions for a VRF
configuration in Unified SRST.
Example:
Router(config)# address-family ipv4

Step 6 exit-address-family Leaves address-family configuration mode without

removing the address family configuration.
Example:
Router(config)# exit-address-family

Step 7 voice vrf vrf-name Configures a voice VRF in global configuration mode.
Example:
Router(config)# voice vrf vrf1

Step 8 interface interface-name Enters the interface configuration mode.

Example:
Router(config)# interface GigabitEthernet0/0/0

Step 9 vrf forwarding customer-vrf-name Associates the customer VRF instance with the tunnel.
Packets exiting the tunnel are forwarded to this VRF (inner
Example:
IP packet routing).
Router(config-if)# vrf forwarding vrf1

Step 10 ip address <ip address> <network mask> IP address is assigned to the interface.
Example:
Router(config-if)# ip address 8.44.22.77
255.255.0.0

Step 11 ip route vrf vrf-name <ip address> <networkmask> (Optional) Generates IP routing information associated
<ip address> with a VRF.
Example: Note Required only if you need to add static routes.
Router(config-if)# ip route vrf vrf1 8.0.0.0
255.0.0.0 8.44.0.1

Step 12 end Exits to privileged EXEC mode.

Example:
Router(config-if)# end

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IPv6 Support for Unified SRST SIP IP Phones

IPv6 Support for Unified SRST SIP IP Phones

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.
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.
For an example of configuring IPv6 Support on Unified SRST, see Examples for Configuring IPv6 Pools for
SIP IP Phones.
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).

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• 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.
• 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
2. configure terminal
3. ipv6 unicast-routing
4. voice service voip
5. sip
6. no ant
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
17. voice register global
18. default mode
19. max-dn max-directory-numbers
20. max-pool max-voice-register-pools
21. exit

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22. voice register poolpool-tag

23. id { network address mask mask | ip address mask mask | mac address }
24. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
Example: • Enter your password if prompted.
Router> enable

Step 2 configure terminal Enters global configuration mode.

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:
Router (config)# voice service voip voip —Specifies Voice over IP (VoIP) parameters.

Step 5 sip Enters SIP configuration mode.

Example:
Router(config-voi-serv)# sip

Step 6 no ant 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

Step 9 exit Exits voice service voip configuration mode.

Example:
Router(config-voi-serv)# exit

Step 10 sip-ua Enters SIP user-agent configuration mode.

Example:
Router(config)# sip-ua

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Command or Action Purpose

Step 11 protocol mode {ipv4 | ipv6 | dual-stack [preference Allows phones to interact with phones on IPv6 voice
{ipv4 | ipv6}]} gateways. You can configure phones for IPv4 addresses,
IPv6 addresses, or for a dual-stack mode.
Example:
Router(config-sip-ua)# protocol mode dual-stack • ipv4—Allows you to set the protocol mode as an IPv4
preference ipv6 address.
• 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.

Step 12 exit Exits SIP configuration mode.

Example:
Router(config-sip-ua)# exit

Step 13 voice service {voip} Enters voice-service configuration mode to specify a voice
encapsulation type.
Example:
Router (config)# voice service voip voip—Specifies Voice over IP (VoIP) parameters.

Step 14 sip Enters SIP configuration mode.

Example:
Router(config-voi-serv)# sip

Step 15 no call service stop Activates SIP call service.

Example:
Router(config-serv-sip)# call service stop

Step 16 exit Exits SIP configuration mode.

Example:
Router(config-serv-sip)# exit

Step 17 voice register global Enters voice register global configuration mode to set
parameters for all supported SIP phones in Unified SRST.
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.
Example:

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Command or Action Purpose

Router(config-register-global)# max-dn 50 Maximum number is platform and version-specific. Type
? for value.

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 poolpool-tag Enters voice register pool configuration mode to set
phone-specific parameters for a SIP phone.
Example:
Router(config)# voice register pool 1

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

Router(config-register-pool)# id network
10.64.88.0 mask 255.255.255.0

Step 24 end Exits to privileged EXEC mode.

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.

Before you begin

• 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.

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• You need to ensure that your router is in default mode (for Unified SRST).

SUMMARY STEPS
1. enable
2. configure terminal
3. voice service voip
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
12. voice register global
13. default mode
14. max-dn max-directory-numbers
15. max-pool max-voice-register-pools
16. exit
17. voice register pool pool-tag
18. id [network address mask mask | ip address mask mask]
19. dtmf-relay rtp-nte
20. no vad
21. codec codec-type [bytes]
22. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
Example: • Enter your password if prompted.
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

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

Step 4 allow-connections from-type to to-type Allows connections between specific types of endpoints
in a VoIP network.
Example:

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Command or Action Purpose

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 refer 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

Step 8 sip Enters SIP configuration mode.

Example: Required only if you perform the following step for
Router(config-voi-serv)# sip enabling the SIP registrar function.

Step 9 registrar server [expires[max sec ][min sec ]] Enables SIP registrar functionality in Unified SRST.
Example: • expires : (Optional) Sets the active time for an
Router(config-serv-sip)# registrar server incoming registration.
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.
• min sec : (Optional) Minimum expiration time for a
registration, in seconds. The range is from 60 to 3600.
The default is 60.

Step 10 exit Exits SIP configuration mode.

Example:
Router(config-serv-sip)# exit

Step 11 exit Exits voice-service configuration mode.

Example:
Router(config-voi-serv)# exit

Step 12 voice register global Enters voice register global configuration mode to set
parameters for all supported SIP phones in Unified SRST.
Example:
Router(config)# voice register global

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Command or Action Purpose

Step 13 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 14 max-dn max-directory-numbers Limits number of directory numbers to be supported by

this router.
Example:
Router(config-register-global)# max-dn 50 Maximum number is platform and version-specific. Type
? for value.

Step 15 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 Maximum number is platform and version-specific. Type
? for value.

Step 16 exit Exits voice register global configuration mode.

Example:
Router(config-register-global)# exit

Step 17 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

Step 18 id [network address mask mask | ip address mask Enters voice service voip configuration mode.
mask]
Example:
Router(config)# voice service voip

Step 19 dtmf-relay rtp-nte Forwards DTMF tones by using Real-Time Transport

Protocol (RTP) with the Named Telephone Event (NTE)
Example:
payload type and enables DTMF relay using the RFC 2833
Router(config-register-pool)# dtmf-relay standard method.
rtp-nte

Step 20 no vad Disables voice activity detection (VAD) on the VoIP dial
peer.
Example:
Router(config-register-pool)# no vad VAD is enabled by default. Because there is no comfort
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 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.
Example:
The codec - type argument specifies the preferred codec
Router(config-register-pool)# codec g729r8 and can be one of the following:
• g711alaw: G.711 a–law 64,000 bps.
• g711ulaw: G.711 mu–law 64,000 bps.

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Command or Action Purpose

• g729r8: G.729 8000 bps (default).

The bytes argument is optional and specifies the number

of bytes in the voice payload of each frame

Step 22 end Returns to privileged EXEC mode.

Example:
Router()# end

Configure Voice Hunt Groups on Unified SRST

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.
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 33

Before you begin

• Cisco IOS XE Denali 16.3.1 or later versions.
• Shared Lines are not supported on Unified SRST.

SUMMARY STEPS
1. enable
2. configure terminal
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 [secondarysecondary-order]
8. hops number
9. timeout seconds
10. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.

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Command or Action Purpose

Example: • Enter your password if prompted.
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 voice hunt-group hunt-tag [longest-idle | parallel | peer Enters voice hunt-group configuration mode to define a
| sequential] hunt group.
Example: • hung-tag —Unique sequence number of the hunt
Router(config)# voice hunt-group 1 longest-idle group to be configured. Range is 1 to100.
• 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:
Router(config-voice-hunt-group)# pilot number • number—String of up to 16 characters that represents
8100 an E.164 phone number.
• 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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Command or Action Purpose

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,
Example:
use the no form of this command.
Router(config-voice-hunt-group)# list 8000,
8010, 8020, 8030 • number—List of extensions to be added as members
to 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.
Example: • If a final number in one hunt group is configured as
Router(config-voice-hunt-group)# final 8888 a pilot number of another hunt group, the pilot number
of the first hunt group cannot be configured as a final
number in any other hunt group.

Step 7 preference preference-order [secondarysecondary-order] Sets the preference order for the directory number
associated with a voice hunt-group pilot number.
Example:
Router(config-voice-hunt-group)# preference 6 Note We recommend that the parallel hunt-group
pilot number be unique in the system. Parallel
hunt 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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Command or Action Purpose

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
Example:
the next number in a peer or longest-idle voice hunt group
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
Example:
a voice hunt-group list. Default is 180 seconds.
Router(config-voice-hunt-group)# timeout 100

Step 10 end Exits to privileged EXEC mode.

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 the 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
2. configure terminal
3. voice register pool tag
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

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
Example: • Enter your password if prompted.
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 voice register pool tag Enters voice register pool configuration mode.
Example: • Use this command to control which phone registrations
Router(config)# voice register pool 15 are accepted or rejected by a Cisco Unified SIP SRST
device.

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
Example:
non-SIP station extension (that is, SIP trunk, H.323 trunk,
Router(config-register-pool)# call-forward 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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Configure Call Blocking Based on Time of Day, Day of Week, or Date

Command or Action Purpose

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:
Router(config-register-pool)# call-forward • directory-number : Phone number to which calls are
b2bua busy 5006 forwarded. Represents a fully qualified E.164 number.
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:
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. Maximum length of the phone number is 32.

Step 7 call-forward b2bua noan directory- number timeout Enables call forwarding for a SIP B2BUA so that incoming
seconds calls to an extension that does not answer after a configured
amount of time are forwarded to another extension.
Example:
Router(config-register-pool)# call-forward This command is used if a phone is registered with a Cisco
b2bua noan 5010 timeout 10 Unified SIP SRST router, but the phone is not reachable
because there is no IP connectivity (there is no response to
Invite requests).
• directory-number : Phone number to which calls are
forwarded. Represents a fully qualified E.164 number.
Maximum length of the phone number is 32.
• 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.

Step 8 end Returns to privileged EXEC mode.

Example:
Router(config-register-pool)# end

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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Configure Call Blocking Based on Time of Day, Day of Week, or Date

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
2. configure terminal
3. call-manager-fallback
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
8. voice register pool tag
9. after-hour exempt
10. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
Example: • Enter your password if prompted.
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 call-manager-fallback Enters call-manager-fallback configuration mode.

Example:
Router(config)# call-manager-fallback

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:

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Command or Action Purpose

Router(config-cm-fallback)# after-hours block • If the 7-24 keyword is specified, the pattern is always
pattern 1 91900
blocked, 7 days a week, 24 hours a day.
• 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
Example:
patterns that are defined using the after-hours block
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 stop-time Defines a recurring time period based on month and date
during which calls are blocked to outgoing dial patterns
Example:
that are defined using the after-hours block pattern
Router(config-cm-fallback)# after-hours date command.
jan 1 00:00 00:00
• month : Month abbreviation. The following are valid
month abbreviations: jan, feb, mar, apr, may,jun,
jul, aug, sep, oct, nov,dec.
• date : Date of the month. Range is from 1 to 31.
• start-time stop-time : Beginning and ending times
for call blocking, in an HH:MM format using a
24-hour clock. The stop time must be larger than the
start time.
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.

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Verification

Command or Action Purpose

Step 7 exit Exits call-manager-fallback configuration mode.
Example:
Router(config-cm-fallback)# exit

Step 8 voice register pool tag Enters voice register pool configuration mode.
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 9 after-hour exempt Specifies that for a particular voice register pool, none of
its outgoing calls are blocked although call blocking is
Example:
enabled.
Router(config-register-pool)# after-hour exempt

Step 10 end Returns to privileged EXEC mode.

Example:
Router(config-register-pool)# end

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 : 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 Command
Reference (All Versions).

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.
No configuration is necessary.

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

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Enabling KPML for SIP Phones

2. configure terminal
3. no telephony-service
4. call-manager-fallback
5. moh enable-g711 "bootflash: filename"
6. moh enable-g729 "bootflash: filename"
7. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
Example: • Enter your password if prompted.
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 no telephony-service Removes all the configurations for IP phones configured

under the telephony-service configuration mode.
Example:
Router# no telephony-service

Step 4 call-manager-fallback Enters call-manager-fallback configuration mode.

Example:
Router(config)# call-manager-fallback

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.
Example:
Router(config-cm-fallback)# moh enable-g711 SRST.
"bootflash:music-on-hold.au"

Step 6 moh enable-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"

Step 7 end Returns to privileged EXEC mode.

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.

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Disabling SIP Supplementary Services for Call Forward and Call Transfer

SUMMARY STEPS
1. enable
2. configure terminal
3. voice register pool pool-tag
4. digit collect kpml
5. end
6. show voice register dial-peers

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
Example: • Enter your password if prompted.
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 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 4 • pool-tag: Unique sequence number of the SIP phone
to 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.
Example: Note This command is enabled by default for
Router(config-register-pool)# digit collect supported phones in Cisco Unified CME and
kpml Cisco Unified SRST.

Step 5 end Exits to privileged EXEC mode.

Example:
Router(config-register-pool)# end

Step 6 show voice register dial-peers Displays details of all dynamically created VoIP dial peers
associated with the Cisco Unified CME SIP register
Example:
including the defined digit collection method.
Router# show voice register dial-peer

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.

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SUMMARY STEPS
1. enable
2. configure terminal
3. voice service voip OR dial-peer voice tag voip
4. no supplementary-service sip {moved-temporarily |refer}
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
Example: • Enter your password if prompted.
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 voice service voip OR dial-peer voice tag voip Enters voice-service configuration mode to set global
parameters for VoIP features.
Example:
Router(config)# voice service voip or
or
Router(config)# dial-peer voice 99 voip Enters dial peer configuration mode to set parameters for
a specific dial peer.

Step 4 no supplementary-service sip {moved-temporarily Disables SIP call forwarding or call transfer supplementary
|refer} services globally or for a dial peer.
Example: • moved-temporarily: SIP redirect response for call
Router(conf-voi-serv)# no supplementary-service forwarding.
sip refer
or • refer: SIP REFER message for call transfers.
Router(config-dial-peer)# no
supplementary-service sip refer • Sending REFER and redirect messages to the
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.

Step 5 end Exits to privileged EXEC mode.

Example:
Router(config-voi-serv)# end
OR
Router(config-dial-peer)# end

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Configuring idle Prompt Status for SIP Phones

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.

SUMMARY STEPS
1. enable
2. configure terminal
3. voice register global
4. system message string
5. end
6. show voice register global

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
Example: • Enter your password if prompted.
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 voice register global Enters voice register global configuration mode to set global
parameters for all supported SIP phones in a Cisco Unified
Example:
CME environment.
Router(config)# voice register global

Step 4 system message string Defines a status message that displays on SIP phones
registered to Cisco Unified SRST.
Example:
Router(config-register-global)# system message • string: Up to 32 alphanumeric characters. Default is
fallback active “CM Fallback Service Operating.”

Step 5 end Exits to privileged EXEC mode.

Example:
Router(config-register-global)# end

Step 6 show voice register global Displays all global configuration parameters associated with
SIP phones.
Example:

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Examples

Command or Action Purpose

Router# show voice register global

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
!
!
voice register global
default mode
max-dn 40
max-pool 40
!
voice register pool 1
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 20 longest-idle
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)# hops 6
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

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

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
under 'voice service voip', 'sip' submode

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.

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Example for Configuring Music On Hold for Unified SIP SRST

call-manager-fallback
after-hours block pattern 1 91
after-hours block pattern 2 9011
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:
voice register pool 1
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
call-manager-fallback
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
voice register pool 15
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
voice register global
system message fallback active
end
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

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Example for Disabling SIP Supplementary Services for Call Forward and Call Transfer

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

end

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 CHAPTER 4
Enhanced SRST
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.
• Migration from Cisco Unified SRST Manager to Unified E-SRST, on page 131
• Licensing, on page 133
• Toll Fraud Prevention for SIP Line Side on Unified E-SRST, on page 136
• Unified E-SRST with Support for Voice Hunt Group, on page 136
• SIP: Configure Unified E-SRST, on page 138
• SCCP: Configure Unified E-SRST, on page 153
• Configure Digest Credentials on Cisco Unified Communications Manager, on page 159

Migration from Cisco Unified SRST Manager to Unified E-SRST

Cisco Unified Survivable Remote Site Telephony Manager is End-of-Life (EOL). Hence, provisioning for
Unified E-SRST through Cisco 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 Cisco Unified Communications Manager. For more information on configuring
Unified E-SRST see SIP: Configure Unified E-SRST and SCCP: Configure Unified E-SRST.
For information on Cisco Unified Survivable Remote Site Telephony Manager End-of-Life announcement,
see Cisco Unified Survivable Remote Site Telephony Manager Product Bulletin.
Cisco Unified SRST Manager is a GUI-based tool that helps to monitor, report, and troubleshoot remote sites.
It performs automatic sync up between the Cisco 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 Cisco
Unified SRST Manager that is End-of-Life, see Administration Guide for Cisco Unified SRST Manager.

Benefits
When you configure Unified E-SRST, it provides the following feature benefits in comparison to the classic
Cisco Unified SRST:
• Voice Hunt Group
• Shared Lines

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Restrictions

• 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.
For more information on configuring Shared Line, BLF, and Video with Unified E-SRST, see SIP: Configure
Unified E-SRST.

Restrictions
• Supports the Version Negotiation feature only on the Cisco Unified 9951, 9971, 8961 SIP IP phones,
Cisco IP Phone 7800, and 8800 Series.
• The phone firmware version is 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, configure the individual voice register
Pools.

Restrictions for Unified E-SRST, Release 12.2

The Unified E-SRST deployment with the voice hunt group has the following restrictions:
• Does not support the auto logout.
• Does not support Programmable Line Keys (PLK).
• Does not support HLog Softkey.

Note The existing support for Cisco Jabber is now End of Life (EOL). Hence, does not support Cisco Jabber
on Cisco Unified SRST, Unified E-SRST.

Support for Cisco Unified IP Phones and Platforms

The following section provides information about platform support for Cisco Unified IP Phones:
• Unified E-SRST is supported on Cisco 1100 Series Integrated Services Router (ISR) Platforms with
Cisco IOS XE Bengaluru 17.5.1a and later releases.

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Licensing

• Unified E-SRST is supported on Cisco 4000 Series ISR Platforms (4321, 4331, 4351, 4431, and 4451)
on all Cisco IOS XE releases.
• Unified E-SRST is supported on Cisco 4461 Series ISR Platforms with Cisco IOS XE 16.10.1a and later
releases.
• Unified E-SRST is supported on Cisco Catalyst 8300 Series Edge Platforms with Cisco IOS XE
Amsterdam 17.3.2 and later releases.
• Unified E-SRST is supported on Cisco Catalyst 8200 Series Edge Platforms with Cisco IOS XE Bengaluru
17.4.1a and later releases.
• Unified E-SRST is supported on Cisco Catalyst 8200L Series Edge Platforms with Cisco IOS XE
Bengaluru 17.5.1a and later releases.

Licensing
This section provides information on licensing of Cisco Unified Enhanced Survivable Remote Site Telephony
(Unified E-SRST).

Cisco Smart Licensing for Unified E-SRST

Cisco Smart Licensing is a software licensing model that provides visibility of ownership and usage through
the Cisco Smart Software Manager (CSSM) portal. CSSM is a central license repository that manages licenses
across all Cisco products that you own, including Unified E-SRST. Devices send license usage to CSSM
either directly or use an on-premises satellite. Your Smart Account Administrator controls your access to
CSSM. Use your Cisco credentials to access the CSSM portal using http://software.cisco.com.
Smart Licensing applies to all platform technology (UCK9, Security) and Unified E-SRST feature licenses
that the router uses. Unified E-SRST requires one license entitlement (SRST_E_EP) for each configured SIP
or SCCP phone.
CSSM shows license usage across all registered devices to a virtual account. A Virtual Account License
Inventory displays the quantity of licenses that are purchased, those licenses in use, and a balance. An
Insufficient Licenses alert is displayed if the license balance is below 0.
For example, consider a smart account in CSSM with 50 SRST_E_EP licenses. If you have a single registered
Unified E-SRST router with 20 configured phones, the CSSM licenses page shows Purchased as 50, In Use
as 20 and Balance as 30.
For more information on Smart Software Manager, see the Cisco Smart Software Manager User Guide.

Note The SRST_E_EP license count reflects the total phone count for both the ephones and voice register
Pools that are configured in the Unified E-SRST irrespective of registered or nonregistered phones.
Reports license usage three minutes after the last configuration change, to avoid unnecessary reporting
while configuring Unified E-SRST.

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Smart License Operation

Note Unified E-SRST Smart Licenses also provide RTU entitlement for routers that are not configured for
Smart Licensing.

Smart License Operation

Cisco IOS XE Everest 16.5.1 Release to Cisco IOS XE Fuji 16.9.1 Release
Cisco 4000 Series Integrated Services Routers support Smart Licensing as an alternative to Cisco Software
RTU Licensing. Use the license smart enable command to enable Smart Licensing. To disable Smart Licensing,
use the no form of the command and re-accept the EULA using the license accept end user agreement
command.

Cisco IOS XE Gibraltar 16.10.1 Release Onwards

The Cisco RTU Licensing and the CLI license smart enable command are deprecated. Smart Licensing is
mandatory from this release.

Cisco IOS XE Everest 16.5.1 Release to Cisco IOS XE Amsterdam 17.3.1a Release
Routers configured to use Smart Licensing offer a 90-day evaluation period, during which you can use all the
features without registering to CSSM. A Unified E-SRST device is associated with CSSM using a registration
token. You can obtain the registration token from the virtual CSSM account or from an on-premises satellite.
Once registered, the evaluation period pauses and you can use the balance later. You cannot renew the
evaluation period on its expiry.

Warning Unified E-SRST shuts down when the router is unregistered and allowed to pass into the Evaluation
Expired state.

To register the Unified E-SRST router with CSSM, use license smart register idtoken command. For
information on registering the device with CSSM, see Software Activation Configuration Guide.
Upon successful registration, the device sends an authorization request to CSSM for the licenses in use. For
each license type requested, if the Smart Account has sufficient licenses, CSSM responds with Authorized.
If the Smart Account does not have sufficient licenses, CSSM responds with Out of Compliance.
Post successful authorization of the request, licenses are bound to the requesting device until the next
authorization request submission. An authorization request is sent every 30 days or when there is any change
in license consumption, to maintain the registration with CSSM. The authorization expires if you do not update
the license request for the router within 90 days. The certificate issued to identify the router at the time of
registration is valid for one year and renewed every six months.
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

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Cisco IOS XE Gibraltar 16.12.1 Release to Cisco IOS XE Amsterdam 17.3.1a Release

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
-----------------------------------------------------------------------------
ISR_4351_UnifiedCommun.. (ISR_4351_UnifiedCommun..) 1 AUTHORIZED
SRST v12 Endpoint Li... (SRST_EP) 4 AUTHORIZED

Cisco IOS XE Gibraltar 16.12.1 Release to Cisco IOS XE Amsterdam 17.3.1a Release
Cisco 4000 Series Integrated Services Routers supports Specific License Reservation (SLR). SLR allows
reservation and utilization of Cisco Smart Licenses without communicating the license information to CSSM.
To reserve specific licenses for a device, generate the request code from the device. Enter the request code in
CSSM along with the required licenses and their quantity, and generate authorization code. Enter the
authorization code on the device to map the license to the Unique Device identifier (UDI).

Cisco IOS XE Amsterdam 17.3.2 and Cisco IOS XE Bengaluru 17.4.1a Release Onwards
This release introduces a new paradigm for tracking license usage across your business. In earlier releases,
license authorization was forward looking, binding licenses to a device until the next authorization request.
Actual license usage during the proceeding reporting period is sent to CSSM, allowing you to plan ongoing
license requirements based on historical usage data. Initial device registration is no longer required to use
most platform functionality and deprecates the evaluation period.
Submits the license usage reports periodically according to a minimum reporting policy set for your account.
Typically, this period could be once per year. However, you can generate reports more frequently if the use
of licensed features varies over time. CSSM acknowledges each Resource Utilization Monitoring (RUM)
report to ensure reliable recording of the usage. If the router does not receive an acknowledgment within the
minimum reporting period, disables the call processing. Resumes the call processing on receiving a valid
acknowledgment.
Submit the reports directly to the CSSM or through a satellite. Cisco Smart Licensing Utility (CSLU)
applications can also receive usage reports, providing you with more flexibility in managing your license
usage. Also, when a device is not able to communicate directly with a licensing server, a signed usage report
can be generated and manually uploaded to CSSM. The acknowledgment generated by CSSM must be uploaded
to the device within the license reporting policy period to ensure continued use.
As license reporting is now based on historical usage, the registration process used previously has been replaced
with a trust association that also defines the reporting policy set in your account. Establishing trust with CSSM
or Cisco Smart Software Manager Satellite uses an identity token similar to earlier registrations. Use the
license smart trust idtoken token command to establish the trust relationship within the initial reporting
period set for the device. The CLI license smart register command is deprecated from this release.
Current license usage for Unified E-SRST is displayed using the show license summary command:
Router# sh license summary
License Usage:

License Entitlement tag Count Status

----------------------------------------------------------------------------
securityk9 (ISR_4400_Security) 1 IN USE
AdvUCSuiteK9 (ISR_4400_AdvancedUCSuite) 1 IN USE
uck9 (ISR_4400_UnifiedCommun...) 1 IN USE
SRST_E_EP (SRST_E_EP) 8 IN USE
SRST_EP (SRST_EP) 592 IN USE

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Toll Fraud Prevention for SIP Line Side on Unified E-SRST

Toll Fraud Prevention for SIP Line Side on Unified E-SRST

Unified E-SRST Release 12.6 enhances the existing Toll Fraud Prevention feature by enforcing security on
the SIP line side of Unified E-SRST. The feature enhancement secures the Unified E-SRST system against
potential toll fraud exploitation by unauthorized users from the SIP line side.
The configuration and characteristics of toll fraud prevention offered on the SIP line side of Unified E-SRST
is same as the support available on Cisco Unified SRST. For more information on the feature, see Toll Fraud
Prevention for SIP Line Side on Unified SRST.

Unified E-SRST with Support for Voice Hunt Group

The Unified E-SRST Release 12.2 supports the Voice Hunt Group with Cisco Unified Enhanced Survivable
Remote Site Telephony (Unified E-SRST). The deployment supports the SIP and SCCP phones. The Cisco
IP Phone 7800 and 8800 Series are the supported SIP phones for this deployment. The Unified E-SRST
deployment introduces the voice hunt group enhancement on the Cisco 4000 Series Integrated Services Routers.
As part of the enhancement, supports the voice hunt group features in the E-SRST mode. The Unified E-SRST
12.2 and later releases supports the voice hunt group deployments with Sequential, Parallel, Longest idle, and
Peer call blasting.
During a WAN outage, the SIP phones on the Cisco Unified Communications Manager (Cisco Unified
Communications Manager) fallback to Unified E-SRST router in mode esrst. By default, logs the SIP phones
in to the hunt group. However, if the CLI command members logout is configured under the voice hunt group
configuration mode, the phones are in logged out state. In the Unified E-SRST mode, the phone that falls
back on Unified E-SRST can toggle state. It can also log in (or log out) to the voice hunt group using HLog
in Feature Access Code (FAC). Displays the DN status (logged in or logged out) on the registered phones
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 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 different across a
deployment with the common voice register Pool configuration and the individual voice register Pool
configuration.
• Common Voice Register Pool Configuration: The DN's log out individually, and not at the phone level.
• Individual Voice Register Pool Configuration: The DN's log out at the phone level, irrespective of the
user providing the DN (primary, secondary, and so on) from which FAC input.

When the WAN is available, the phones register back with Cisco 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.
The Unified E-SRST 12.2 Release introduces support for the voice hunt group with shared lines and mixed
shared lines (SCCP and SIP phones). For a mixed shared line supported with the voice hunt group, configure
only individual voice register Pools. Does not support the common voice register Pools. For a sample

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Support for B-ACD in Unified E-SRST

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.
Also, supports hunt statistic collection for Unified E-SRST 12.2 and later releases.
A mixed deployment of SIP and SCCP phones supports the Unified E-SRST, Release 12.2. Supports Hunt
Group Logout from a mixed deployment of SIP and SCCP phones using:
• FAC
• Feature Button, or DND

Supports Line level logout and phone level log out using FAC (*4).

Note Does not support Hunt Group logout for shared lines. Shared lines retain their logged in status.

Support for B-ACD in Unified E-SRST

The Unified E-SRST Release 12.2 enhancement supports B-ACD. For SIP phones that fall back to Unified
E-SRST router in mode esrst, you must ensure that the CLI command members logout is configured. The
Members Logout functionality handles the 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.

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 Cisco Unified Communications Manager, a common
voice register Pool configuration and 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.
• 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. This scenario occurs where the IP address of the phone
changes 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 Cisco Unified Communications Manager.

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SIP: Configure Unified E-SRST

• 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 Cisco
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 must configure the CLI command fac custom
under telephony-service configuration mode.
• Ensure that the CLI commands call-park system application andhunt-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.
For a sample configuration of voice hunt groups on Unified E-SRST, see Example for Configuring Unified
E-SRST with Voice Hunt Group Enhancements.

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 E-SRST mode. In the current scenario, when
the SIP phones fall back to the SRST mode, disables features such as Shared-Line, Busy-Lamp-Field (BLF),
and Video call on the phones. The SRST mode does not support these features. 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

The following table contains a list of supported features and the expected behavior of the features in the
E-SRST mode.

Feature Supported Features Expected Behavior in the 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 monitoring Not supported (Not supported in RT

phones)

BLF call-list monitoring Supported

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Restrictions

Feature Supported Features Expected Behavior in the E-SRST

Mode

Monitoring of a Call-park slot Not supported

Monitoring of Paging DN Not supported

Monitoring of Conference Not supported

DN

• To enable version negotiation feature between ESRST & phone, you must configure "mode esrst" under
the voice register global mode.
• We recommended using the 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
• Supports the Version Negotiation feature only on the Cisco Unified 9951, 9971, 8961 SIP IP phones,
Cisco IP Phone 7800, and 8800 Series.
• The phone firmware version is 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, configure the individual voice register
Pools.

Enable the E-SRST Mode

SUMMARY STEPS
1. enable
2. configure terminal
3. voice register global
4. mode esrst
5. exit

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
Example: • Enter your password if prompted.
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:

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Command or Action Purpose

Router# configure terminal

Step 3 voice register global Enters the voice register global configuration mode to set
the parameters for all the supported SIP phones in Cisco
Example:
Unified Communications Manager Express.
Router(config)# voice register global

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:
Router(config-register-global)# exit

Configure SIP shared-line

To configure SIP shared-line, perform the following procedure:

SUMMARY STEPS
1. enable
2. configure terminal
3. voice register dn dn-tag
4. shared-line [max-calls number-of-calls ]
5. huntstop channel number-of-channels
6. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
Example: • Enter your password if prompted.
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 voice register dn dn-tag

Step 4 shared-line [max-calls number-of-calls ]
Step 5 huntstop channel number-of-channels
Step 6 end Returns to privileged EXEC mode.

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Configure BLF

Configure BLF
Before you begin
To enable the version negotiation feature in the Unified E-SRST mode, perform the following procedure.

SUMMARY STEPS
1. enable
2. configure terminal
3. sip-ua
4. presence enable
5. exit
6. max-subscription number
7. presence call-list
8. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
Example: • Enter your password if prompted.
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 sip-ua
Step 4 presence enable
Step 5 exit
Step 6 max-subscription number
Step 7 presence call-list
Step 8 end

Enable a SIP Directory Number to Be Watched

To enable a directory number to be watched, perform the following procedure:

SUMMARY STEPS
1. enable
2. configure terminal
3. voice register dn dn-tag
4. numbernumber

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Enable BLF on a Voice Register Pool

5. allow watch
6. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable
Step 2 configure terminal
Step 3 voice register dn dn-tag
Step 4 numbernumber
Step 5 allow watch
Step 6 end

Enable BLF on a Voice Register Pool

To enable BLF on a voice register pool, perform the following steps:
For configuration information, see the Cisco Unified Communications Manager Administration Guide.

SUMMARY STEPS
1. enable
2. configure terminal
3. voice register pool pool-tag
4. number tagdn dn-tag ]
5. blf-speed-dial tag numberlabelstring[device]
6. presence call-list(To enable Presence feature for all the missed/received/placed calls)
7. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable
Step 2 configure terminal
Step 3 voice register pool pool-tag
Step 4 number tagdn dn-tag ]
Step 5 blf-speed-dial tag numberlabelstring[device]
Step 6 presence call-list(To enable Presence feature for all the
missed/received/placed calls)
Step 7 end

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Example: ESRST Mode

Example: ESRST Mode

The following example shows how to enable the E-SRST mode:
Router# configure terminal
Router(config)# voice register global
Router(config-register-global)# mode esrst

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)#voice register pool 2
Router(config-register-pool)#id mac 000D.39F9.3A58
Router(config-register-pool)#type 7965
Router(config-register-pool)#number 1 dn 1

Example: Configuring BLF

The following example shows how to configure BLF:
Router(config)#voice register dn 1Router (config-register-dn)#number 1111Router
(config-register-dn)#allow watchRouter(config)#voice register dn 1Router
(config-register-dn)#number 2222Router(config)#voice register pool
1Router(config-register-pool)#id mac 0015.6247.EF90Router(config-register-pool)#type
7971Router(config-register-pool)#number 1 dn 1Router(config)#voice register pool
2Router(config-register-pool)#id mac 0012.0007.8D82Router(config-register-pool)#type
7912Router(config-register-pool)#number 1 dn 2Router(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. Configure the digest credentials on Cisco 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 in Cisco
Unified SRST Release 12.2.

Configure Unified E-SRST

The mode esrst under telephony-service and voice register global configuration mode supports SCCP and
SIP phones respectively to enable the enhanced services in Unified E-SRST mode. While Cisco Unified SRST
supports only the basic voice hunt group features, Unified E-SRST supports the 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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2. configure terminal
3. telephony-service
4. mode esrst
5. max-ephones max-phones
6. max-dnmax-directory-numbers
7. ip source-address ip-address [ port port] [any-match | strict-match]
8. call-park system {application |redirect}
9. hunt-group logout {DND | HLog}
10. transfer-system full-consult
11. transfer-pattern transfer-pattern
12. fac { standard | custom { alias alias-tag | feature } }
13. create cnf-files
14. exit
15. voice register global
16. mode esrst
17. max-dn max-directory-numbers
18. max-pool max-phones
19. exit
20. voice register dn dn-tag
21. number number
22. exit
23. voice register pool pool-tag
24. id [{network address mask mask | ip address mask mask | mac address}] [device-id-name
devicename]
25. dtmf-relay rtp-nte
26. exit

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
Example:
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service

Step 4 mode esrst Configures the E-SRST mode under the telephony-service
configuration mode.
Example:
Router(config)# telephony-service

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Command or Action Purpose

Step 5 max-ephones max-phones Configures the maximum supported IP phones by the
router. The default is 0.
Example:
Router(config-telephony)# max-ephones 40 The maximum number is platform-dependent.

Step 6 max-dnmax-directory-numbers Sets the maximum supported directory numbers (DNs) by

the router.
Example:
Router(config-telephony)# max-dn 15 • Max-directory-numbers: Maximum supported
directory numbers (DNS) or virtual voice ports by
the router. The maximum number is
platform-dependent. The default is 0.

Step 7 ip source-address ip-address [ port port] [any-match | Enables the router to receive messages from the Cisco IP
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.
Example: • application : Enables the Call Park features supported
Router(config-telephony)# call-park system in Cisco Unified SRST.
application

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.

Example: • full-consult—Calls are transferred with consultation
Router(config-telephony)# transfer-system using H.450.2 standard methods and a second phone
full-consult line if available. Calls fall back to full-blind if the
second line is unavailable.

Step 11 transfer-pattern transfer-pattern Allows transfer of the phone calls by Cisco Unified IP
phones to specified phone number patterns. If you have
Example:
set no transfer pattern, defaults to other local IP phones.
Router(config-telephony)# transfer-pattern .T
• transfer-pattern—A string of digits for permitted Call
Transfers.

Step 12 fac { standard | custom { alias alias-tag | feature } } Enables all standard feature access codes (FACs) or creates
and enables individual custom FACs in telephony-service
Example:
configuration mode.
Router(config-telephony)# fac standard

Step 13 create cnf-files Builds the required XML configuration files for IP phones
in the telephony-service configuration mode.
Example:

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Command or Action Purpose

Router(config-telephony)# create cnf-files
version-stamp

Step 14 exit Exits the telephony-service configuration mode

Example:
Router(config-telephony)# exit

Step 15 voice register global Enter the voice register global configuration mode.
Example:
Router(config)# voice register global

Step 16 mode esrst Configures the E-SRST mode under the voice register
global mode.
Example:
Router(config-register-global)# mode esrst

Step 17 max-dn max-directory-numbers Set the maximum supported SIP phone directory numbers
(extensions) by a Cisco router in the voice register global
Example:
configuration mode.
Router(config-register-global)# max-dn 40

Step 18 max-pool max-phones Sets maximum supported SIP phones by the Cisco Unified
SRST router.
Example:
Router(config-register-global)# max-pool 40 • Version- and platform-dependent; type? For range.

Step 19 exit Exits the voice register global configuration mode.

Example:
Router(config-register-global)# exit

Step 20 voice register dn dn-tag Enter the voice register directory number configuration
mode to define a directory number for a SIP phone.
Example:
Router(config)# voice register dn 17 Use the same directory number (DN) configured in Cisco
Unified Communications Manager to configure the voice
register directory number in Unified E-SRST.

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 directory number configuration

mode.
Example:
Router(config-register-dn)# exit

Step 23 voice register pool pool-tag Enters the voice register Pool configuration mode to set
phone-specific parameters for a SIP phone.
Example:
Router(config)# voice register pool 1

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Command or Action Purpose

Step 24 id [{network address mask mask | ip address mask Explicitly identifies a locally available individual or set of
mask | mac address}] [device-id-name devicename] SIP IP phones. The keywords and arguments are defined
as follows:
Example:
Router(config-register-pool)# id network 8.55.0.0 • network address mask mask: The network address
mask 255.255.0.0 mask mask keyword/argument combination is used
to accept SIP Register messages for the indicated
phone numbers from any IP phone within the
indicated IP subnet.
• ipaddress maskmask : The ip address mask mask
keyword/argument combination is used to identify
an individual phone.
• macaddress : MAC address of a particular
Cisco Unified IP Phone.
• device-id-namedevicename : Defines the device name
to be used to download the phone’s configuration file.

Step 25 dtmf-relay rtp-nte Forwards DTMF tones by using Real-Time Transport

Protocol (RTP) with the Named phone Event (NTE)
Example:
payload type and enables the DTMF relay using the RFC
Router(config-register-pool)# dtmf-relay rtp-nte 2833 standard method.

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
2. configure terminal
3. voice hunt-group hunt-tag {longest-idle | parallel | peer | sequential}
4. members logout
5. list number [, number...]
6. timeout seconds
7. statistics collect
8. exit

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
Example:
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 voice hunt-group hunt-tag {longest-idle | parallel | peer Enters voice hunt-group configuration mode to define a
| sequential} hunt group.
Example: • Hunt-tag—Unique sequence number for configuring
Router(config)# voice hunt-group 1 sequential the hunt group. Range is 1–100.
• 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.

Step 4 members logout (optional) Configures a Cisco Unified SRST system for all
non-shared static members or agents in a voice hunt group
Example:
with the Hlogout initial state.
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 directs the
unanswered calls to the next number in a voice hunt-group
Example:
list.
Router(config-voice-hunt-group)# timeout 30

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:

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Command or Action Purpose

Router(config-voice-hunt-group)# exit

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
voice register global
mode esrst
max-dn 40
max-pool 40
Router#
voice register pool 1
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
call-park system application
transfer-system full-consult
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.
voice register dn 2
number 4000
!
voice register dn 3
number 4002
!
voice register pool 2
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
!

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voice register pool 3

busy-trigger-per-button 2
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
!
voice register pool 2
busy-trigger-per-button 2
id device-id-name SEP00EBD5CD77ED
type 8811
number 1 dn 2
dtmf-relay rtp-nte
codec g711u;aw
voice register pool 3
busy-trigger-per-button 2
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 16param 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

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param number-of-hunt-grps 1
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 number-of-hunt-grps 4
param queue-manager-debugs 1
!
call-park system application

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
voice register global
mode esrst
no allow-hash-in-dn
max-dn 40
max-pool 40
Router# sh run | sec pool
max-pool 40
voice register pool 1
busy-trigger-per-button 2
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
voice register pool 2
busy-trigger-per-button 2
id device-id-name SEP00CCFC177A4E
type 8841
number 1 dn 2
dtmf-relay rtp-nte
username xxxx password uvwx
codec g711ulaw
no vad
voice register pool 3
busy-trigger-per-button 2
id device-id-name SEP0076861ADEF0
type 7841
number 1 dn 3
number 2 dn 22
dtmf-relay rtp-nte
username xxxx password uvwx
codec g711ulaw
no vad
voice register pool 4
busy-trigger-per-button 2
id device-id-name SEP00EBD5CD270C
type 8811
number 1 dn 4
number 2 dn 22

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dtmf-relay rtp-nte
username xxxx password uvwx
codec g711ulaw
no vad
voice register pool 5
busy-trigger-per-button 2
id device-id-name SEP94D4692A2553
type 8841
number 1 dn 5
dtmf-relay rtp-nte
username xxxx password uvwx
codec g711ulaw
no vad
voice register pool 6
busy-trigger-per-button 2
id device-id-name SEP00CAE540C4B5
type 8811
number 1 dn 6
number 2 dn 21
dtmf-relay rtp-nte
username xxxx password uvwx
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
voice-hunt-groups login
number 1812
voice register dn 3
voice-hunt-groups login
number 1813
voice register dn 4
voice-hunt-groups login
number 1814
voice register dn 5
voice-hunt-groups login
number 1815
voice register dn 6
voice-hunt-groups login
number 1816
voice register dn 21
voice-hunt-groups login
number 1821
shared-line
voice register dn 22
voice-hunt-groups login
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

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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
device-security-mode none
mac-address 1111.1111.1912
feature-button 1 HLog
type 7970
button 1:12 2:21
ephone 13
device-security-mode none
mac-address 1111.1111.1913
feature-button 1 HLog
type 7970
button 1:13 2:21
ephone 14
device-security-mode none
mac-address 1111.1111.1914
feature-button 1 HLog
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
ip source-address 8.39.23.24 port 2000
service phone sshAccess 0
service phone webAccess 0
max-conferences 8 gain -6
call-park system application
hunt-group logout HLog
transfer-system full-consult
fac standard

SCCP: Configure Unified E-SRST

You need to configure mode esrst under telephony-service to enable ESRST mode for SCCP Phones.

Before you begin

To enable the version negotiation feature in the Unified E-SRST mode, perform the following procedure.
• Cisco Unified Communications Manager Express 10.5 or later version
• Configure the telephony-services command.

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Note For SCCP phones, CME-as-SRST mode is provisioned using the SRST mode autoprovision command.
From 10.5 release onwards, deprecates this command. When you try to configure CME-as-SRST mode,
displays the following message: “Note: This configuration is being deprecated. Please configure "mode
esrst" to use the enhanced SRST mode.”

SUMMARY STEPS
1. enable
2. configure terminal
3. telephony-service
4. mode esrst
5. max-ephonesmax-phones
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. (Optional) namename
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...]
17. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
Example: • Enter your password if prompted.
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service

Step 4 mode esrst Enters telephony-service configuration mode.

Example:
Router(config-telephony)# mode esrst

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Command or Action Purpose

Step 5 max-ephonesmax-phones Enters telephony-service configuration mode.
Example:
Router(config-telephony)# max-ephones 24

Step 6 max-dn max-directory-numbers [preference Limits the number of directory numbers supported by this
preference-order] [no-reg primary | both] router.
Example: • Maximum number is the platform and
Router(config-telephony)# max-dn 24 no-reg primary version-specific. Type? For value.

Step 7 ip source-address ip-address [port port] [any-match | Identifies the IP address and port number that the Cisco
strict-match] Unified SRST router uses for IP phone registration.
Example: • port port—(Optional) TCP/IP port number to use for
Router(config-telephony)# ip source-address SCCP. Range is 2000–9999. Default is 2000.
192.168.11.1 port 2000
• Any-match—(Optional) Disables the strict IP address
checking for registration. It is the default setting.
• 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:
Router(config-telephony)# exit

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
Example:
message-waiting indicator (MWI).
Router(config)# ephone-dn 1
• Dn-tag—Identifies a particular directory number
during configuration tasks. Range is 1 to the
maximum number of directory numbers allowed on
the router platform. Type? To display 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
Example: extension or E.164 phone number.
Router(config-ephone-dn)# number 1001

Step 11 (Optional) namename Associates a name with this directory number.

Example: • Uses the Name for caller-ID displays and in the local
Router(config-ephone-dn)# name Smith, John directory listings.
• Follows the name order in the directory command.

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Command or Action Purpose

Step 12 exit Exits ephone-dn configuration mode.
Example:
Router(config-telephony)# end

Step 13 ephone phone-tag Enters ephone configuration mode to set ephone specific
parameters.
Example:
Router(config)# ephone 1 • Phone-tag—Unique sequence number that identifies
the 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:
Router(config-ephone)# mac-address 0022.555e.00f1 • Mac-address—Identifying MAC address of an IP
phone found on a sticker on the bottom of the phone.

Step 15 type phone-type [addon 1 module-type [2 module-type]] Specifies the type of phone.
Example:
Router(config-ephone)# type 7960

Step 16 button button-number{separator}dn-tag Associates a button number and line characteristics with
[,dn-tag...][button-number{x}overlay-button-number] an ephone-dn. Determines the maximum number of buttons
[button-number...] by phone type.
Example:
Router(config-ephone)# button 1:7

Step 17 end Returns to privileged EXEC mode.

Example:
Router(config-telephony)# end

Example
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.).

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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
2. configure terminal
3. ephone-dn dn-tag [dual-line]
4. number [secondary [number] [no-reg [both|primary]]
5. shared-line sip
6. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
Example: • Enter your password if prompted.
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

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
Example:
message-waiting indicator (MWI).
Router(config)# ephone-dn 1
• Dn-tag—Identifies a particular directory number
during configuration task. Range is 1 to the maximum
number of directory numbers allowed on the router
platform. Type? To display the range.

Step 4 number [secondary [number] [no-reg [both|primary]] Associates an extension number with this directory number.
Example: • number—String of up to 16 digits that represents an
Router(config-ephone-dn)# number 1001 extension or E.164 phone number.

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
Example:
Unified SCCP IP phones.
Router(config-ephone-dn)# shared-line sip

Step 6 end Returns to privileged EXEC mode.

Example:
Router(config-ephone-dn)# end

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Configure BLF for SCCP Phones

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

SUMMARY STEPS
1. enable
2. configure terminal
3. presence
4. max-subscriptionnumber
5. presence call-list
6. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
Example: • Enter your password if prompted.
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 presence
Step 4 max-subscriptionnumber
Step 5 presence call-list (To enable Presence feature for all the missed or received
or placed calls)

Step 6 end

Enable an SCCP Directory Number to Be Watched

To enable a directory number to be watched, perform the following procedure:

SUMMARY STEPS
1. ephone-dndn-tag
2. numbernumber
3. allow watch
4. end

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DETAILED STEPS

Command or Action Purpose

Step 1 ephone-dndn-tag
Step 2 numbernumber
Step 3 allow watch
Step 4 end

Enable BLF on an Ephone

To enable BLF on an ephone, perform the following steps:

SUMMARY STEPS
1. enable
2. configure terminal
3. ephoneephone-tag
4. buttonbutton-number{separator}dn-tag [,dn-tag...]
[button-number{x}overlay-button-number][button-number...]
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

DETAILED STEPS

Command or Action Purpose

Step 1 enable
Step 2 configure terminal
Step 3 ephoneephone-tag
Step 4 buttonbutton-number{separator}dn-tag [,dn-tag...]
[button-number{x}overlay-button-number][button-number...]
Step 5 blf-speed-dial tag number label string [device]
Step 6 presence call-list(To enable Presence feature for all the
missed/received/placed calls)
Step 7 end

Configure Digest Credentials on Cisco Unified Communications

Manager
To configure the username and password with Digest Authentication on Cisco Unified Communications
Manager, perform the following steps:

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SUMMARY STEPS
1. Log in to Cisco Unified Communications Manager.
2. Go to System>Security->Phone Security Profile.
3. Go to User Management > End User.
4. Go to the Phone Settings page and associate the user in the Digest User field.

DETAILED STEPS

Command or Action Purpose

Step 1 Log in to Cisco Unified Communications Manager.
Step 2 Go to System>Security->Phone Security Profile.
Step 3 Go to User Management > End User.
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:

Note Digest authentication does not work with 'id network' configuration in 'voice register pool'. It requires
'id device-id-name' or 'id Mac' configuration for individual pools. Also DN association on 'voice register
pool' is required.

SUMMARY STEPS
1. enable
2. configure terminal
3. voice register pool <pool-tag>
4. username <username> password <password>
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
Example: • Enter your password if prompted.
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 voice register pool <pool-tag>

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Example: Configuring Digest Credentials on ESRST

Command or Action Purpose

Step 4 username <username> password <password>
Step 5 end

Example: Configuring Digest Credentials on ESRST

The following example shows how to configure digest credentials on ESRST:
Router# conf terminal
Router(config)#voice register pool 10
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
2. configure terminal
3. ephone ephone tag
4. username <username> password <password>
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
Example: • Enter your password if prompted.
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 ephone ephone tag

Step 4 username <username> password <password>
Step 5 end

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Configure Digest Credentials on Unified E-SRST for SCCP

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 CHAPTER 5
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.
• Information About Setting Up the Network, on page 163
• How to Set Up the Network, on page 163

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 Communications Manager. 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 Communications Manager 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 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 now.

How to Set Up the Network

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 in the following section 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.

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Configuring Cisco Unified SRST on an MGCP Gateway Before Cisco IOS Release 12.3(14)T

Note The commands in the configuration section are ineffective unless both commands are configured. For
instance, your configuration will not work if you only configure the ccm-manager fallback-mgcp
command.

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 Before Cisco IOS Release 12.3(14)T
Perform this task to enable SRST on an MGCP Gateway if you are using software release before Cisco IOS
Release 12.3(14)T.

SUMMARY STEPS
1. enable
2. configure terminal
3. ccm-manager fallback-mgcp
4. call application alternate [ application-name] OR service [alternate |default ] service-name location
5. exit

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
Example: • Enter your password if prompted.
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

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

Step 4 call application alternate [ application-name] OR service The call application alternate command specifies that the
[alternate |default ] service-name location default voice application takes over if the MGCP application
is not available. The application-name argument is optional
Example:
and indicates the name of the specific voice application to
Router(config)# call application alternate use if the application in the dial peer fails. If a specific
OR application name is not entered, the gateway uses the default
application.
Router(config)# service default

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Command or Action Purpose

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 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.

Before you begin

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.

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

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
Example: • Enter your password if prompted.
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 ccm-manager fallback-mgcp Enables the gateway fallback feature and allows an MGCP
voice gateway to provide call processing services through
Example:
SRST or other configured applications when Cisco Unified
Router(config)# ccm-manager fallback-mgcp 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
Example:
application in the dial peer fails. If a specific application
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

Step 6 service[ alternate | default] service-name location Loads and configures a specific, standalone application on
a dial peer.
Example:
Router(config) service myapp • Alternate (Optional). Alternate service to use if the
https://myserver/myfile.vxml service 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 7 exit Exits global configuration mode and returns to privileged

EXEC mode.
Example:
Router(config)# exit

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Configuration Example of Enabling SRST on a MGCP Gateway using Cisco IOS Release 12.3(14)T

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 switch-type primary-net5
isdn incoming-voice voice
isdn bind-l3 ccm-manager
no cdp enable
!
!
!
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

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Configuring DHCP for Cisco Unified SRST Phones

port 1/0:15
!
!
call-manager-fallback
limit-dn 7960 2
ip source-address 10.48.80.9 port 2000
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

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, on page 169—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, on page 169—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, on page 170—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.

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Defining a Single DHCP IP Address Pool

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

DETAILED STEPS

Command or Action Purpose

Step 1 ip dhcp poolpool-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,
Example:
preceded by a forward slash.
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
Example:
to be the IP address of Cisco Unified CM.
Router(config-dhcp)# option 150 ip 10.0.22.1

Step 4 default-router ip-address Specifies the router to which the Cisco Unified IP phones
are connected directly.
Example:
Router(config-dhcp)# default-router 10.0.0.1 This router should be the Cisco Unified SRST 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.

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Defining the DHCP Relay Server

SUMMARY STEPS
1. ip dhcp poolpool-name
2. host ip-address subnet-mask
3. option 150 ip ip-address
4. default-router ip-address
5. exit

DETAILED STEPS

Command or Action Purpose

Step 1 ip dhcp poolpool-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
Example:
to be the IP address of Cisco Unified CM.
Router(config-dhcp)# option 150 ip 10.0.22.1

Step 4 default-router ip-address Specifies the router to which the Cisco Unified IP phones
are connected directly.
Example:
Router(config-dhcp)# default-router 10.0.0.1 This router should be the Cisco Unified SRST 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

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Specifying Keepalive Intervals

3. ip helper-address ip-address
4. exit

DETAILED STEPS

Command or Action Purpose

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
Example:
Command Reference, Release 12.3T for more information.
Router(config)# interface serial 0

Step 3 ip helper-address ip-address Specifies the helper address for any unrecognized broadcast
for TFTP server and Domain Name System (DNS) requests.
Example:
For each server, a separate ip helper-address command is
Router(config-if)# ip helper-address 10.0.22.1 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
1. call-manager-fallback
2. keepalive seconds
3. exit

DETAILED STEPS

Command or Action Purpose

Step 1 call-manager-fallback Enters call-manager-fallback configuration mode.
Example:

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Specifying Keepalive Intervals

Command or Action Purpose

Router(config)# call-manager-fallback

Step 2 keepalive seconds Sets the time interval, in seconds, between keepalive
messages that are sent to the router by Cisco Unified IP
Example:
Phones.
Router(config-cm-fallback)# keepalive 60
Seconds: Range is 10 to 65535. Default is 30.

Step 3 exit Exits call-manager-fallback configuration mode.

Example:
Router(config-cm-fallback)# exit

Example
The following example sets a keepalive interval of 45 seconds:
call-manager-fallback
keepalive 45

What to do next
The next step is setting up the phone and getting a dial tone. For instructions, see the Cisco Unified SIP SRST
section.

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 CHAPTER 6
Cisco Unified SIP SRST
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.

• Prerequisites for Cisco Unified SIP SRST 4.1, on page 173

• Restrictions for Cisco Unified SIP SRST 4.1, on page 174
• Information About Cisco Unified SIP SRST 4.1, on page 174
• How to Configure Cisco Unified SIP SRST 4.1 Features, on page 178

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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 Cisco Unified SIP SRST
Restrictions for Cisco Unified SIP SRST 4.1

Restrictions for Cisco Unified SIP SRST 4.1

• Cisco Unified SRST does not support line status speed-dial notification, Call Forward All synchronization,
dial plans, directory services, or Music On Hold (MOH).
• Before 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
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 phone number, off-hook
the desk phone, and dial the called number. The application initiates the call setup without the user having to
outdial 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.

Digit Collection on SIP Phones

When you dial a phone, the digit strings must be collected and matched against predefined patterns to place
calls to the destination corresponding to your input. Previously, SIP phones in a Cisco Unified SRST system
required you to press the DIAL softkey or # key, or wait for the interdigit-timeout to trigger the 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
• SIP Dial Plans

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KPML Digit Collection

KPML Digit Collection

The Key Press Markup Language (KPML) uses SIP SUBSCRIBE and NOTIFY methods to report a user
input digit by digit. Each digit you dial generates its own signaling message to Cisco Unified SRST. Cisco
Unified SRST performs a pattern recognition by matching the destination pattern to the 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 to press the dial softkey 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 Enabling KPML for SIP Phones section.

SIP Dial Plans

A dial plan is a set of dial patterns that SIP phones use to determine when a digit collection is complete after
you go off-hook and dial a destination number. Dial plans enable SIP phones to perform local digit collection
and recognize dial patterns that you have keyed. After a pattern is recognized, the SIP phone sends an INVITE
message to Cisco Unified SRST to initiate the call to the number matching your input. All 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 softkey 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 softkey 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 softkey or wait for the interdigit timeout before digits are sent to
Cisco Unified SRST for processing.

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 Caller ID display includes the name and number of the caller 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

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 Cisco Unified SIP SRST
Disabling SIP Supplementary Services for Call Forward and Call Transfer

ID. The change in the caller ID is of the originating party such as with the call forwarding or Call Transfer.
No new configuration is required to support these enhancements.

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 Cisco Unified SRST. You can disable these supplementary
features if the destination gateway does not support them.

SUMMARY STEPS
1. enable
2. configure terminal
3. voice service voip OR dial-peer voice tag voip
4. no supplementary-service sip {moved-temporarily | refer}
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
Example: • Enter your password if prompted.
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 voice service voip OR dial-peer voice tag voip Enters voice-service configuration mode to set global
parameters for VoIP features.
Example:
Router(config)# voice service voip OR
OR Enters dial peer configuration mode to set parameters for
a specific dial peer.
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.
Example: • Moved-temporarily: SIP redirect response for call
Router(conf-voi-serv)# no supplementary-service forwarding.
sip refer
• Refer: SIP REFER message for Call Transfers.
OR
• Sending REFER and redirect messages to the
Router(config-dial-peer)# no
supplementary-service sip refer
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.

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Idle Prompt Status

Command or Action Purpose

Step 5 end Exits to privileged EXEC mode.
Example:
Router(config-voi-serv)# end

OR
Router(config-dial-peer)# end

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 you register a SIP phone 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 supports the Cisco Unified IP Phone 7911G, 7941G, 7941GE, 7961G, 7961GE,
7970G, and 7971GE with Cisco Unified SRST 4.1 onwards. 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 enable 911 operators to:
• Immediately pinpoint the location of the 911 caller based on the calling number.
• Call back 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 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 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, emergency 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 to contact the 911 caller.
See Configuring Enhanced 911 Services from Cisco Unified Communications Manager Express System
Administrator Guide for more information.

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How to Configure Cisco Unified SIP SRST 4.1 Features

How to Configure Cisco Unified SIP SRST 4.1 Features

Enabling KPML for SIP Phones
Perform the following steps to enable KPML digit collection on a SIP phone.

Before you begin

• 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
2. configure terminal
3. voice register pool pool-tag
4. digit collect kpml
5. end
6. show voice register dial-peers

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
Example: • Enter your password if prompted.
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 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 4 pool-tag: Unique sequence number of the SIP phone to 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.
Example: Note This command is enabled by default for
Router(config-register-pool)# digit collect supported phones in Cisco Unified
kpml Communications Manager Express and Cisco
Unified SRST.

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Disabling SIP Supplementary Services for Call Forward and Call Transfer

Command or Action Purpose

Step 5 end Exits to privileged EXEC mode.
Example:
Router(config-register-pool)# end

Step 6 show voice register dial-peers Displays details of all dynamically created VoIP dial peers
associated with the Cisco Unified Communications Manager
Example:
Express SIP register including the defined digit collection
Router# show voice register dial-peers method.

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.

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 Cisco Unified SRST. You can disable these supplementary
features if the destination gateway does not support them.

SUMMARY STEPS
1. enable
2. configure terminal
3. voice service voip OR dial-peer voice tag voip
4. no supplementary-service sip {moved-temporarily | refer}
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
Example: • Enter your password if prompted.
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 voice service voip OR dial-peer voice tag voip Enters voice-service configuration mode to set global
parameters for VoIP features.
Example:
Router(config)# voice service voip OR
OR Enters dial peer configuration mode to set parameters for
a specific dial peer.
Router(config)# dial-peer voice 99 voip

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Configuring Idle Prompt Status for SIP Phones

Command or Action Purpose

Step 4 no supplementary-service sip {moved-temporarily | Disables SIP call forwarding or Call Transfer supplementary
refer} services globally or for a dial peer.
Example: • Moved-temporarily: SIP redirect response for call
Router(conf-voi-serv)# no supplementary-service forwarding.
sip refer
• Refer: SIP REFER message for Call Transfers.
OR
• Sending REFER and redirect messages to the
Router(config-dial-peer)# no
supplementary-service sip refer
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.

Step 5 end Exits to privileged EXEC mode.

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.

Before you begin

Cisco Unified SRST 4.1 or a later version.

SUMMARY STEPS
1. enable
2. configure terminal
3. voice register global
4. system message string
5. end
6. show voice register global

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
Example: • Enter your password if prompted.
Router> enable

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Command or Action Purpose

Step 2 configure terminal Enters global configuration mode.
Example:
Router# configure terminal

Step 3 voice register global Enters voice register global configuration mode to set global
parameters for all supported SIP phones in a Cisco Unified
Example:
Communications Manager Express environment.
Router(config)# voice register global

Step 4 system message string Defines a status message that displays on SIP phones
registered to Cisco Unified SRST.
Example:
Router(config-register-global)# system message • String: Up to 32 alphanumeric characters. Default
fallback active value is CM Fallback Service Operating.

Step 5 end Exits to privileged EXEC mode.

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

What to do next
The next step is configuring Cisco Unified IP phones using SCCP. For instructions, see Setting Up Cisco
Unified IP Phones using SCCP section.

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 CHAPTER 7
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.

• Information About Setting Up Cisco Unified IP Phones, on page 183

• How to Set Up Cisco Unified IP Phones, on page 183
• How to Set Up Cisco IP Communicator for Cisco Unified SRST, on page 198
• Where to Go Next, on page 199

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.

How to Set Up Cisco Unified IP Phones

This section contains the following tasks:
• Configuring Cisco Unified SRST to Support Phone Functions (Required)
• Configuring Cisco Unified 8941 and 8945 SCCP IP Phones (Required)
• Verifying That Cisco Unified SRST Is Enabled(Optional)
• Configuring IP Phone Clock, Date, and Time Formats (Optional)
• Configuring IP Phone Language Display (Optional)

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Configuring Cisco Unified SRST to Support Phone Functions

• Configuring Customized System Messages for Cisco Unified IP Phones (Optional)

• Configuring a Secondary Dial Tone (Optional)
• Configuring Dual-Line Phones (Required Under Certain Conditions)
• Configuring Eight Calls per Button (Octo-Line)(Optional)
• Configuring the Maximum Number of Calls (Optional)
• Troubleshooting (Optional)

Configuring Cisco Unified SRST to Support Phone Functions

Note 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
1. call-manager-fallback
2. ip source-address ip-address [port port ] [ any-match | strict-match ]
3. max-dnmax-directory-numbers[dual-line][preferencepreference-order]
4. max-ephones max-ephones
5. limit-dn phone-type max-lines
6. exit

DETAILED STEPS

Command or Action Purpose

Step 1 call-manager-fallback Enters call-manager-fallback configuration mode.
Example:
Router(config)# call-manager-fallback

Step 2 ip source-address ip-address [port port ] [ any-match | Enables the router to receive messages from the Cisco IP
strict-match ] phones through the specified IP addresses and provides for
strict IP address verification. The default port number is
Example:
2000.
Router(config-cm-fallback)# ip source-address
10.6.21.4 port 2002 strict-match

Step 3 max-dnmax-directory-numbers[dual-line][preferencepreference-order] Sets the maximum number of directory numbers (DNs) or

virtual voice ports that can be supported by the router and
Example:
activates dual-line mode.
Router(config-cm-fallback)# max-dn 15 dual-line
preference 1

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Command or Action Purpose

• max-directory-numbers:Maximum number of
directory numbers (dns) or virtual voice ports
supported by the router. The maximum number is
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 commandpreference keyword allows the default
preference set with the max-dn command to be
overridden. See the Configuring Call Rerouting section
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-ephones Configures the maximum number of Cisco IP phones that
can be supported by the router. The default is 0. The
Example:
maximum number is platform dependent. See Compatibility
Router(config-cm-fallback)# max-ephones 24 Information for further details.
Note You must reboot the router to reduce the limit
of Cisco IP phones after the maximum allowable
number is configured.

Step 5 limit-dn phone-type max-lines Optional) Limits the directory number lines on Cisco IP
phones during Cisco Unified CM fallback.
Example:
Router(config-cm-fallback)# limit-dn 7945 2 Note You must configure this command during initial
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).

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Configuring Cisco Unified 8941 and 8945 SCCP IP Phones

Command or Action Purpose

Step 6 exit Exits call-manager-fallback configuration mode.
Example:
Router(config-cm-fallback)# exit

Configuring Cisco Unified 8941 and 8945 SCCP IP Phones

To configure Cisco Unified 8941 and 8945 SCCP IP Phones in Unified SRST mode, perform the following
commands:

Note This section is required only in SRST version 8.6 and is not required for version 8.8 and higher.

SUMMARY STEPS
1. enable
2. configure terminal
3. ephone-type phone-type
4. device-idnumber
5. device-type phone-type
6. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
Example: • Enter your password if prompted.
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 ephone-type phone-type Enters phone type to configure.

Example: • 8941
phone-type
• 8945

Step 4 device-idnumber Specifies the device ID for the phone type.

Example: • 8941—586
Router(config-ephone-type)# device-id 586
• 8945—585

Step 5 device-type phone-type Specifies the device type for the phone.

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Verifying That Cisco Unified SRST Is Enabled

Command or Action Purpose

Example: • 8941
Router(config-ephone-type)# device-type 8941
• 8945

Step 6 end Returns to privileged EXEC mode.

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:
1. Enter the show running-config command to verify the configuration.
2. Enter the show call-manager-fallback all command to verify that the Cisco Unified SRST feature is
enabled.
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.
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

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Configuring IP Phone Clock, Date, and Time Formats

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 timezonezone hours-offset[minutes-offset]
2. call-manager-fallback
3. date-format {mm-dd-yy|dd-mm-yy|yy-dd-mm|yy-mm-dd}
4. time-format [12 | 24 ]
5. exit

DETAILED STEPS

Command or Action Purpose

Step 1 clock timezonezone hours-offset[minutes-offset] Sets the time zone for display purposes.
Example: • zone: Name of the time zone to be displayed when
Router(config)# clock timezone PST -8 standard time is in effect. The length of the zone
argument is limited to 7 characters.
• hours-offset: The number of hour difference from
Coordinated Universal Time (UTC).
• minutes-offset (Optional). Minutes difference from
UTC.

Step 2 call-manager-fallback Enters call-manager-fallback configuration mode.

Example:
Router(config)# call-manager-fallback

Step 3 date-format Sets the date format for IP phone display. The choices are
{mm-dd-yy|dd-mm-yy|yy-dd-mm|yy-mm-dd} mm-dd-yy, dd-mm-yy, yy-dd-mm, and yy-mm-dd, where
Example: • dd: day
Router(config-cm-fallback)# date-format
yy-dd-mm
• mm: month
• 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
Example:
clock.
Router(config-cm-fallback)# time-format 24

Step 5 exit Exits call-manager-fallback configuration mode.

Example:

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Command or Action Purpose

Router(config-cm-fallback)# exit

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

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
1. call-manager-fallback
2. configure terminal
3. user-locale country-code
4. exit

DETAILED STEPS

Command or Action Purpose

Step 1 call-manager-fallback Enters call-manager-fallback configuration mode.
Example:
Router(config)# call-manager-fallback

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 user-locale country-code Selects a language by country for displays on the Cisco IP
Phone 7940 and Cisco IP Phone 7960.
Example:

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Command or Action Purpose

Router(config-cm-fallback)# user-locale ES The following ISO-3166 codes are available to Cisco SRST
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).

Step 4 exit Exits call-manager-fallback configuration mode.

Example:
Router(config-cm-fallback)# exit

Example
The following example offers a configuration for the Portugal user locale:
call-manager-fallback
user-locale PT

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.”

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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
1. call-manager-fallback
2. system message {primaryprimary-string|secondarysecondary-string}
3. exit

DETAILED STEPS

Command or Action Purpose

Step 1 call-manager-fallback Enters call-manager-fallback configuration mode.
Example:
Router(config)# call-manager-fallback

Step 2 system message Declares the text for the system display message on IP
{primaryprimary-string|secondarysecondary-string} phones in fallback mode.
Example: • primary primary-string: For Cisco Unified IP Phones
Router(config-cm-fallback)# system message primary that can support static text messages during fallback,
Custom Message such as the Cisco Unified IP Phone 7940 and Cisco
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.

Step 3 exit Exits call-manager-fallback configuration mode.

Example:
Router(config-cm-fallback)# exit

Example
The following example sets “SRST V3.0” as the system display message for all Cisco Unified IP
Phones on a router:

call-manager-fallback
system message primary SRST V3.0
system message secondary SRST V3.0
exit

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Configuring a Secondary Dial Tone

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
1. call-manager-fallback
2. secondary-dialtonedigit-string
3. exit

DETAILED STEPS

Command or Action Purpose

Step 1 call-manager-fallback Enters call-manager-fallback configuration mode.
Example:
Router(config)# call-manager-fallback

Step 2 secondary-dialtonedigit-string Activates a secondary dial tone when a digit string is dialed.
Example:
Router(config-cm-fallback)# secondary-dialtone 9

Step 3 exit Exits call-manager-fallback configuration mode.

Example:
Router(config-cm-fallback)# exit

Example
The following example sets the number 8 to trigger a secondary dial tone:
call-manager-fallback
secondary-dialtone 8

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 Unified
SRST 3.0 section.
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.

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It is recommended that hunting be disabled to the second channel. For more information, see the Configuring
Dial-Peer and Channel Hunting section.

SUMMARY STEPS
1. call-manager-fallback
2. max-dnmax-directory-numbers[dual-line][preferencepreference-order]
3. exit

DETAILED STEPS

Command or Action Purpose

Step 1 call-manager-fallback Enters call-manager-fallback configuration mode.
Example:
Router(config)# call-manager-fallback

Step 2 max-dnmax-directory-numbers[dual-line][preferencepreference-order] Sets the maximum number of directory numbers (DNs) or

virtual voice ports that can be supported by the router and
Example:
activates dual-line mode.
Router(config-cm-fallback)# max-dn 15 dual-line
preference 1 • max-directory-numbers:Maximum number of
directory numbers (dns) or virtual voice ports
supported by the router. The maximum number is
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 commandpreference keyword allows the default
preference set with the max-dn command to be
overridden. See the Configuring Call Rerouting section
for more information on using the max-dn command
with the alias command.

Step 3 exit Exits call-manager-fallback configuration mode.

Example:
Router(config-cm-fallback)# exit

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Example
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:

call-manager-fallback
max-dn 10 dual-line
exit

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 tagcommand, 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

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.

Before you begin

• Cisco Unified SRST 7.0/4.3
• Cisco Unified CM 6.0

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• Cisco IOS Release 12.4(15)XZ

SUMMARY STEPS
1. enable
2. configure terminal
3. call-manager-fallback
4. max-dn max-no-of-directories[dual-line |octo-line] [number octo-line]
5. huntstop channel1-8
6. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
Example: • Enter your password if prompted.
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 call-manager-fallback Enters call-manager-fallback configuration mode.

Example:
Router(config)# call-manager-fallback

Step 4 max-dn max-no-of-directories[dual-line |octo-line] Sets the maximum number of DNs or virtual voice ports
[number octo-line] that can be supported by the router and activates dual-line
mode, octo-line mode, or both modes.
Example:
Router(config-cm-fallback)# max-dn 15 dual-line • max-no-of-directories: Maximum number of directory
6 octo-line numbers (dns) or virtual voice ports supported by the
router. The maximum number is platform-dependent.
The default is 0.
• dual-line: (Optional) Allows IP phones in Cisco
Unified Communications Manager fallback mode to
have a virtual voice port with two channels.
• octo-line: (Optional) Allows IP phones in Cisco
Unified Communications Manager fallback mode to
have a virtual voice port with eight channels.
• number (Optional): Sets the number of directory
numbers for octo-mode.

Step 5 huntstop channel1-8 Enables channel huntstop on an octo-line, which keeps a

call from hunting to the next channel of a directory number
Example:
if the last allowed channel is busy or does not answer.
Router(config-cm-fallback)# huntstop channel 4

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Command or Action Purpose

• 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.

Step 6 end Returns to privileged EXEC mode.

Example:
Router(config)# end

Example
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:

!
call-manager-fallback
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.

Before you begin

• Cisco Unified SRST 9.0 and later versions.
• Correct firmware, 9.2(1) or a later version, is installed.

SUMMARY STEPS
1. enable
2. configure terminal
3. call-manager-fallback
4. max-dn max-no-of-directories [dual-line | octo-line ]
5. timeouts busy seconds
6. end

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
Example: • Enter your password if prompted.
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 call-manager-fallback Enables Cisco Unified SRST support and enters

call-manager-fallback configuration mode.
Example:
Router(config)# call-manager-fallback

Step 4 max-dn max-no-of-directories [dual-line | octo-line ] Sets the maximum possible number of directory numbers
or virtual voice ports that can be supported by a router and
Example:
enables dual-line mode, octo-line mode, or both modes.
Router(config-cm-fallback)# max-dn 10 octo-line
• max-no-of-directories—Maximum number of
directory 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.
Example: • seconds—Number of seconds after connection to a
Router(config-cm-fallback)# timeouts busy 10 busy destination before a transferred call is
disconnected. Range is 0 to 30. Default: 10.

Step 6 end Exits configuration mode and enters privileged EXEC mode.
Example:
Router(config-cm-fallback)# end

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.

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• 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)

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.

2. (Optional) Attach a headset to your PC.

3. Start the Cisco IP Communicator software application.
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.

5. Define the IP address of the Cisco Unified SRST as secondary TFTP server

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a. Open the Network > User Preferences window.

b. Enter the IP address of the Cisco Unified SRST TFTP server.

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.
7. Wait for the Cisco IP Communicator to connect to the Cisco Unified SRST system (upon Cisco Unified
CM Failure) and register itself.
8. Cisco IP Communicator should have retained the original buttons and numbers for Cisco IP Communicator.

Verifying Cisco IP Communicator

SUMMARY STEPS
1. Use the show running-config command to display ephone-dn and ephone information associated with
this phone.
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.
3. Make a local call from the phone and ask someone to call you. Verify that you have a two-way voice path.

DETAILED STEPS

Command or Action Purpose

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 Setting Up
Cisco Unified IP Phones using SIP section.
For additional information, see the Related Documents section.

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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.
• Prerequisites for Configuring the SIP Registrar, on page 201
• Restrictions for Configuring the SIP Registrar, on page 201
• Information About Configuring the SIP Registrar, on page 201
• How to Configure the SIP Registrar, on page 202

Prerequisites for Configuring the SIP Registrar

Complete the prerequisites documented in the Prerequisites for Configuring Cisco Unified SIP SRST section
in Cisco Unified SCCP and SIP SRST Feature Overview chapter.

Restrictions for Configuring the SIP Registrar

See the restrictions documented in the Restrictions for Configuring Cisco Unified SIP SRST section in the
Cisco Unified SCCP and SIP SRST Feature Overview chapter.

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:
• 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.

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How to Configure the SIP Registrar

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
2. configure terminal
3. voice service voip
4. allow-connections sip to sip
5. sip
6. registrar server [ expires [ maxsec] [minsec] ]
7. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
Example: • Enter your password if prompted.
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

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

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Command or Action Purpose

Step 5 sip Enters SIP configuration mode.
Example:
Router(config-voi-srv)# sip

Step 6 registrar server [ expires [ maxsec] [minsec] ] Enables SIP registrar functionality. The keywords and
arguments are defined as follows:
Example:
Router(conf-serv-sip)# registrar server expires • expires: (Optional) Sets the active time for an incoming
max 600 min 60 registration.
• 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

What to do next
For incoming SIP Register messages to be successfully accepted, users must also set up a voice register pool.
See the section Configuring Backup Registrar Service to SIP Phones.

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

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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).

Before you begin

The SIP registrar must be configured before a voice register pool is set up. See the section Configuring the
SIP Registrar.
Restrictions
• The id command identifies the individual SIP IP phone or sets of SIP IP phones that are to be configured.
Thus, theidcommand configured in Step 5 is required and must be configured before any other voice
register pool commands. When themacaddress 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.
• If Jabber for desktop clients must register with Unified SRST, ensure thatvoice register pools are
configured for all desktop computer networks.

Note To monitor SIP proxies, the call fallback active command must be configured, as described in Step 3

SUMMARY STEPS
1. enable
2. configure terminal
3. call fallback active
4. voice register pool tag
5. id { network address mask mask | ip address mask mask | mac address }
6. preference preference-order
7. proxy ip-address [preference value [ monitor probe {icmp-ping | rtr } alternate-ip-address ]]
8. voice-class codec tag
9. (Optional) application application-name
10. end

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
Example: • Enter your password if prompted.
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 call fallback active Enables a call request to fall back to alternate dial peers
in case of network congestion.
Example:
Router(config)# call fallback active This command is used if you want to monitor the proxy
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:
Router(config)# voice register pool 12 Use this command to control which registrations are
accepted or rejected by a Cisco Unified SIP SRST device.

Step 5 id { network address mask mask | ip address mask mask Explicitly identifies a locally available individual or set of
| mac address } SIP IP phones. The keywords and arguments are defined
as follows:
Example:
Router(config-register-pool)# id network • network address mask mask : The network address
172.16.0.0 mask 255.255.0.0 mask mask keyword/argument combination is used
to accept SIP Register messages for the indicated
phone 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.

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
Example:
highest preference.
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.

Step 7 proxy ip-address [preference value [ monitor probe Autogenerates additional VoIP dial peers to reach the main
{icmp-ping | rtr } alternate-ip-address ]] SIP proxy whenever a Cisco Unified SIP IP Phone registers

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Command or Action Purpose

Example: with a Cisco Unified SIP SRST gateway. The keywords
Router(config-register-pool)# proxy and arguments are defined as follows:
10.2.161.187 preference 1
• ip-address : The ip-address of the SIP Proxy.
• preference value : Defines the preference of the
proxy dial peers that are created. The preference must
be less (higher priority) than the preference configured
with the reference command.
Range is from 0 to 10. The highest preference is 0.
There is no default.
• monitor probe : 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 : 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 (Optional) application application-name Selects the session-level application on the VoIP dial peer.
Use the application-name argument to define a specific
Example:
interactive voice response (IVR) application.
Router(config-register-pool)# application
SIP.App

Step 10 end Returns to privileged EXEC mode.

Example:
Router(config-register-pool)# end

What to do next
There are several more voice register pool commands that add functionality, but that are not required. See the
section Configuring Backup Registrar Service to SIP Phone (Using Optional Commands) for these commands.

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ConfiguringBackupRegistrarServicetoSIPPhone(UsingOptionalCommands)
The prior configurations set up a basic voice register pool. The configuration in this procedure adds optional
attributes to increase functionality.

Before you begin

• Prerequisites as described in the Configuring Backup Registrar Service to SIP Phones section.
• Configuration of the required commands as described in the Configuring Backup Registrar Service to
SIP Phones section .
• Before configuring the alias command, translation rules must be set using the translate-outgoing (voice
register pool) command.

SUMMARY STEPS
1. enable
2. configure terminal
3. voice register pool tag
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

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
Example: • Enter your password if prompted.
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 voice register pool tag Enters voice register pool configuration mode.
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 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
Example:
a SIP phone.
Router(config-register-pool)#
voice translation-rule 1

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Command or Action Purpose

rule 1 /1000/ /1006/ Profile-tag: Translation profile name to handle translation
!
to outgoing calls.
!
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
Example:
the main proxy is not available. The keywords and
Router(config-register-pool)# alias 1 94... to arguments 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
telephone number and may include wildcards.
• to: Connects the tag number pattern to the alternate
number.
• target : The target number; an alternate telephone
number to route incoming calls to match the number
pattern.
• preferencevalue : 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.

Step 6 cor {incoming | outgoing} cor-list-name {cor-list-number Configures a class of restriction (COR) on the VoIP dial
starting-number [- ending-number] | default } peers associated with directory numbers. COR specifies
which incoming dial peers can use which outgoing dial
Example:
peers to make a call. Each dial peer can be provisioned
Router(config-register-pool)# cor incoming with an incoming and outgoing COR list. The keywords
call91 1 91011
and arguments are defined as follows:
• 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.
• Indicator that a full range is configured.

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Command or Action Purpose

• ending-number : 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
Example:
a sequence of digits that represent a phone number prefix.
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
[huntstop ] permits to handle the Register message from the Cisco
Unified SIP IP Phone. The keywords and arguments are
Example:
defined as follows:
Router(config-register-pool)# number 1 50..
preference 2 • tag : Number from 1 to 10 and the distinguishing
factor 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 : Defines the number list preference
order.
• huntstop : Stops hunting if the dial peer is busy.

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

Step 10 end Returns to privileged EXEC mode.

Example:
Router(config-register-pool)# end

Example
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

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172.16.0.0/16 network. Autogenerated dial peers for registrations that match pool 12 have attributes
configured in this pool.
.
.
.
voice register pool 12
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

DETAILED STEPS

Command or Action Purpose

Step 1 debug voice register errors Use this command to debug errors that happen during
registration.
Example:
Router# debug voice register errors If there are no voice register pools configured for a
*Apr 22 11:52:54.523 PDT: VOICE_REG_POOL: Contact particular registration request, the message Contact
doesn't match any pools doesn’t match any pools is displayed.
*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.
*Apr 22 11:52:54.559 PDT: VOICE_REG_POOL: Register
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
Example:
includes matching of pools, registration creation, and
Router# debug voice register events automatic creation of dial peers. For more details and error
Apr 22 10:50:21.731 PDT: VOICE_REG_POOL: Contact
matches pool 1 conditions, you can use the debug voice register errors
Apr 22 10:50:21.731 PDT: VOICE_REG_POOL: key(91011) command.
contact(192.168.0.2) add to contact table

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Command or Action Purpose

Apr 22 10:50:21.731 PDT: VOICE_REG_POOL: key(91011) The phone number 91011 registered successfully, and type
exists in contact table
1 is reported, which means there is a pre-existing VoIP dial
Apr 22 10:50:21.731 PDT: VOICE_REG_POOL:
contact(192.168.0.2) exists in contact table, ref peer.
updated
Apr 22 10:50:21.731 PDT: VOICE_REG_POOL: Created
dial-peer entry of type 1
Apr 22 10:50:21.731 PDT: VOICE_REG_POOL:
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

Verifying Proxy Dial-Peer Configuration

To use the icmp-ping keyword with the proxy command to assist in troubleshooting proxy dial peers, perform
the following steps.

SUMMARY STEPS
1. configure terminal
2. voice register pool
3. proxy ip-address[preferencevalue] [monitor probe {icmp-ping|rtr}[alternate-ip-address]]
4. end
5. show voice register dial-peers
6. show dial-peer voice

DETAILED STEPS

Command or Action Purpose

Step 1 configure terminal Use this command to enter global configuration mode.
Example:
Router# configure terminal

Step 2 voice register pool Use this command to enter voice register pool configuration
mode.
Example:
Router(config)# voice register pool 1

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Command or Action Purpose

Step 3 proxy ip-address[preferencevalue] [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

Step 4 end Returns to privileged EXEC mode.

Example:
Router(config-register-pool)# end

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
dial-peer voice 40036 voip
preference 1
destination-pattern 91011
session target ipv4:10.2.161.187
session protocol sipv2
voice-class codec 1
monitor probe icmp-ping 10.2.161.187

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 Note Also highlighted is the output of the cor and
VoiceOverIpPeer40036 incoming called-number commands.
peer type = voice, information type = voice,
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 = `unknown'
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

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Command or Action Purpose

incoming COR list:maximum capability
! Default output for cor command
outgoing COR list:minimum requirement
! Default output for cor command
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 = `ipv4:10.2.161.187',
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,
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

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Command or Action Purpose

Calls = 0
Accepted Calls = 0, Refused Calls = 0,
Last Disconnect Cause is "",
Last Disconnect Text is "",
Last Setup Time = 0.

What to do next
The next step is configuring incoming and outgoing calls for Cisco Unified SRST. For more information, see
the Configuring Call Handling section.

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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.

• Prerequisites for Configuring SIP SRST Features Using Back-to-Back User Agent Mode, on page 215
• Restrictions for Configuring SIP SRST Features Using Back-to-Back User Agent Mode, on page 216
• Information About Configuring SCCP SRST Call Handling, on page 216
• Information About Configuring SIP SRST Features Using Back-to-Back User Agent Mode, on page 217
• How to Configure Cisco Unified SCCP SRST, on page 220
• Configuring Outgoing Calls, on page 235
• How to Configure Cisco Unified SIP SRST, on page 250
• How to Configure Optional Features, on page 260
• Configuration Examples for Call Handling, on page 262
• Where to Go Next, on page 263

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 in the Cisco Unified SCCP and SIP SRST Feature Overview, on page 41.
• 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 Prerequisites for Configuring the
SIP Registrar section.

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Restrictions for Configuring SIP SRST Features Using Back-to-Back User Agent Mode

Restrictions for Configuring SIP SRST Features Using

Back-to-Back User Agent Mode
• See the restrictions documented in the Restrictions for Configuring Cisco Unified SIP SRST section in
the Cisco Unified SRST Feature Overview.

Information About Configuring SCCP SRST Call Handling

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

H.323 VoIP Call Preservation Enhancements for WAN Link Failures

H.323 VoIP call preservation enhancements for WAN link failures sustain 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 Chapter “Configuring H.323 Gateways” in Cisco IOS H.323 Configuration
Guide, Release 12.4T.

Note H.323 is deprecated from IOS XE 17.6.1.

Toll Fraud Prevention

When a Cisco router platform is installed with a voice-capable Cisco IOS Software, appropriate features must
be enabled on the platform to prevent potential toll fraud exploitation by unauthorized users. Deploy these
features on all Cisco router Cisco Unified Communications applications that process voice calls, such as Cisco
Unified Communications Manager Express (Cisco Unified Communications Manager Express), 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.

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

Information About Configuring SIP SRST Features Using

Back-to-Back User Agent Mode
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 give 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 SIP phone
• SIP phone to PSTN / router voice port
• SIP phone to SCCP phone

Cisco Unified SIP SRST and Cisco SIP Cisco Unified Communications Manager
Express Feature Crossover
The voice register directory number, voice register global, and voice register Pool configuration mode
commands are accessible in both Cisco Unified SIP Cisco Unified Communications Manager Express and
Cisco Unified SIP SRST modes of operation. However, not all the commands within these modes are intended
for use in SIP SRST mode. The following table provides a summary guide to which commands are relevant
to the Cisco Unified Communications Manager Express or SRST modes of operation.
For more detailed information, refer to the command reference pages for each of the individual commands.

Note The following table is not all-inclusive; more commands may exist.

Command Dial Voice Register Configurable for Cisco Applicable to Cisco

Peer Mode Unified (SIP) Cisco Unified Unified (SIP) Cisco Unified
Communications Manager Communications Manager
Express and Cisco Unified Express Only
SIP SRST

After-hour X DN X —
exempt

Auto-answer — DN — X

Call forward X DN X —

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Command Dial Voice Register Configurable for Cisco Applicable to Cisco

Peer Mode Unified (SIP) Cisco Unified Unified (SIP) Cisco Unified
Communications Manager Communications Manager
Express and Cisco Unified Express Only
SIP SRST

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

Time zone — Global — X

Upgrade — Global — X

Url — Global — X

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Command Dial Voice Register Configurable for Cisco Applicable to Cisco

Peer Mode Unified (SIP) Cisco Unified Unified (SIP) Cisco Unified
Communications Manager Communications Manager
Express and Cisco Unified Express Only
SIP SRST

Voicemail — Global — X

After-hour X Pool X —
exempt

Application X Pool X —

Call-forward — Pool X —

Call-waiting — Pool — X

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

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Command Dial Voice Register Configurable for Cisco Applicable to Cisco

Peer Mode Unified (SIP) Cisco Unified Unified (SIP) Cisco Unified
Communications Manager Communications Manager
Express and Cisco Unified Express Only
SIP SRST

Dnd-control — Template — X

Forward — Template — X

Transfer — Template — X

How to Configure Cisco Unified SCCP SRST

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 (Optional)
• Configuring Call Rerouting (Optional)
• Configuring Call Pickup (Optional)
• Configuring Transfer Digit Collection Method (Optional)

• Phone Number Conversion and Translation

• Configuring Global Prefixes(Optional)
• Enabling Digit Translation Rules (Optional)
• Enabling Translation Profiles (Optional)
• Verifying Translation Profiles (Optional)

• Hunting and Ringing Timeout Behavior

• Configuring Dial-Peer and Channel Hunting (Optional)
• Configuring Busy Timeout (Optional)
• Configuring the Ringing Timeout Default (Optional)

Configuring Call Forwarding During a Busy Signal or No Answer

Configure the incoming calls that reach busy signal or go unanswered during the Cisco Unified Communications
Manager fallback to call forwarding to one or more E.164 numbers.

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SUMMARY STEPS
1. call-manager-fallback
2. call-forward busy directory-number
3. call-forward noan directory-number timeout seconds
4. exit

DETAILED STEPS

Command or Action Purpose

Step 1 call-manager-fallback Enters call-manager-fallback configuration mode.
Example:
Router(config)# call-manager-fallback

Step 2 call-forward busy directory-number Configures call forwarding to another number when the
Cisco IP phone is busy.
Example:
Router(config-cm-fallback)# call-forward busy directory-number : Selected directory number representing
50.. a fully qualified E.164 number. This 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 seconds Configures call forwarding to another number when
receiving no answer from the Cisco IP phone.
Example:
Router(config-cm-fallback)# call-forward noan • directory-number : Selected directory number
5005 timeout 10 representing a fully qualified E.164 number or a local
extension number. This number can contain “.”
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 3–60000.

Step 4 exit Exits call-manager-fallback configuration mode.

Example:
Router(config-cm-fallback)# exit

Example
The following example forwards calls to extension number 5005 when an incoming call reaches a
busy or unattended IP phone extension number. Incoming calls ring for 15 seconds before forwarding
to extension 5005.
call-manager-fallback
call-forward busy 5005
call-forward noan 5005 timeout seconds 15

The following example transforms an extension number for a call forwarding when the extension
number is busy or unattended. The call-forward busy command has an argument of 50.., which

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prepends the digits 50 to the last two digits of the called extension. The resulting extension is the
call forwarding number when the original extension number is busy or unattended. For instance,
forwards an incoming call to busy extension 6002 to extension 5002, and forwards an incoming call
to busy extension 3442 to extension 5042. Incoming calls ring for 15 seconds before being forwarded.
call-manager-fallback
call-forward busy 50..
call-forward noan 50.. timeout seconds 15

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 phone numbers that are unavailable during
fallback. Up to 50 sets of rerouting alias rules can be created for calls to phone numbers that are unavailable
during a Cisco Unified Communications Manager fallback. Sets of alias rules are created using the alias
command. An alias is activated when a phone 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 multiplealias 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 extra 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).

Configuring Call Pickup

Configuring the pickup command enables the PickUp softkey 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) softkey.
When a user presses the PickUp softkey, SRST searches through all the SRST phones to find a incoming 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.

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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 softkey during fallback. To enable the PickUp
softkey on Cisco 7905 and Cisco 7912 IP phones, upgrade your default phone load to Cisco Unified
Communications Manager, 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
1. call-manager-fallback
2. no huntstop
3. alias tag number-pattern to alternate-number
4. pickup telephone number
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 call-manager-fallback Enters call-manager-fallback configuration mode.
Example:
Router(config)# call-manager-fallback

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 rule for rerouting calls to sets of phones that
are unavailable during Cisco Unified Communications
Example:
Manager fallback.
Router(config-cm-fallback)# alias 1 8005550100
to 5001 • tag : Identifier for the alias rule range. Range is from
1to 50.
• number-pattern : Pattern to match the incoming phone
number. This pattern may include wildcards.
• to : Connects the tag number pattern to the alternate
number.
• alternate-number : Alternate phone 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 phone
number can be used in multiple alias commands.

Step 4 pickup telephone number Enables the PickUp softkey on all Cisco Unified IP Phones,
allowing an external Direct Inward Dialing (DID) call
Example:
coming into one extension to be picked up from another

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Command or Action Purpose

Router(config-cm-fallback)# pickup 8005550100 extension during SRST. The telephone-number argument
is the phone number to match an incoming called number.

Step 5 end Returns to privileged EXEC mode.

Example:
Router(config-cm-fallback)# end

Example
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:
call-manager-fallback
no huntstop
alias 1 8005550100 to 5001
alias 2 8005550100 to 5002
alias 3 8005550100 to 5003
alias 4 8005550100 to 5004
pickup 8005550100

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 (5001–5004). 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 softkey.
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 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.
Requires two lines for 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, consider another idle line on the transferor phone. 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, initiates blind transfer instead of the
consultative transfer.

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Configuring Transfer Digit Collection Method

During the consultative transfer, blocks the transferor line to the transferee party on the transferor phone to
prevent being stolen by other phones sharing the same directory number. When you press the Transfer softkey
for consultative transfer, does not display the Transfer softkey while collecting and dialing the digits 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. On meeting the criteria for blind transfer, Cisco Unified SMST stops the consultative
transfer call leg, informs the Cisco IOS Software to transfer the call, and then stops the original call bubble.
Handles the PARK FAC code in the same way as an incoming call which requires applying a ten-second timer
by the Cisco IOS Software.

Note The enhancement, by default, collects the transfer digits from the incoming call leg. If necessary, you
can configure the system to collect the transfer digits from the original call leg. See the Configuring
Transfer Digit Collection Method section.

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 stopping the consultative transfer call leg.
Requires no new configuration to support these enhancements.

Configuring Transfer Digit Collection Method

By default, collects transfer digits from the incoming call leg. To change the transfer digit collection method,
perform the following steps.

Before you begin

• Cisco Unified SRST 4.3
• Cisco Unified Communications Manager 6.0
• Cisco IOS Release 12.4(15)XZ
The Cisco 3200 Series Mobile Access Router does not support SRST.

SUMMARY STEPS
1. enable
2. configure terminal
3. call-manager-fallback
4. transfer-digit-collect {new-call | orig-call}
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
Example: • Enter your password if prompted.
Router# enable

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Configuring Global Prefixes

Command or Action Purpose

Step 2 configure terminal Enters global configuration mode.
Example:
Router# configure terminal

Step 3 call-manager-fallback Enters call-manager-fallback configuration mode.

Example:
Router(config)# call-manager-fallback

Step 4 transfer-digit-collect {new-call | orig-call} Selects the digit-collection method used for consultative
Call Transfers.
Example:
Router(config-cm-fallback)# • new-call : Digits are collected from the incoming call
transfer-digit-collect orig-call leg.
• orig-call : Digits are collected from the original
call-leg. It was the default behavior in versions before
Cisco Unified SRST 4.3.

Step 5 end Returns to privileged EXEC mode.

Example:
Router(config)# end

Example
The following example shows the transfer-digit-collect method set to the legacy value of orig-call:
!
call-manager-fallback
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 extra 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.

SUMMARY STEPS
1. call-manager-fallback

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2. dialplan-pattern tag pattern extension-length length [ extension-pattern extension-pattern ] [no-reg

]
3. exit

DETAILED STEPS

Command or Action Purpose

Step 1 call-manager-fallback Enters call-manager-fallback configuration mode.
Example:
Router(config)# call-manager-fallback

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

Step 3 exit Exits call-manager-fallback configuration mode.

Example:
Router(config-cm-fallback)# exit

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Enabling Digit Translation Rules

Example
The following example shows how to create dial-plan pattern 1 for extension numbers 101–199 with
the phone prefix starting with 4085550. If the following example is set, the router recognizes that
4085550144 matches dial-plan pattern 1. It uses 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.
call-manager-fallback
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
0–4, so that the extension-number range becomes 400–499:
call-manager-fallback
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
801–899 with the phone 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).
call-manager-fallback
dialplan-pattern 2 40855590.. extension-length 3 extension-pattern 8..

Enabling Digit Translation Rules

Digit translation rules can be enabled during Cisco Unified Communications Manager fallback. Translation
rules are a number-manipulation mechanism that performs operations such as automatically adding phone
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 Unified SRST 3.2 and later or Cisco Unified SRST 4.0 and later, use the
configuration described in the Enabling Translation Profiles section instead of using the translate
command as described below. Translation Profiles are new to Cisco Unified 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 phone 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
1. call-manager-fallback
2. translate {called | calling} translation-rule-tag
3. exit

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Enabling Translation Profiles

DETAILED STEPS

Command or Action Purpose

Step 1 call-manager-fallback Enters call-manager-fallback configuration mode.
Example:
Router(config)# call-manager-fallback

Step 2 translate {called | calling} translation-rule-tag Applies a translation rule to modify the phone number dialed
or received by any Cisco Unified IP Phone user while Cisco
Example:
Unified Communications Manager fallback is active.
Router(config-cm-fallback)# translate called 20
• called : Applies the translation rule to an outbound
call number.
• calling : Applies the translation rule to an inbound call
number.
• translation-rule-tag : The reference number of the
translation rule 1–2147483647.

Step 3 exit Exits call-manager-fallback configuration mode.

Example:
Router(config-cm-fallback)# exit

Example
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 Communications Manager fallback.
translation-rule 10
rule 1 1111 2222 abbreviated
exit
call-manager-fallback
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 1–15) 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 Unified 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

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Enabling Translation Profiles

• Redirected called numbers

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 Cisco Unified Communications Manager fallback
mode.
Cisco IP phones support one incoming and one outgoing translation profile when in SRST mode.

Note For Cisco Unified 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. 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-rulenumber
2. rule precedence/match-pattern/ /replace-pattern/
3. exit
4. voice translation-profilename
5. translate {called | calling | redirect-called} translation-rule-number
6. exit
7. call-manager-fallback
8. translation-profile {incoming | outgoing} name
9. exit

DETAILED STEPS

Command or Action Purpose

Step 1 voice translation-rulenumber Defines a translation rule for voice calls and enters voice
translation-rule configuration mode.
Example:
Router(config)# voice translation-rule 1 numbernumber: Number that identifies the translation rule.
Range is 1–2147483647.

Step 2 rule precedence/match-pattern/ /replace-pattern/ Defines a translation rule.

Example: • precedence : Priority of the translation rule. Range is
Router(cfg-translation-rule)# rule 1/^9/ // 1–15.
• match-pattern : Stream editor (SED) expression used
to match incoming call information. The slash (/) is a
delimiter in the pattern.

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Enabling Translation Profiles

Command or Action Purpose

• 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-profilename Defines a translation profile for voice calls.

Example: name : Name of the translation profile. Maximum length
Router(config)# voice translation-profile name1 of the voice translation profile name is 31 alphanumeric
characters.

Step 5 translate {called | calling | redirect-called} Associates a voice translation rule with a voice translation
translation-rule-number profile.
Example: • called : Associates the translation rule with called
Router(cfg-translation-profile)# translate numbers.
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 1–2147483647.

Step 6 exit Exits translation-profile configuration mode.

Example:
Router(cfg-translation-profile)# exit

Step 7 call-manager-fallback Enters call-manager-fallback configuration mode.

Example:
Router(config)# call-manager-fallback

Step 8 translation-profile {incoming | outgoing} name Assigns a translation profile for incoming or outgoing call
legs on a Cisco IP phone.
Example:
Router(config-cm-fallback)# translation-profile • incoming : Applies the translation profile to incoming
outgoing name1 calls.
• outgoing : Applies the translation profile to outgoing
calls.
• name : The name of the translation profile.

Step 9 exit Exits call-manager-fallback configuration mode.

Example:

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Verifying Translation Profiles

Command or Action Purpose

Router(config-cm-fallback)# exit

Example
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

call-manager-fallback
translation-profile incoming name1

Verifying Translation Profiles

Before you begin
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 [ testmatch-type [plan match-type ] ]

DETAILED STEPS

Command or Action Purpose

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. See the
testmatch-type [plan match-type ] ] test voice translation-rule command in Cisco IOS Voice
Command Reference for more information.
Example:
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

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Configuring Dial-Peer and Channel Hunting

Command or Action Purpose

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 Communications Manager 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. 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
1. call-manager-fallback
2. huntstop [channel]
3. exit

DETAILED STEPS

Command or Action Purpose

Step 1 call-manager-fallback Enters call-manager-fallback configuration mode.
Example:
Router(config)# call-manager-fallback

Step 2 huntstop [channel] Sets the huntstop attribute for the dial peers associated with
the Cisco Unified IP Phone dial peers created during Cisco
Example:
Unified Communications Manager fallback.
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.

Step 3 exit Exits call-manager-fallback configuration mode.

Example:
Router(config-cm-fallback)# exit

Example
The following example disables dial-peer hunting during Cisco Unified Communications Manager
fallback and hunting to the secondary channels in dual-line phone configurations:

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Configuring Busy Timeout

call-manager-fallback
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
1. call-manager-fallback
2. timeouts busy seconds
3. exit

DETAILED STEPS

Command or Action Purpose

Step 1 call-manager-fallback Enters call-manager-fallback configuration mode.
Example:
Router(config)# call-manager-fallback

Step 2 timeouts busy seconds Sets the amount of time for disconnecting the calls before
transferring to busy destinations.
Example:
Router(config-cm-fallback)# timeouts busy 20 seconds : Number of seconds. Range is 0–30. Default is 10.
Note This command sets the busy timeout only for
calls before transferring to busy destinations and
does not affect the timeout for calls that directly
dial busy destinations.

Step 3 exit Exits call-manager-fallback configuration mode.

Example:
Router(config-cm-fallback)# exit

Example
The following example sets a timeout of 20 seconds for transferring calls to busy destinations:
call-manager-fallback
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.

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Configuring Outgoing Calls

SUMMARY STEPS
1. call-manager-fallback
2. timeouts ringing seconds
3. exit

DETAILED STEPS

Command or Action Purpose

Step 1 call-manager-fallback Enters call-manager-fallback configuration mode.
Example:
Router(config)# call-manager-fallback

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

Step 3 exit Exits call-manager-fallback configuration mode.

Example:
Router(config-cm-fallback)# exit

Example
The following example sets the ringing timeout default to 30 seconds:
call-manager-fallback
timeouts ringing 30

Configuring Outgoing Calls

Configuring Local and Remote Call Transfer
Configure the Cisco Unified SRST to allow Cisco Unified IP Phones to transfer phone 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.

SUMMARY STEPS
1. call-manager-fallback
2. transfer-pattern transfer-pattern
3. exit

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Enabling Consultative Call Transfer and Forward Using H.450.2 and H.450.3 with Cisco Unified SRST 3.0

DETAILED STEPS

Command or Action Purpose

Step 1 call-manager-fallback Enters call-manager-fallback configuration mode.
Example:
Router(config)# call-manager-fallback

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
Example:
using the specified transfer pattern.
Router(config-cm-fallback)# transfer-pattern
52540.. transfer-pattern : String of digits for permitted call
Transfers. Wildcards are permitted.

Step 3 exit Exits call-manager-fallback configuration mode.

Example:
Router(config-cm-fallback)# exit

Example
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 range
5550100–5550199:
call-manager-fallback
transfer-pattern 55501..

Enabling Consultative Call Transfer and Forward Using H.450.2 and H.450.3
with Cisco Unified 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.
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 Unified 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.

Before you begin

Call Transfer with consultation is available only when an IP phone supports a second line or call instance.
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.

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Enabling Consultative Call Transfer and Forward Using H.450.2 and H.450.3 with Cisco Unified SRST 3.0

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 Unified SRST 3.0

Restrictions
Does not implement a H.450.12 Supplementary Services Capabilities Exchange among routers.

SUMMARY STEPS
1. call-manager-fallback
2. call-forward pattern pattern
3. transfer-system {blind | full-blind | full-consult | local-consult}
4. transfer-pattern transfer-pattern
5. exit
6. (Optional) voice service voip
7. (Optional) h323
8. (Optional) h450 h450-2 timeout {T1 | T2 | T3 | T4}milliseconds
9. (Optional) end

DETAILED STEPS

Command or Action Purpose

Step 1 call-manager-fallback Enters call-manager-fallback configuration mode.
Example:
Router(config)# call-manager-fallback

Step 2 call-forward pattern pattern Specifies the H.450.3 standard for a call forwarding.
Example: pattern : Digits to match for a call forwarding using the
Router(config-cm-fallback)# call-forward H.450.3 standard. If an incoming calling-party number
pattern 4... matches the pattern, it can be forwarded using the 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 | full-consult | Not supported if the transfer-to destination is on the Cisco
local-consult} ATA, Cisco VG224, or an SCCP-controlled FXS port.
Example: Defines the call-transfer method for all lines served by the
Router(config-cm-fallback)# transfer-system Cisco Unified SRST router.
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.

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Enabling Consultative Call Transfer and Forward Using H.450.2 and H.450.3 with Cisco Unified SRST 3.0

Command or Action Purpose

• 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 the phone calls by Cisco Unified IP
Phones to specified phone number patterns.
Example:
Router(config-cm-fallback)# transfer-pattern transfer-pattern : String of digits for permitted Call
52540.. Transfers. Wildcards are allowed.

Step 5 exit Exits call-manager-fallback configuration mode.

Example: Timesaver : Before exiting call-manager-fallback
Router(config-cm-fallback)# exit configuration mode, configure any other parameters that
you must set for the entire Cisco Unified SRST phone
network.

Step 6 (Optional) voice service voip Enters voice service configuration mode.
Example:
Router(config)# voice service voip

Step 7 (Optional) h323 Enters H.323 voice service configuration mode.

Example:
Router(conf-voi-serv)# h323

Step 8 (Optional) h450 h450-2 timeout {T1 | T2 | T3 | Sets timeouts for supplementary service timers, in
T4}milliseconds 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 response.
Default is 2000.
• T2 : Timeout value to wait for a call setup. Default is
5000.
• T3 : Timeout value to wait to initiate response. Default
is 5000.
• T4 : Timeout value to wait for setup of response.
Default is 5000.
• milliseconds : Number of milliseconds. Range is
500–60000.

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Enabling Analog Transfer Using Hookflash and the H.450.2 Standard with Cisco Unified SRST 3.0 or Earlier

Command or Action Purpose

Step 9 (Optional) end Returns to privileged EXEC mode.
Example:
Router(conf-serv-h323)# end

Example
The following example specifies transfer with consultation using the H.450.2 standard for all IP
phones serviced by the Cisco Unified SRST router:
dial-peer voice 100 pots
destination-pattern 9.T
port 1/0/0
dial-peer voice 4000 voip
destination-pattern 4…
session-target ipv4:10.1.1.1
call-manager-fallback
transfer-pattern 4…
transfer-system full-consult

The following example enables call forwarding using the H.450.3 standard:

dial-peer voice 100 pots

destination-pattern 9.T
port 1/0/0
!
dial-peer voice 4000 voip
destination-pattern 4
session-target ipv4:10.1.1.1
!
call-manager-fallback
call-forward pattern 4

Enabling Analog Transfer Using Hookflash and the H.450.2 Standard with Cisco
Unified 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 transfer. Hookflash refers to the short on-hook period generated
by a telephone-like device during a call to indicate that the phone is attempting to perform the dial-tone recall
from a PBX. Uses Hookflash 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. Download the script
app-h450-transfer.tcl 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. On collecting all digits, the delay timer starts. The call setup to the

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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 blind transfer rather than 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 consultative transfer.

In addition to the Tcl script, a ReadMe file describes the script and the configurable attribute-value pairs.
Read this file whenever you download a new version of the script because it may contain more script-specific
information, such as configuration parameters and user interface descriptions.

Note For Cisco Unified 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.

Restrictions
• When 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.

Before you begin

Download the H.450 Tcl script named app-h450-transfer.Tcl 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 Unified SRST V3.0 or a lower version
• 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.

SUMMARY STEPS
1. call application voice application-name location
2. (Optional) call application voice application-name language number language

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3. call application voice application-name set-location language category location

4. (Optional) call application voice application-name delay-time seconds
5. dial-peer voice number pots
6. application application-name
7. exit
8. dial-peer voice number voip
9. application application-name
10. exit

DETAILED STEPS

Command or Action Purpose

Step 1 call application voice application-name location Loads the Tcl script and specifies its application name.
Example: • application-name : User-defined name for the IVR
Router(config)# call application voice application. This name does not have to match the
transfer_app flash:app-h450-transfer.tcl script filename.
• location : Script directory and filename in URL
format. For example, flash memory (flash:filename),
a TFTP (tftp://../filename), or an HTTP server
(http://../filename) are valid locations.

Step 2 (Optional) call application voice application-name Sets the language for dynamic prompts by the application.
language number language
• application-name : IVR application name that was
Example: assigned in Step 1.
Router(config)# call application voice
transfer_app language 1 en
• number : Specify the number of languages for 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 set-location Defines the location and category of the audio files that
language category location are used by the application for dynamic prompts.
Example: • application-name : Name of the Tcl IVR application.
Router(config)# call application voice
transfer_app set-location en 0 flash:/prompts
• language : Two-character code to specify the
language of the prompts. Valid entries are en (English:
default), sp (Spanish), ch (Chinese), or aa (all).
• category : Category group (0–4) for the audio files
from this location. The value 0 means all categories.
.
• location : URL of the directory that contains the
language audio files in the application, without
filenames. Flash memory (flash) or a directory on a
server (TFTP, HTTP, or RTSP) are all valid.

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Command or Action Purpose

Prompts are required for Call Transfer from analog FXS
phones. No prompts are needed for Call Transfer from IP
phones.

Step 4 (Optional) call application voice application-name Sets the delay time for consultation call setup for an analog
delay-time seconds phone that is making a Call Transfer using the H.450
application. This command passes a value to the Tcl script
Example:
by using an attribute-value (AV) pair.
Router(config)# call application voice
transfer_app delay-time 1 • seconds : Number of seconds to delay call setup.
Range is 1–10. Default is 2.

Delay of more than 2 seconds is noticeable to users.

For more information about attribute-value 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.

Example: Timesaver : Before exiting dial-peer configuration mode,
Router(config-dial-peer)# exit configure any other dial-peer parameters that you must set
for this dial peer.

Step 8 dial-peer voice number voip Enters dial-peer configuration mode to configure a VoIP
dial peer.
Example:
Router(config)# dial-peer voice 29 voip

Step 9 application application-name Loads the application named in Step 1 onto the dial peer.
Example:
Router(config-dial-peer)# application
transfer_app

Step 10 exit Exits dial-peer configuration mode.

Example: Timesaver : Before exiting dial-peer configuration mode,
Router(config-dial-peer)# exit configure any other dial-peer parameters that you must set
for this dial peer.

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Example
The following example enables the H.450 Tcl script for analog transfer using hookflash and sets
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
!
dial-peer voice 25 pots
destination-pattern 9.T
port 1/0/0
application transfer_app
!
dial-peer voice 29 voip
destination-pattern 4…
session-target ipv4:10.1.10.1
application transfer_app

Configuring Trunk Access Codes

Note Configure trunk access codes only if your normal network dial-plan configuration prevents you from
configuring a permanent POTS voice dial peer 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 Communications Manager
fallback by creating POTS voice dial peers that are active during Cisco Unified Communications Manager
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 Communications Manager 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
1. call-manager-fallback
2. access-code { { fxo | e&m } dial-string | { bri | pri } dial-string [ direct-inward-dial ] }
3. exit

DETAILED STEPS

Command or Action Purpose

Step 1 call-manager-fallback Enters call-manager-fallback configuration mode.
Example:
Router(config)# call-manager-fallback

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Command or Action Purpose

Step 2 access-code { { fxo | e&m } dial-string | { bri | pri } Configures trunk access codes for each type of line so that
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 : Enables Direct Inward Dialing
(DID) on the POTS dial peer.

Step 3 exit Exits call-manager-fallback configuration mode.

Example:
Router(config-cm-fallback)# exit

Example
The following example creates access code number 8 for BRI and enables DID on the POTS dial
peer:
call-manager-fallback
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 stopped.

SUMMARY STEPS
1. call-manager-fallback
2. (Optional) timeouts interdigit seconds
3. exit

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DETAILED STEPS

Command or Action Purpose

Step 1 call-manager-fallback Enters call-manager-fallback configuration mode.
Example:
Router(config)# call-manager-fallback

Step 2 (Optional) timeouts interdigit seconds Configures the interdigit timeout value for all Cisco IP
phones that are attached to the router.
Example:
Router(config-cm-fallback)# timeouts interdigit seconds : Interdigit timeout duration, in seconds, for all
5 Cisco Unified IP Phones. Valid entries are integers from 2
to 120.

Step 3 exit Exits call-manager-fallback configuration mode.

Example:
Router(config-cm-fallback)# exit

Example
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.
call-manager-fallback
timeouts interdigit 5

Configuring Class of Restriction

The class of restriction (COR) functionality provides the ability to deny a certain call attempt on the basis of
the incoming and outgoing class of restrictions that are provisioned on the dial peers. This functionality
provides flexibility in the network design, allows you 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 the dial peers associated with the directory numbers that are created during Cisco Unified
Communications Manager 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 the COR list numbers or number ranges. An assigned COR is invoked for the dial
peers and created for each directory number automatically during Cisco Unified 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 goes through. Voice ports determine whether a
call is considered incoming or outgoing. If you hook up a phone to an FXS port on a Cisco Unified SRST
router and try to call from that phone, the call will be considered an incoming call to the router and voice port.
If you call the FXS phone, consider it as an outgoing call.
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 call from a phone that is attached

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to the dial peer, so that the call goes out of any dial peer regardless of the COR configuration on that dial peer.
The following table describes the call functionality that is based on your COR lists configuration.

COR List on Incoming Dial COR List on Outgoing Dial Result

Peer Peer

No COR No COR The call succeeds.

No COR COR list applied for The call succeeds. By default, the incoming dial
outgoing calls peer 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 call of
any other dial peer regardless of the COR
configuration on the outgoing dial peer.

COR list applied for No COR The call succeeds. By default, the outgoing dial
incoming calls peer 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 stopping dial peer.

COR list applied for COR list applied for The call succeeds. The COR list for incoming
incoming calls (superset of outgoing calls (subsets of a calls on the incoming dial peer is a superset of
a COR list applied for COR list applied for the COR list for outgoing calls on the outgoing
outgoing calls on the incoming calls on the dial peer.
outgoing dial peer) incoming dial peer)

COR list applied for COR list applied for The call does not succeed. The COR list for
incoming calls (subset of a outgoing calls (supersets of incoming calls on the incoming dial peer is not
COR list applied for a COR list applied for a superset of the COR list for outgoing calls on
outgoing calls on the incoming calls on the the outgoing dial peer.
outgoing dial peer) incoming dial peer)

SUMMARY STEPS
1. call-manager-fallback
2. cor {incoming | outgoing} cor-list-name [ cor-list-number starting-number - ending-number | default
]
3. exit

DETAILED STEPS

Command or Action Purpose

Step 1 call-manager-fallback Enters call-manager-fallback configuration mode.
Example:
Router(config)# call-manager-fallback

Step 2 cor {incoming | outgoing} cor-list-name [ cor-list-number Configures a COR on dial peers that are associated with
starting-number - ending-number | default ] directory numbers.

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Command or Action Purpose

Example: • incoming : COR list to be used by incoming dial
Router(config-cm-fallback)# cor outgoing peers.
LockforPhoneC 1 5010 – 5020
• outgoing : COR list to be used by outgoing dial peers.
• cor-list-name : COR list name.
• cor-list-number : COR list identifier. You can create
maximum 20 COR lists, comprising of incoming or
outgoing dial peers. The first six COR lists are applied
to range of directory numbers. Assign the COR
configuration to the default COR list if the directory
numbers do not have a COR configuration.
• starting-number- ending-number: : Directory number
range; for example, 2000–2025.
• default : Instructs the router to use an existing default
COR list.

Step 3 exit Exits call-manager-fallback configuration mode.

Example:
Router(config-cm-fallback)# exit

Example
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 that are created during fallback:
call-manager-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:
call-manager-fallback
cor incoming LockforPhoneC default

The following example shows creation of a sub- and super-COR sets. First, create a custom dial-peer
COR with declared names under it:
dial-peer cor custom
name 911
name 1800
name 1900
name local_call

The following configuration example creates the COR lists and applies to the dial peer:
dial-peer cor list call911
member 911
dial-peer cor list call1800
member 1800
dial-peer cor list call1900
member 1900
dial-peer cor list calllocal

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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

The following example configures five dial peers for destination numbers 734….,
1800…….,1900……., 316…., and 911. A COR list is applied to each of the dial peers.
dial-peer voice 1 voip
destination pattern 734....
session target ipv4:10.1.1.1
cor outgoing calllocal
dial-peer voice 2 voip
destination pattern 1800.......
session target ipv4:10.1.1.1
cor outgoing call1800
dial-peer voice 3 pots
destination pattern 1900.......
port 1/0/0
cor outgoing call1900
dial-peer voice 5 pots
destination pattern 316....
port 1/1/0
! No COR is applied.
dial-peer voice 4 pots
destination pattern 911
port 1/0/1
cor outgoing call911

Finally, the COR list is applied to the individual phone numbers.

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

The sample configuration allows for the following:

• Extension 1001 to call 734... numbers, 911, and 316....
• Extension 1002 to call 734..., toll-free numbers, 911, and 316....
• Extension 1003–1008 to call all 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 specified time of day and day of the week or date. Specify up to 32 patterns of digits. Supports call
blocking on IP phones only and not on analog Foreign Exchange Station (FXS) phones.

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When you call to digits that match a pattern for call blocking during a defined time period for a call blocking,
fast busy signal plays for approximately 10 seconds. The call stops and places the line 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
1. call-manager-fallback
2. after-hours block pattern tag pattern [ 7-24 ]
3. after-hours day day start-time stop-time
4. after-hours date month date start-time stop-time
5. exit

DETAILED STEPS

Command or Action Purpose

Step 1 call-manager-fallback Enters call-manager-fallback configuration mode.
Example:
Router(config)# call-manager-fallback

Step 2 after-hours block pattern tag pattern [ 7-24 ] For blocking, defines a pattern of outgoing digits. Define
up to 32 patterns, using individual commands.
Example:
Router(config-cm-fallback)# after-hours block • If you specify the 7–24 keyword, always blocks the
pattern 1 91900 pattern, 7 days a week, 24 hours a day.
• If you do not specify the 7–24 keyword, blocks the
pattern during the days and dates as defined in the
after-hours day and after-hours date commands.

Step 3 after-hours day day start-time stop-time Defines a recurring time period for the day of the week
during which calls are blocked to outgoing dial patterns that
Example:
are defined using the after-hours block pattern command.
Router(config-cm-fallback)# after-hours day mon
19:00 7:00 • 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 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.”.

Step 4 after-hours date month date start-time stop-time Defines a recurring time period for the month and date for
blocking calls to outgoing dial patterns defined in the
Example:
after-hours block pattern command.
Router(config-cm-fallback)# after-hours date
jan 1 0:00 0:00

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Command or Action Purpose

• month : Month abbreviation. The following are valid
month abbreviations: jan, feb, mar, apr, may, jun, jul,
aug, sep, oct, nov, dec.
• date : Date of the month. Range is 1–31.
• start-time stop time : Beginning and ending times for
call blocking, in an HH:MM format using a 24-hour
clock. The stop time must be larger than the start time.
Value 24:00 is not valid. If you enter 00:00 as stop
time, it changes to 23:59. If you enter 00:00 for both
start time and stop time, blocks call for the entire
24-hour period on the specified date.

Step 5 exit Exits call-manager-fallback configuration mode.

Example:
Router(config-cm-fallback)# exit

Example
The following example defines several patterns of digits for which blocks outgoing calls. Patterns 1
and 2, blocks call to external numbers that begin with “1” and “011”:
• On Monday through Friday before 7 a.m. and after 7 p.m.
• On Saturday before 7 a.m. and after 1 p.m.
• All day Sunday.

Pattern 3 blocks call to 900 numbers 7 days a week, 24 hours a day.

call-manager-fallback
after-hours block pattern 1 91
after-hours block pattern 2 9011
after-hours block pattern 3 91900 7-24
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

How to Configure Cisco Unified SIP SRST

Configuring SIP Phone Features
After setting the voice register Pool, the procedure adds optional features to increase functionality. Some
features are per Pool or globally.

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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 the voice register global mode, you can globally assign characteristics to phones.

SUMMARY STEPS
1. enable
2. configure terminal
3. voice register global tag
4. max-pool max-voice-register-pools
5. application application-name
6. external-ring {bellcore-dr1 | bellcore-dr2 |bellcore-dr3 | bellcore-dr4 | bellcore-dr5}
7. exit
8. voice register pool tag
9. no vad
10. codec codec-type [bytes]
11. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
Example: • Enter your password if prompted.
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 voice register global tag Enters voice register global configuration mode to set
global parameters for all supported Cisco SIP IP phones
Example:
in a Cisco Unified SIP SRST environment.
Router(config)# voice register global 12

Step 4 max-pool max-voice-register-pools Set the maximum number of supported SIP voice register
Pools in a Cisco Unified SIP SRST environment.
Example:
Router(config-register-global)# max-pool 10 The max-voice-register-pools argument represents the
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
Example:
argument to define specific interactive voice response
Router(config-register-global)# application (IVR) application.
global_app

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Command or Action Purpose

Step 6 external-ring {bellcore-dr1 | bellcore-dr2 |bellcore-dr3 Specifies the type of ring sound on Cisco SIP or Cisco
| bellcore-dr4 | bellcore-dr5} SCCP IP phones for external calls. Each bellcore-dr 1–5
keyword supports standard distinctive ringing patterns as
Example:
defined in the standard GR-506-CORE, LSSGR: Signaling
Router(config-register-global)# external-ring for Analog Interfaces.
bellcore-dr1

Step 7 exit Exits voice register global configuration mode.

Example:
Router(config-register-global)# exit

Step 8 voice register pool tag Enters voice register Pool configuration mode for SIP
phones.
Example:
Router(config)# voice register pool 20 Use this command to control which phone registrations
are accepted or rejected by a Cisco Unified SIP SRST
device.

Step 9 no vad Disables voice activity detection (VAD) on the VoIP dial
peer.
Example:
Router(config-register-pool)# no vad VAD is enabled by default. Because there is no comfort
noise during periods of silence, the call may disconnect.
You may prefer to set no VAD on the SIP phone pool.

Step 10 codec codec-type [bytes] Specifies the supported codec by a single SIP phone or a
VoIP dial peer in a Cisco Unified SIP SRST environment.
Example:
The codec-type argument specifies the preferred codec
Router(config-register-pool)# codec g729r8 and 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.

Step 11 end Returns to privileged EXEC mode.

Example:
Router(config-register-pool)# end

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 voicemail systems,
or an auto attendant or IVR system such as Cisco Unified Contact Center and Cisco Unified Contact Center
Express). In addition, SCCP IP phones may be forwarded to SIP phones.

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Configuring SIP-to-SIP Call Forwarding

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. Use any of
the following commands to configure the 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]

A typical Cisco Unified SIP SRST setup does not use the call-forward b2bua mailbox command. However,
Cisco Unified SIP Cisco Unified Communications Manager Express environment uses this command. You
can find the detailed procedures for configuring the call-forward b2bua mailbox command in the Cisco Unified
Communications Manager documentation on Cisco.com.

SUMMARY STEPS
1. enable
2. configure terminal
3. voice register pool tag voip
4. encall-forward b2bua alld 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

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
Example: • Enter your password if prompted.
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 voice register pool tag voip Enters voice register Pool configuration mode.

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Command or Action Purpose

Example: Use this command to control the acceptance or rejections
Router(config)# voice register pool 15 of the phone registrations on a Cisco Unified SIP SRST
device.

Step 4 encall-forward b2bua alld directory-number Enables call forwarding for a SIP back-to-back user agent
(B2BUA) to forward all incoming calls to another non-SIP
Example:
station extension. Namely to SIP trunk, H.323 trunk, SCCP
Router(config-register-pool)# call-forward device, and analog or 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.

Step 5 call-forward b2bua busy directory-number Enables call forwarding for a SIP B2BUA to forward
incoming calls to a busy extension to another extension.
Example:
Router(config-register-pool)# call-forward directory-number : phone number to which calls are
b2bua busy 5006 forwarded. Represents a fully qualified E.164 number.
Maximum length of the phone number is 32.

Step 6 call-forward b2bua mailbox directory-number Controls the specific voicemail box in a voicemail system
at the end of a call forwarding Exchange.
Example:
Router(config-register-pool)# call-forward directory-number : phone number to which calls are
b2bua mailbox 5007 forwarded. Represents a fully qualified E.164 number.
Maximum length of the phone number is 32.

Step 7 call-forward b2bua noan directory-number timeout Enables call forwarding for a SIP B2BUA to forward
seconds incoming calls to an extension that does not answer after a
configured amount of time to another extension.
Example:
Router(config-register-pool)# call-forward Use this command if a phone is registered with a Cisco
b2bua noan 5010 timeout 10 Unified SIP SRST router, but the phone is not reachable
because there is no IP connectivity (there is no response to
Invite requests).
• directory-number : phone number to which calls are
forwarded. Represents a fully qualified E.164 number.
Maximum length of the phone number is 32.
• timeout seconds : Duration, in seconds, that a call
can ring with no answer before the call is forwarded
to another extension. Range is 3–60000. The default
value is 20.

Step 8 end Returns to privileged EXEC mode.

Example:
Router(config-register-pool)# end

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Configuring Call Blocking Based on Time of Day, Day of Week, or Date

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. It
is implemented by matching a pattern of up to 32 digits during specified time of day, day of the week, or date.
Cisco Unified SIP SRST provides SIP endpoints the same time-based call blocking mechanism as provided
for SCCP phones. The call blocking feature supports all incoming calls, including incoming SIP and analog
FXS calls.

Note The Cisco Unified SIP SRST does not support the Pin-based exemptions and the “Login” toll-bar
override.

Use the same commands for SIP phone call blocking and for SCCP phones on your Cisco Unified SRST
system. The Cisco Unified SRST session application accesses the current after-hours configuration under
call-manager-fallback mode. It applies to calls originated by Cisco SIP phones and registered to the Cisco
Unified SRST router. The commands used in call-manager-fallback mode that set block criteria (time or date
or 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 you call to digits that match the specified patterns for call blocking during a defined time period for
call blocking, the call stops 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
2. configure terminal
3. call-manager-fallback
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
8. voice register pool tag
9. after-hour exempt
10. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
Example: • Enter your password if prompted.
Router> enable

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Command or Action Purpose

Step 2 configure terminal Enters global configuration mode.
Example:
Router# configure terminal

Step 3 call-manager-fallback Enters call-manager-fallback configuration mode.

Example:
Router(config)# call-manager-fallback

Step 4 after-hours block pattern tag pattern [ 7-24 ] For blocking, defines a pattern of outgoing digits. Define
up to 32 patterns, using individual commands.
Example:
Router(config-cm-fallback)# after-hours block • If you specify the 7–24 keyword, always blocks the
pattern 1 91900 pattern, 7 days a week, 24 hours a day.
• If you do not specify the 7–24 keyword, blocks the
pattern during the days and dates as defined in the
after-hours day and after-hours date commands.

Step 5 after-hours day day start-time stop-time Defines a recurring time period for the day of the week
during which calls are blocked to outgoing dial patterns
Example:
that are defined using the after-hours block pattern
Router(config-cm-fallback)# after-hours day mon command.
19:00 7:00
• 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 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.”.
Value 24:00 is not valid. If you enter 00:00 as stop
time, it changes to 23:59. If you enter 00:00 for both
start time and stop time, blocks call for the entire
24-hour period on the specified date.

Step 6 after-hours date month date start-time stop-time Defines a recurring time period for the month and date for
blocking calls to outgoing dial patterns defined in the
Example:
after-hours block pattern command.
Router(config-cm-fallback)# after-hours date
jan 1 0:00 0:00 • month : Month abbreviation. The following are valid
month abbreviations: jan, feb, mar, apr, may, jun, jul,
aug, sep, oct, nov, dec.
• date : Date of the month. Range is 1–31.

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Command or Action Purpose

• start-time stop time : Beginning and ending times
for call blocking, in an HH:MM format using a
24-hour clock. The stop time must be larger than the
start time.
Value 24:00 is not valid. If you enter 00:00 as stop
time, it changes to 23:59. If you enter 00:00 for both
start time and stop time, blocks call for the entire
24-hour period on the specified date.

Step 7 exit Exits call-manager-fallback configuration mode.

Example:
Router(config-cm-fallback)# exit

Step 8 voice register pool tag Enters voice register Pool configuration mode.
Example: Use this command to control the accepted or rejected
Router(config)# voice register pool 12 registrations by a Cisco Unified SIP SRST device.

Step 9 after-hour exempt Specifies that for a particular voice register Pool, does not
block the outgoing calls although call blocking is enabled.
Example:
Router(config-register-pool)# after-hour exempt

Step 10 end Returns to privileged EXEC mode.

Example:
Router(config-register-pool)# end

Example
The following example defines several patterns of digits for which blocks outgoing calls. Patterns 1
and 2, blocks call to external numbers that begin with “1” and “011”:
• On Monday through Friday before 7 a.m. and after 7 p.m.
• On Saturday before 7 a.m. and after 1 p.m.
• All day Sunday.

Pattern 3 blocks call to 900 numbers 7 days a week, 24 hours a day.

call-manager-fallback
after-hours block pattern 1 91
after-hours block pattern 2 9011
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:

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Verification

voice register pool 1

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 pooltag : 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 Command
Reference (All Versions).

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. It also includes support for consultative hold where A calls B, B place A on hold, B calls C, and B
disconnects from C and then resumes with A. Support for a call hold is signaled by SIP phones using
“re-INVITE c=0.0.0.0” and also by the receive-only mechanism.
No configuration is necessary.

Router# show running-config

Building configuration...
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
!
!

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SIP Call Hold and Resume

voice service voip

allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
sip
registrar server expires max 600 min 60
!
!
!
voice register global
max-dn 10
max-pool 10
!
! Define call forwarding under a voice register pool
voice register pool 1
id mac 0012.7F57.60AA
number 1 1000
call-forward b2bua busy 2413
call-forward b2bua noan 2414 timeout 30
codec g711ulaw
!
voice register pool 2
id mac 0012.7F3B.9025
number 1 2800
codec g711ulaw
!
voice register pool 3
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
!
!
!
dial-peer voice 1000 voip
destination-pattern 24..
session protocol sipv2
session target ipv4:10.0.2.5
codec g711ulaw
!
! Define call blocking under call-manager-fallback mode
call-manager-fallback

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 Configuring Call Handling
How to Configure Optional Features

max-conferences 4 gain -6
after-hours block pattern 1 2417
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
ntp server 10.0.2.10
!
end

How to Configure Optional Features

This section describes the following optional more call features:
• Three-party G.711 ad hoc conferencing—Cisco Unified Survivable Remote Site Telephony (SRST)
support for simultaneous three-party conferences.
• 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
• Defining XML API Schema

Enabling Three-Party G.711 Ad Hoc Conferencing

The enabling three-party G.711 ad hoc conferencing involves configuring the maximum number supported
simultaneous three-party conferences by the Cisco Unified SRST router. For conferencing to be available,
connect minimum of two lines to one or more buttons in an IP phone. See the Configuring a Secondary Dial
Tone section.

SUMMARY STEPS
1. call-manager-fallback
2. max-conferences max-conference-numbers
3. exit

DETAILED STEPS

Command or Action Purpose

Step 1 call-manager-fallback Enters call-manager-fallback configuration mode.
Example:
Router(config)# call-manager-fallback

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Defining XML API Schema

Command or Action Purpose

Step 2 max-conferences max-conference-numbers Sets the maximum number of supported simultaneous
three-party conferences by the router. The maximum number
Example:
possible is platform-dependent:
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 3725 router: 16
• Cisco 3745 router: 16
• Cisco 3800 Series router: 24

Step 3 exit Exits call-manager-fallback configuration mode.

Example:
Router(config-cm-fallback)# exit

Example
The following example configures up to eight simultaneous three-way conferences on a router:
call-manager-fallback
max-conferences 8

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 Unified SRST 3.0 section for configuration
instructions.

SUMMARY STEPS
1. call-manager-fallback
2. xmlschema schema-url
3. exit

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Configuration Examples for Call Handling

DETAILED STEPS

Command or Action Purpose

Step 1 call-manager-fallback Enters call-manager-fallback configuration mode.
Example:
Router(config)# call-manager-fallback

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

Step 3 exit Exits call-manager-fallback configuration mode.

Example:
Router(config-cm-fallback)# exit

Configuration Examples for Call Handling

Example: Monitoring the Status of Key Expansion Modules
Use the Show commands 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 Communications Manager Express configurations and registration information:
show voice register all
VOICE REGISTER GLOBAL
=====================
CONFIG [Version=9.1]
========================
............
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
configured phones with add-on KEMs in Cisco Unified Communications Manager Express:

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 Configuring Call Handling
Example: Configuring Voice Hunt Groups in Cisco Unified SIP SRST

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 directing a call six times (makes 6 hops), it is redirected to the
final number 5000.
Router(config)# voice hunt-group 20 longest-idle
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)# hops 6
Router(config-voice-hunt-group)# timeout 20
Router(config-voice-hunt-group)# exit

Where to Go Next
If you must configure security, see the section, or if you must configure voicemail, see the Integrating Voice
Mail with Cisco Unified SRST section. If you must configure video parameters, see the Setting Video
Parameters section. If you do not need any of those features, go to the Monitoring and Maintaining Cisco
Unified SRST section.
For additional information, see the Related Documents section in the Cisco Unified SCCP and SIP SRST
Feature Overview chapter.

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Where to Go Next

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 CHAPTER 10
Configuring Secure SRST for SCCP and SIP
The Secure SRST adds security functionality to the Unified SRST.

Note Unified Secure SRST 12.6 on Cisco IOS XE Gibraltar 16.11.1a Release is not a recommended release
version for Unified Secure SCCP SRST call flows and call flows that include stcapp configuration.

• Prerequisites for Configuring Secure SRST, on page 265

• Restrictions for Configuring Secure SRST, on page 266
• Information About Configuring Secure SRST, on page 268

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.
• 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.
• 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.

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 Configuring Secure SRST for SCCP and SIP
Restrictions for Configuring Secure SRST

• 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.

Restrictions for Configuring Secure SRST

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.

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.

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Configuring Secure SRST for SCCP and SIP
Restrictions for Configuring Secure SRST

• 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.

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

Information About Configuring Secure SRST

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
• 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.

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Secure Music On Hold for Unified Secure SRST (SIP)

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.
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.

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 VG204 Analog Voice Gateway
• Cisco VG204XM Analog Voice Gateway
• Cisco VG224 Analog Voice Gateway
• Cisco VG300 Series Gateways (VG310, VG320, VG350)

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.

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Secure Music On Hold for Secure Unified SRST (SCCP)

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.

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.
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.

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Feature Support for Secure SRST (SCCP), Unified SRST Release 12.3

• 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 SRST.
• 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

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Cisco IP Phones Clear-Text Fallback During Non-Secure SRST

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
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 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

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SRST Routers and the TLS Protocol

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
v1.1 Enable TLS Version 1.1
v1.2 Enable TLS Version 1.2

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 ?
v1.0 Enable TLS Version 1.0
v1.1 Enable TLS Version 1.1
v1.2 Enable TLS Version 1.2

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TLS Cipher Support for Secure SRST 12.6 and Later Releases

TLS Cipher Support for Secure SRST 12.6 and Later Releases
From Unified Secure SRST 12.6 onwards, the TLS cipher support offered on Secure SRST is modified to
enhance security.

TLS Cipher Support for SCCP/TLS (Ports 2443 and 2445)

The following cipher suites are supported (offer and accept):

Note ECDSA cipher is not supported with Secure SRST.

• TLS_RSA_WITH_AES_128_CBC_SHA
• TLS_RSA_AES_GCM_SHA2

The following cipher suites are not supported:

• TLS_RSA_WITH_NULL_SHA

TLS Cipher Support for SIP/TLS (Port 5061)

The following cipher suites are supported (offer and accept):
• TLS_RSA_WITH_AES_128_CBC_SHA
• TLS_ECDHE_RSA_WITH_AES_128_GCM_SHA256
• TLS_ECDHE_ECDSA_WITH_AES_128_GCM_SHA256
• TLS_ECDHE_RSA_WITH_AES_256_GCM_SHA384
• TLS_ECDHE_ECDSA_WITH_AES_256_GCM_SHA384

The following cipher suites are not supported:

• TLS_RSA_WITH_RC4_128_MD5
• TLS_DHE_RSA_WITH_AES_128_CBC_SHA1

Certificates Operation on Secure SRST

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. The following table shows the secure
SRST-supported Cisco Unified IP Phones and the appropriate certificate for each phone. The Additional
References section 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).

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Cisco IOS Credentials Server on Secure SRST Routers

Note Cisco supports Cisco IP Phones 7900 series phone memory reclamation phones that use MIC or locally
significant certificate (LSC) certificates.

Table 1: Supported Cisco Unified IP Phones and Certificates

Cisco Unified IP Phone 7940 Cisco Unified IP Phone 7960 Cisco Unified IP Phone 7970

The phone receives locally The phone receives locally The phone contains a
significant certificate (LSC) from significant certificate (LSC) from manufacturing installed certificate
Certificate Authority Proxy Certificate Authority Proxy (MIC) used for device
Function (CAPF) in Distinguished Function (CAPF) in Distinguished authentication. If the Cisco 7970
Encoding Rules (DER) format. Encoding Rules (DER) format. implements MIC, two public
certificate files are needed:
59fe77ccd.0 59fe77ccd.0
CiscoCA.pem (Cisco Root CA,
The filename may change based on The filename may change based on
used to authenticate the certificate.)
the CAPF certificate subject name the CAPF certificate subject name
and the CAPF certificate issuer. and the CAPF certificate issuer. Note The name of the
manufacturing
If Cisco Unified Communications If Cisco Unified Communications
certificate can vary
Manager is using a third-party Manager is using a third-party
depending on your
certificate provider, there can be certificate provider, there can be
configuration.
multiple .0 files (from two to ten). multiple .0 files (from two to ten).
Each .0 certificate file must be Each .0 certificate file must be
a69d2e04.0, in Privacy Enhanced
imported individually during the imported individually during the
Mail (PEM) format
configuration. configuration.
If Cisco Unified Communications
Manual enrollment supported only Manual enrollment supported only.
Manager is using a third-party
certificate provider, there can be
multiple .0 files (from two to ten).
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)

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Generating a Certificate for the Credentials Server

Two Cisco IOS commands provide credential server debugging and verification capabilities:
• debug credentials
• show credentials

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 " 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
• Enabling Credentials Service on the Secure Cisco Unified SRST Router
• Configuring SRST Fallback on Cisco Unified Communications Manager

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.

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.

SIP OAuth Client Registration for Unified Secure SRST

Unified Secure SIP SRST enables routers to provide secure call-handling for Unified IP phones during an
outage. The support is for endpoints that lose connection to the remote primary, secondary, or tertiary Unified
CM installations during a WAN outage. If SIP OAuth is configured, SIP clients can securely register to the
SRST during WAN link failures. The SRST can provide secure call control for the following SIP clients:
• Cisco Jabber Client
• Cisco Webex Client
• Cisco IP Phone 78xx Series
• Cisco IP Phone 88xx Series

Dynamic, token-based authentication provides improved security for devices registering to Unified CM.
When registering to the SRST during an outage, a client uses the authentication token that is issued by UCM.
A challenge is issued when a new registration request does not include a token. The SRST attempts to validate
the token using keys previously received securely from UCM. If the validation is successful, the SRST allows
the client to register and place calls locally. Clients presenting a token that cannot be validated by the SRST
are not allowed to register.

Note Key pairs are stored in persistent memory, ensuring that clients can register if the SRST router reloads
during a service outage.

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Feature Characteristics

To configure SIP OAuth for the Unified Secure SIP SRST, perform the following:
1. Configure a TLS listen port without client validation for use by SIP OAuth clients.

Note The TLS listen port is open in addition to the default secure port that uses mTLS.

voice service voip

sip
listen-port secure no-client-validation ?
<1024-49151> Port number

2. Perform call service stop before configuring the listen port and no call service stop after configuring the
listen port.
3. Configure access to the UCM key server with appropriate authentication details. Passwords that are entered
in clear text are stored using type 6 encryption.
voice register global
sip-oauth SIP OAuth parameters for Unified SRST
key-server key-server ipv4:10.5.10.50:8443 username administrator password 0
abcd12345

4. Configure device pools for compatible clients to use SIP OAuth —Enables SIP OAuth for compatible
clients using the voice register pool configuration.
voice register pool <tag>
sip-oauth

Feature Characteristics
• SRST is configured to use a TLS socket without mTLS validation for clients that use SIP OAuth.
• Registration using SIP OAuth is enabled for clients through their voice register pool configuration.
• Cisco Unified SRST accepts new registration from clients with a valid SIP OAuth token.
• Protocol mode should be either "IPV4 only " or "IPV6 only" for SIP OAuth.

Restrictions
ECDSA cipher suite is not supported on port 2445.

Configure SIP OAuth-based Listener Port

SUMMARY STEPS
1. configure terminal
2. voice service voip
3. sip
4. call service stop
5. listen-port secure no-client-validation <1024-49151>
6. no call service stop

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Retrieve SIP OAuth Keys from CUCM

DETAILED STEPS

Command or Action Purpose

Step 1 configure terminal Enters global configuration mode.
Example:

Router#conf t

Step 2 voice service voip Enters voice service VoIP mode.

Example:
Router(config)#voice service voip

Step 3 sip Enters voice service VoIP sip mode.

Example:
Router(conf-voi-serv)#sip

Step 4 call service stop Shuts down VoIP call service on a gateway.

Step 5 listen-port secure no-client-validation <1024-49151> Configures a TLS listen port with mTLS disabled.
Example: Note Default port is 5090.
Router(conf-serv-sip)#listen-port secure
no-client-validation 5090

Step 6 no call service stop Enables VoIP call service.

Retrieve SIP OAuth Keys from CUCM

Note Execute voice sip oauth get-keys to retrieve sip-oauth keys anytime from CUCM.

SUMMARY STEPS
1. voice register global
2. sip-oauth
3. key-server word username word password 0/6 word
4. key-server source-interface <options>

DETAILED STEPS

Command or Action Purpose

Step 1 voice register global Enters voice register global configuration mode.
Example:
Router(config)#voice register global

Step 2 sip-oauth Enables SIP OAuth feature.

Example:

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Command or Action Purpose

Router(config-register-global)#sip-oauth
Router(config-oauth)#

Step 3 key-server word username word password 0/6 word Configures key-server details for SIP OAuth. The key server
provides the keys in JSON format to authenticate the token
Example:
sent by phones. The key-server address is usually the
voice register global CUCM IP address. The <word> must be in one of the
sip-oauth
key-server ipv4:10.5.10.50:8443 username following formats:
administrator password 0 C1sco123=
ipv4:X.X.X.X
ipv4:X.X.X.X:port-number
ipv6:[X:X:X:X:X:X]
ipv6:[X:X:X:X:X:X]:port-number
dns:hostname.com
dns:hostname.com:port-number
Note If the port is not configured, then 443 secure port
is used for HTTPS communication.

Note If dns hostname is configured with a port, then

SRV query is performed.

Step 4 key-server source-interface <options> (Optional) Configures interface specification of source

address for OAuth server.

Enable SIP OAuth-based Registration

SUMMARY STEPS
1. voice register pool tag
2. sip-oauth
3. end

DETAILED STEPS

Command or Action Purpose

Step 1 voice register pool tag Enters voice register pool configuration mode.
Example:
Router(config)#voice register pool 20
Router(config-register-pool)#

Step 2 sip-oauth Enables SIP OAuth on Pool.

Example:
Router(config-register-pool)#sip-oauth

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Command or Action Purpose

Step 3 end Returns to privileged EXEC mode.
Example:
Router(config-register-pool)#end

Verify SIP OAuth for Secure SRST

SUMMARY STEPS
1. show running-config all | sec listen-port
2. show sip-ua connections tcp tls detail
3. show sip status registrar
4. show voice register pool <index>
5. show voice register statistics
6. show voip sip-oauth key-server status

DETAILED STEPS

Step 1 show running-config all | sec listen-port

Show command output that displays information on the listen-port configuration in SIP OAuth.
Example:
Router#show running-config | section listen-port
listen-port secure no-client-validation 5090
Router#

Step 2 show sip-ua connections tcp tls detail

Show command to display the status, port details, and negotiated ciphers for SIP OAuth.
Note The Conn-Id suffixed with * are the client connections using SIP OAuth port.

Example:
Router#show sip-ua connections tcp tls detail
Total active connections : 4
No. of send failures : 0
No. of remote closures : 8
No. of conn. failures : 0
No. of inactive conn. ageouts : 0
TLS client handshake failures : 0
TLS server handshake failures : 0

---------Printing Detailed Connection Report---------

Note:
** Tuples with no matching socket entry
- Do 'clear sip <tcp[tls]/udp> conn t ipv4:<addr>:<port>'
to overcome this error condition
++ Tuples with mismatched address/port entry
- Do 'clear sip <tcp[tls]/udp> conn t ipv4:<addr>:<port> id <connid>'
to overcome this error condition
* Connections with SIP OAuth ports

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Remote-Agent:10.5.10.200, Connections-Count:0

Remote-Agent:10.5.10.201, Connections-Count:0

Remote-Agent:10.5.10.202, Connections-Count:0

Remote-Agent:10.5.10.212, Connections-Count:1
Remote-Port Conn-Id Conn-State WriteQ-Size Local-Address TLS-Version Cipher
Curve
=========== ======= =========== =========== ============= =========== ==============================
=====
52248 27 Established 0 - TLSv1.2 ECDHE-RSA-AES256-GCM-SHA384
P-256

Remote-Agent:10.5.10.213, Connections-Count:1
Remote-Port Conn-Id Conn-State WriteQ-Size Local-Address TLS-Version Cipher
Curve
=========== ======= =========== =========== ============= =========== ==============================
=====
50901 28* Established 0 - TLSv1.2 ECDHE-RSA-AES256-GCM-SHA384
P-256

Remote-Agent:10.5.10.209, Connections-Count:1
Remote-Port Conn-Id Conn-State WriteQ-Size Local-Address TLS-Version Cipher
Curve
=========== ======= =========== =========== ============= =========== ==============================
=====
51402 29* Established 0 - TLSv1.2 ECDHE-RSA-AES256-GCM-SHA384
P-256

Remote-Agent:10.5.10.204, Connections-Count:1
Remote-Port Conn-Id Conn-State WriteQ-Size Local-Address TLS-Version Cipher
Curve
=========== ======= =========== =========== ============= =========== ==============================
=====
50757 30* Established 0 - TLSv1.2 ECDHE-RSA-AES256-GCM-SHA384
P-256

Remote-Agent:10.5.10.218, Connections-Count:0

-------------- SIP Transport Layer Listen Sockets ---------------

Conn-Id Local-Address
=========== =============================
0 [0.0.0.0]:5061:
2 [0.0.0.0]:5090:

Step 3 show sip status registrar

Show command to display the registration status of a SIP client.
Note Transport parameter (TLS) suffixed with * are the endpoints registered using SIP OAuth port.

Example:
Router#show sip status registrar
Line destination expires(sec) contact
transport call-id
peer
=============================================================================================================
2999904 10.5.10.204 76 10.5.10.204

TLS* [email protected]
40004

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2999901 10.5.10.212 74 10.5.10.212

TLS [email protected]
40001

2999902 10.5.10.213 75 10.5.10.213

TLS* [email protected]
40002

2999905 10.5.10.209 76 10.5.10.209

TLS* [email protected]
40003
* TLS without client validation

Step 4 show voice register pool <index>

Show command to display whether sip-oauth is enabled in a Secure SIP SRST voice register pool.
Example:
Router#show voice register pool 1
Pool Tag 1
Config:
Proxy Ip address is 0.0.0.0
DTMF Relay is disabled
kpml signal is enabled
Lpcor Type is none

SIP OAuth is enabled

Reason for unregistered state: reboot

paging-dn: config 0 [multicast] effective 0 [multicast]

VRF:
NA

Dialpeers created:

Statistics:
Active registrations : 0

Total SIP phones registered: 0

Total Registration Statistics
Registration requests : 0
Registration success : 0
Registration failed : 0
unRegister requests : 0
unRegister success : 0
unRegister failed : 0
Auto-Register requests : 0
Attempts to register
after last unregister : 0
Last register request time :
Last unregister request time :
Register success time :
Unregister success time :

Step 5 show voice register statistics

Show command to display statistics and ouput for success and error registration flows.

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Example:
gw1-2a#show voice register statistics
Global statistics
Active registrations : 0

Total SIP phones registered: 0

Total Registration Statistics
Registration requests : 244
Registration success : 125
Registration failed : 119
unRegister requests : 121
unRegister success : 121
unRegister failed : 0
Auto-Register requests : 0
Attempts to register
after last unregister : 0
Last register request time : 22:04:30.574 clock Tue Dec 21 2021
Last unregister request time : 22:08:38.146 clock Tue Dec 21 2021
Register success time : 22:04:30.577 clock Tue Dec 21 2021
Unregister success time : 22:08:38.147 clock Tue Dec 21 2021

Register pool 29 statistics

Active registrations : 0

Total SIP phones registered: 0

Total Registration Statistics
Registration requests : 12
Registration success : 12
Registration failed : 0
unRegister requests : 12
unRegister success : 12
unRegister failed : 0
Auto-Register requests : 0
Attempts to register
after last unregister : 0
Last register request time : 13:07:53.523 clock Tue Dec 21 2021
Last unregister request time : 13:12:01.716 clock Tue Dec 21 2021
Register success time : 13:07:53.523 clock Tue Dec 21 2021
Unregister success time : 13:12:01.716 clock Tue Dec 21 2021

Reason for unregistered state:

No registration request since last reboot/unregister

Step 6 show voip sip-oauth key-server status

Show command to display key retrieval details for SIP OAuth.
Note The output is the same for IPv6 except that the key-server address is an IPv6 address.

Example:
Router#show voip sip-oauth key-server status
Key-server: 10.1.10.50
Last Request Time: 11:40:58.389 UTC Fri Nov 12 2021
Last Success response Time: 11:40:58.456 UTC Fri Nov 12 2021
Current Status: SUCCESS
Next Request Time: 11:40:58.389 UTC Sat Nov 13 2021
Total requests sent: 13
Total success responses: 3
Total failure responses: 10

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SHA2-Cipher-Only Mode for Unified Secure SRST

SHA2-Cipher-Only Mode for Unified Secure SRST

From SRST 14.2 onwards, the ciphers that are offered by Secure SIP SRST may be limited to only those that
meet your compliance requirements. Similarly, Secure SCCP SRST may be configured to only allow SHA2
TLS1.2 ciphers.
SCCP Client Registration
When SCCP SRST is configured with SHA2 ciphers, SCCP clients must use one of the following SHA2
cipher suites to establish a TLS connection:
• ECDHE_RSA_AES_256_GCM_SHA256
• ECDHE_RSA_AES_256_GCM_SHA384
• DHE_RSA_AES128_GCM_SHA256
• DHE_RSA_AES256_GCM_SHA384
• ECDHE_ECDSA_AES128_GCM_SHA256
• ECDHE_ECDSA_AES256_GCM_SHA384

Media packets are encrypted and sent using the AEAD_AES_256_GCM SRTP cipher suite.

Note When Secure SCCP SRST is configured to require SHA2 ciphers, only clients using SCCP protocol
version 23 or higher are allowed to register. If SHA2 is not configured as a requirement for Secure SCCP
SRST, then clients using SCCP protocol version 23 or lesser may be used.

SIP Client Registration

Secure SIP SRST may be configured to allow SIP clients to establish a TLS connection using single or multiple
preferred cipher suites.
For example:
Router(config)#voice class tls-cipher 333
Router(config-class)#cipher 1 ?
DHE_RSA_AES128_GCM_SHA256 supported in TLS 1.2 & above
DHE_RSA_AES256_GCM_SHA384 supported in TLS 1.2 & above
DHE_RSA_WITH_AES_128_CBC_SHA supported in TLS 1.0 & above
DHE_RSA_WITH_AES_256_CBC_SHA supported in TLS 1.0 & above
ECDHE_ECDSA_AES128_GCM_SHA256 supported in TLS 1.2 & above
ECDHE_ECDSA_AES256_GCM_SHA384 supported in TLS 1.2 & above
ECDHE_RSA_AES128_GCM_SHA256 supported in TLS 1.2 & above
ECDHE_RSA_AES256_GCM_SHA384 supported in TLS 1.2 & above
RSA_WITH_AES_128_CBC_SHA supported in TLS 1.0 & above
RSA_WITH_AES_256_CBC_SHA supported in TLS 1.0 & above

Router(config-class)#cipher 1 DHE_RSA_AES128_GCM_SHA256
Router(config-class)#end

Note Configure SRST TLS cipher policy before a SIP client is allowed to connect and register.

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Benefits

After the successful signalling, media packets are encrypted based on the srtp-crypto configuration. Configure
an SRTP cipher list first using the voice class srtp-crypto <tag> command. Associate the SRTP cipher list
with the voice register pool.
Router(config)#voice class srtp-crypto 22
GW1-2A(config-class)#?
VOICECLASS configuration commands:
crypto Configure preferred SRTP cipher-suite
exit Exit from voice class configuration mode
help Description of the interactive help system
no Negate a command or set its defaults

router(config-class)#crypto ?
<1-4> Set the preference order for the cipher-suite (1 = Highest)

Router(config-class)#crypto 1 ?
AEAD_AES_128_GCM Allow secure calls with SRTP AEAD_AES_128_GCM cipher-suite
AEAD_AES_256_GCM Allow secure calls with SRTP AEAD_AES_256_GCM cipher-suite
AES_CM_128_HMAC_SHA1_32 Allow secure calls with SRTP AES_CM_128_HMAC_SHA1_32 cipher-suite

AES_CM_128_HMAC_SHA1_80 Allow secure calls with SRTP AES_CM_128_HMAC_SHA1_80 cipher-suite

Router(config-class)#crypto 1 AEAD_AES_256_GCM
Router(config-class)#do show run | sec srtp-cry
voice class srtp-crypto 22
crypto 1 AEAD_AES_256_GCM

Router(config)# voice register pool 17

Router(config-register-pool)# id network 10.1.10.217 mask 255.255.255.255
Router(config-register-pool)# dtmf-relay rtp-nte
Router(config-register-pool)# codec g711ulaw

Show run output for pool:

=============================================
Router#show running-config | sec voice register pool 17
voice register pool 17
id network 10.1.10.217 mask 255.255.255.255
dtmf-relay rtp-nte
voice-class srtp-crypto 22
codec g711ulaw
Router#show run

When you configure srtp-crytpto 23, which is not present, you get the following error:
Router(config-register-pool)#voice-class srtp-crypto 23
ERROR: There is no voice-class srtp-crypto 23

When you configure srtp-crytpto 22, which is present, you get the following output:
Router(config-register-pool)#voice-class srtp-crypto 22
Router(config-register-pool)#

Note An SRTP crypto policy must be configured before it can be used in a voice register pool configuration.

Benefits
When you configure SHA2 cipher suite with TLS version 1.2, you get the following benefits:
• Improved security as SHA2 cipher suites provides more reliable security certificates.
• Fast computation.

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• Resistance to collision attacks.

Configure SHA2 Cipher Suite with TLS

SUMMARY STEPS
1. configure terminal
2. call-manager-fallback
3. transport-tcp-tls v1.2 sha2

DETAILED STEPS

Command or Action Purpose

Step 1 configure terminal Enters global configuration mode.
Example:
Router#configure terminal

Step 2 call-manager-fallback Enters config-cm-fallback mode.

Example:
Router(config)#call-manager-fallback
Router(config-cm-fallback)

Step 3 transport-tcp-tls v1.2 sha2 Configures the SHA2 cipher suite on the router.
Example:
Router(config-cm-fallback)#transport-tcp-tls v1.2
sha2
<cr> <cr>

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.

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

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.

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

The following figure shows the interworking of the credentials server on the SRST router, Cisco Unified
Communications Manager, and the Cisco Unified IP Phone. The following table describes the establishment
of secure SRST to the Cisco Unified IP Phone.
Figure 2: Interworking of Credentials Server on SRST Router, Cisco Unified Communications Manager, and Cisco Unified IP Phone

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Table 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 The CTL file contains the

retrieves a CTL file from the TFTP certificates that the phone should
server. trust.

The Cisco IP Phone opens a Cisco Unified Communications

Transport Layer Security (TLS) Manager exports secure Cisco
protocol channel and registers to Unified SRST router information
Cisco Unified Communications and the Cisco Unified SRST router
Manager. 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 Interworking of
Credentials Server on SRST
Router, Cisco
Unified Communications Manager,
and Cisco Unified IP Phone.

If the Cisco Unified IP Phone is The connection to the SRST router

configured as “authenticated” or happens automatically, assuming
“encrypted” and Cisco Unified there is not a secondary Cisco
Communications Manager is Unified Communications Manager
configured in mixed mode, the and Cisco Unified SRST is
phone looks for an SRST certificate configured as the backup device.
in its configuration file. If it finds See Figure Interworking of
an SRST certificate, it opens a Credentials Server on SRST
standby TLS connection to the Router, Cisco
default port. The default port is the Unified Communications Manager,
Cisco Unified IP Phone TCP port and Cisco Unified IP Phone.
plus 443; that is, port 2443 on a
Cisco Unified Communications
Cisco Unified SRST router.
Manager should be configured in
mixed mode, which is its secure
mode.

In case of WAN failure, the Cisco Unified IP Phone starts Cisco Unified SRST registration.

SRST The Cisco Unified IP Phone —

registers with the SRST router at
the default port for secure
communications.

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Secure SRST Authentication and Encryption

Secure SRST Authentication and Encryption

The following figure illustrates the process of secure SRST authentication and encryption, and the following
table describes the process.
Figure 3: Secure Cisco Unified SRST Authentication and Encryption

Process Description or Detail

Steps

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.

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.
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.

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Process Description or Detail

Steps

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.

How to Configure Secure Unified SRST

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.
This section contains the following procedures:

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

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

Command or Action Purpose

Step 1 crypto pki server cs-label Enables the certificate server and enters certificate server
configuration mode.
Example:
Router (config)# crypto pki server srstcaserver Note If you manually generated an RSA key pair, the
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:
Router (cs-server)# database level complete • minimal: Enough information is stored only to
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
Example:
certificate server using the crypto pki server command,
Router (cs-server)# database url nvram use 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:
Router (cs-server)# issuer-name CN=srstcaserver issuer-name CN= cs-label .

Step 5 grant auto Allows an automatic certificate to be issued to any requestor.

Example:

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Command or Action Purpose

Router (cs-server)# grant auto • This command is used only during enrollment and will
be removed in the Disabling Automatic Certificate
Enrollment section.

Step 6 no shutdown Enables the Cisco IOS certificate server.

Example: • You should issue this command only after you have
Router (cs-server)# no shutdown completely configured your certificate server.

Example
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 .

SUMMARY STEPS
1. crypto pki trustpointname
2. rsakeypair keypair-label
3. enrollment url url
4. revocation-check method1
5. exit
6. crypto pki authenticate name
7. crypto pki enroll name

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DETAILED STEPS

Command or Action Purpose

Step 1 crypto pki trustpointname Declares the CA that your router should use and enters
ca-trustpoint configuration mode.
Example:
Router(config)# crypto pki trustpoint srstca • The name provided will be the same as the trustpoint
name that will be declared in the Enabling Credentials
Service on the Secure Cisco Unified SRST Router
section.

Step 2 rsakeypair keypair-label To specify a named Rivest, Shamir, and Adelman (RSA)
key pair for this trustpoint, use the rsakeypair command
Example:
in trustpoint configuration mode.
Router(config-trustp)# rsakeypair srstcakey
2048 • Configure the RSA key length to 2048 bits or above.

Step 3 enrollment url url Specifies the enrollment parameters of your CA.
Example: • url url: Specifies the URL of the CA to which your
Router(ca-trustpoint)# enrollment url router should send certificate requests.
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, 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
Example:
revocation status of the certificate. For this task, the only
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

Step 6 crypto pki authenticate name Authenticates the CA (by getting the certificate from the
CA).
Example:
Router(config)# crypto pki authenticate srstca • Takes the name of the CA as the argument.

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Command or Action Purpose

Step 7 crypto pki enroll name Obtains the SRST router certificate from the CA.
Example: • Takes the name of the CA as the argument.
Router(config)# crypto pki enroll srstca

Example
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(ca-trustpoint)# revocation-check none
Router(ca-trustpoint)# exit
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

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 section.

Note You should disable the grant auto command so that certificates cannot be continually granted.

SUMMARY STEPS
1. crypto pki servercs-label

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

DETAILED STEPS

Command or Action Purpose

Step 1 crypto pki servercs-label Enables the certificate server and enters certificate server
configuration mode.
Example:
Router (config)# crypto pki server srstcaserver Note If you manually generated an RSA key pair, the
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.

Example: • You should issue this command only after you have
Router (cs-server)# no shutdown completely configured your certificate server.

What to do next
For manual enrollment instructions, see the Manual Certificate Enrollment (TFTP and Cut-and-Paste) feature.

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

DETAILED STEPS

Command or Action Purpose

Step 1 show running-config Use the show running-config command to verify the
creation of the CA server (01) and device (02) certificates.
Example:
This example shows the enrolled certificates.
Router# show running-config
.
.

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Command or Action Purpose

.
! 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

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Command or Action Purpose

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 the Control Plane Policing: Example section.

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

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Troubleshooting Credential Settings

DETAILED STEPS

Command or Action Purpose

Step 1 credentials Provides the Cisco Unified SRST Router certificate to
Cisco Unified Communications Manager and enters
Example:
credentials configuration mode.
Router(config)# credentials

Step 2 ip source-address ip-address [portport] Enables the Cisco Unified SRST Router to receive messages
from Cisco Unified Communications Manager through the
Example:
specified IP address and port.
Router(config-credentials)# ip source-address
10.1.1.22 port 2445 • ip-address: The IP address is 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 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
Example:
trustpoint-name argument is the name of the trustpoint and
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
section.

Step 4 exit Exits credentials configuration mode.

Example:
Router(config-credentials)# exit

Example
Router(config)# credentials
Router(config-credentials)# ip source-address 10.1.1.22 port 2445
Router(config-credentials)# trustpoint srstca
Router(config-credentials)# exit

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

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Importing Phone Certificate Files in PEM Format to the Secure SRST Router

DETAILED STEPS

Command or Action Purpose

Step 1 show credentials Use the show credentials command to display the
credential settings on the Cisco Unified SRST Router that
Example:
are supplied to Cisco Unified Communications Manager
Router# show credentials 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
Sep 29 01:01:51.903: Credentials service: TLS
Handshake returns OPSSLReadWouldBlockErr
Sep 29 01:01:52.907: Credentials service: TLS
Handshake returns OPSSLReadWouldBlockErr
Sep 29 01:01:53.927: Credentials service: TLS
Handshake completes.

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.

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. The Supported
Cisco Unified IP Phones and Certificates table 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.

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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 the Supported Cisco
Unified IP Phones and Certificates table, which must be copied and pasted to Cisco Unified SRST Routers.

Note CiscoRootCA is also called CiscoRoot2048CA.

For Cisco Unified CM 5.0 and later versions, perform the following steps:

Before you begin

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

SUMMARY STEPS
1. Login to Cisco Unified Communications Manager.
2. Go to Security > Certificate Management > Download Certificate/CTL.
3. Select Download Trust Cert and click Next.
4. Select CAPF-trust and click Next.
5. Select CiscoCA and click Next.
6. Click Continue.
7. Click the file name.
8. Copy all the contents that appear between “-----BEGIN CERTIFICATE-----” and “-----END
CERTIFICATE-----” to a location where you can retrieve it later.
9. Repeat Steps 5 to 8 for CiscoManufactureCA, CiscoRootCA2048, and CAPF.

DETAILED STEPS

Command or Action Purpose

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.
Step 8 Copy all the contents that appear between “-----BEGIN
CERTIFICATE-----” and “-----END CERTIFICATE-----”
to a location where you can retrieve it later.

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Cisco Unified Communications Manager 6.0 and Later Versions

Command or Action Purpose

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

DETAILED STEPS

Command or Action Purpose

Step 1 crypto pki trustpoint name Declares the CA that your router should use and enters
ca-trustpoint configuration mode.
Example:
Router (config)# crypto pki trustpoint CAPF • name: Enter the name of each certificate individually
(for example, CAPF, CiscoCA, CiscoManufactureCA,
and CiscoRootCA2048).

Step 2 revocation-check none Checks the revocation status of a certificate using the
selected method.
Example:
Router(ca-trustpoint)# revocation-check none • Using the none keyword is mandatory for this task.
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

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Command or Action Purpose

Step 4 exit Exits ca-trustpoint configuration mode and returns to global
configuration.
Example:
Router(ca-trustpoint)# exit

Step 5 crypto pki authenticate name Authenticates the CA (by getting the certificate from the
CA).
Example:
Router(config)# crypto pki authenticate CAPF • Enter the same name argument used in the crypto pki
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

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(ca-trustpoint)# revocation-check none
Router(ca-trustpoint)# enrollment terminal
Router(ca-trustpoint)# exit
Router(config)# crypto pki authenticate 7970
Enter the base 64 encoded CA certificate.
End with a blank line or the word "quit" on a line by itself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quit
Certificate has the following attributes:
Fingerprint MD5: F7E150EA 5E6E3AC5 615FC696 66415C9F
Fingerprint SHA1: 1BE2B503 DC72EE28 0C0F6B18 798236D8 D3B18BE6
% Do you accept this certificate? [yes/no]: y
Trustpoint CA certificate accepted.
% Certificate successfully imported

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

Router(config)# crypto pki trustpoint 7960

Router(ca-trustpoint)# revocation-check none
Router(ca-trustpoint)# enrollment terminal
Router(ca-trustpoint)# exit
Router(config)# crypto pki authenticate 7960
Enter the base 64 encoded CA certificate.
End with a blank line or the word "quit" on a line by itself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quit
Certificate has the following attributes:
Fingerprint MD5: 4B9636DF 0F3BA6B7 5F54BE72 24762DBC
Fingerprint SHA1: A9917775 F86BB37A 5C130ED2 3E528BB8 286E8C2D
% Do you accept this certificate? [yes/no]: y
Trustpoint CA certificate accepted.
% Certificate successfully imported
Router(config)# crypto pki trustpoint PEM
Router(ca-trustpoint)# revocation-check none
Router(ca-trustpoint)# enrollment terminal
Router(ca-trustpoint)# exit
Router(config)# crypto pki authenticate PEM
Enter the base 64 encoded CA certificate.
End with a blank line or the word "quit" on a line by itself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quit
Certificate has the following attributes:
Fingerprint MD5: 233C8E33 8632EA4E 76D79FEB FFB061C6
Fingerprint SHA1: F7B40B94 5831D2AB 447AB8F2 25990732 227631BE
% Do you accept this certificate? [yes/no]: y
Trustpoint CA certificate accepted.
% Certificate successfully imported
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

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Cisco Unified Communications Manager 5.0 and Later Versions Example

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:
Issuing CA certificate configured:
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:
Keys generated ............. Yes (General Purpose)
Issuing CA authenticated ....... Yes
Certificate request(s) ..... None
Trustpoint PEM:
Issuing CA certificate configured:
Subject Name:
cn=CAP-RTP-001,o=Cisco Systems
Fingerprint MD5: 233C8E33 8632EA4E 76D79FEB FFB061C6
Fingerprint SHA1: F7B40B94 5831D2AB 447AB8F2 25990732 227631BE
State:
Keys generated ............. Yes (General Purpose)
Issuing CA authenticated ....... Yes
Certificate request(s) ..... None
Trustpoint srstcaserver:
Issuing CA certificate configured:
Subject Name:
cn=srstcaserver
Fingerprint MD5: 6AF5B084 79C93F2B 76CC8FE6 8781AF5E
Fingerprint SHA1: 47D30503 38FF1524 711448B4 9763FAF6 3A8E7DCF
State:
Keys generated ............. Yes (General Purpose)
Issuing CA authenticated ....... Yes
Certificate request(s) ..... None
Trustpoint srstca:
Issuing CA certificate configured:
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:
Keys generated ............. Yes (General Purpose)
Issuing CA authenticated ....... Yes
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(ca-trustpoint)# revocation-check none
Router(ca-trustpoint)# enrollment terminal

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Router(ca-trustpoint)# exit
Router(config)# crypto pki authenticate CAPF

Enter the base 64 encoded CA certificate.

End with a blank line or the word "quit" on a line by itself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Certificate has the following attributes:
Fingerprint MD5: 1951DJ4E 76D79FEB FFB061C6 233C8E33
Fingerprint SHA1: 222891BE Z7B89B94 447AB8F2 5831D2AB 25990732
% Do you accept this certificate? [yes/no]: y
Trustpoint CA certificate accepted.
% Certificate successfully imported

Router(config)# crypto pki trustpoint CiscoCA

Router(ca-trustpoint)# revocation-check none
Router(ca-trustpoint)# enrollment terminal
Router(ca-trustpoint)# exit
Router(config)# crypto pki authenticate CiscoCA

Enter the base 64 encoded CA certificate.

End with a blank line or the word "quit" on a line by itself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quit
Certificate has the following attributes:
Fingerprint MD5: 21956CBR 4B9706DF 0F3BA6B7 7P54AZ72
Fingerprint SHA1: A9917775 F86BB37A 7H130ED2 3E528BB8 286E8C2D
% Do you accept this certificate? [yes/no]: y
Trustpoint CA certificate accepted.
% Certificate successfully imported

Router(config)# crypto pki trustpoint CiscoManufactureCA

Router(ca-trustpoint)# revocation-check none
Router(ca-trustpoint)# enrollment terminal
Router(ca-trustpoint)# exit
Router(config)# crypto pki authenticate CiscoManufactureCA

Enter the base 64 encoded CA certificate.

End with a blank line or the word "quit" on a line by itself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I+ii6itvaSN6go4cTAnPpE+rhC836WVg0ZrG2PML9d7QJwBcbx2RvdFOWFEdyeP3
OOfTC9Fovo4ipUsG4eakqjN9GnW6JvNwxmEApcN5JlunGdGTjaubEBEpH6GC/f08
S25l3JNFBemvM2tnIwcGhiLa69yHz1khQhrpz3B1iOAkPV19TpY4gJfVb/Cbcdi6
YBmlsGGGrd1lZva5J6LuL2GbuqEwYf2+rDUU+bgtlwavw+9tzD0865XpgdOKXrbO
+nmka9eiV2TEP0zJ2+iC7AFm1BCIolblPFft6QKoSJFjB6thJksaE5/k3Npf
quit
Certificate has the following attributes:
Fingerprint MD5: 0F3BA6B7 4B9636DF 5F54BE72 24762SBR
Fingerprint SHA1: L92BB37A S9919925 5C130ED2 3E528UP8 286E8C2D
% Do you accept this certificate? [yes/no]: y
Trustpoint CA certificate accepted.
% Certificate successfully imported

Router(config)# crypto pki trustpoint CiscoRootCA2048

Router(ca-trustpoint)# revocation-check none
Router(ca-trustpoint)# enrollment terminal
Router(ca-trustpoint)# exit
Router(config)# crypto pki authenticate CiscoRootCA2048

Enter the base 64 encoded CA certificate.

End with a blank line or the word "quit" on a line by itself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quit
Certificate has the following attributes:
Fingerprint MD5: 2G3LZ6B7 2R1995ER 6KE4WE72 3E528BB8
Fingerprint SHA1: M9912245 5C130ED2 24762JBC 3E528VF8 956E8S5H
% Do you accept this certificate? [yes/no]: y
Trustpoint CA certificate accepted.
% Certificate successfully imported

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

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.
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.
1. In the menu bar in Cisco Unified Communications Manager, choose CCMAdmin > System > SRST .
2. Click Add New SRST Reference .

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3. Enter the appropriate settings. The following figure 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

4. To add the new SRST reference, click Insert . The message “Status: Insert completed” displays.
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
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

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SUMMARY STEPS
1. In the menu bar in Cisco Unified Communications Manager, choose CCMAdmin > System > Device
Pool .
2. Use one of the following methods to add a device pool:
3. In the upper, right corner of the window, click the Add New Device Pool link. The Device Pool.
Configuration window displays.
4. Enter the SRST reference.
5. Click Update to save the device pool information in the database.

DETAILED STEPS

Command or Action Purpose

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.

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Command or Action Purpose

Step 3 In the upper, right corner of the window, click the Add
New Device Pool link. The Device Pool. Configuration
window displays.

Step 4 Enter the SRST reference.

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

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.

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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
1. call-manager-fallback
2. secondary-dialtone digit-string
3. transfer-system {blind | full-blind |full-consult | local-consult}
4. ip source-address ip-address [portport]
5. max-ephones max-phones
6. max-dn max-directory-numbers
7. transfer-pattern transfer-pattern
8. exit

DETAILED STEPS

Command or Action Purpose

Step 1 call-manager-fallback Enters call-manager-fallback configuration mode.
Example:
Router(config)# call-manager-fallback

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 |full-consult | Defines the call-transfer method for all lines served by the
local-consult} Cisco Unified SRST Router.
Example: • blind : Calls are transferred without consultation with
Router(config-cm-fallback)# transfer-system a single phone line using the Cisco proprietary 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.

Step 4 ip source-address ip-address [portport] Enables the router to receive messages from the Cisco IP
Phones through the specified IP addresses and provides for
Example:

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Command or Action Purpose

Router(config-cm-fallback)# ip source-address strict IP address verification. The default port number is
10.1.1.22 port 2000
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
Example:
platform dependent. The default is 0. See the Platform and
Router(config-cm-fallback)# max-ephones 15 Memory Support section 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.
Example:
Router(config-cm-fallback)# max-dn 30 • max-directory-numbers : Maximum number of
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 for further details.

Step 7 transfer-pattern transfer-pattern Allows transfer of phone calls by Cisco Unified IP Phones
to specified phone number patterns.
Example:
Router(config-cm-fallback)# transfer-pattern • transfer-pattern: String of digits for permitted call
..... transfers. Wildcards are allowed.

Step 8 exit Exits call-manager-fallback configuration mode.

Example:
Router(config-cm-fallback)# exit

Example
The following example enables SRST mode on your router:
Router(config)# call-manager-fallback
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-ephones 15
Router(config-cm-fallback)# max-dn 30
Router(config-cm-fallback)# transfer-pattern .....
Router(config-cm-fallback)# exit

Configuring Secure SCCP SRST

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. 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.

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Restrictions for Configuring Secure SCCP SRST

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.
• 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.
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.

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

• Adding an SRST Reference to Cisco Unified Communications Manager
• Enabling SRST Mode on the Secure Cisco Unified SRST Router

SUMMARY STEPS
1. show ephone
2. show ephone offhook

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3. show voice call status

4. debug ephone register
5. debug ephone state

DETAILED STEPS

Command or Action Purpose

Step 1 show ephone Use this command to display registered Cisco Unified IP
Phones and their capabilities. The show ephone command
Example:
also displays authentication and encryption status when
Router# show ephone used for secure SCCP SRST. In this example, authentication
ephone-1 Mac:1000.1111.0002 TCP socket:[5] and encryption status 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
mediaActive:0 offhook:0 ringing:0 reset:0
reset_sent:0 paging 0 debug:0
IP:10.1.1.40 32718 7970 keepalive 390 max_line 8
button 1: dn 21 number 2011 CM Fallback CH1 IDLE
ephone-3 Mac:1000.1111.000A TCP socket:[16]
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 32862 7970 keepalive 390 max_line 8
button 1: dn 2 number 2010 CM Fallback CH1 IDLE

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,
Example:
authentication and encryption status is active with a TLS
Router# show ephone offhook 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
mediaActive:1 offhook:1 ringing:0 reset:0
reset_sent:0 paging 0
:0
IP:10.1.1.40 32626 7970 keepalive 391 max_line 8
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]
activeLine:1 REGISTERED in SCCP ver
5 + Authentication + Encryption with TLS connection
mediaActive:1 offhook:1 ringing:0 reset:0
reset_sent:0 paging 0 debug:0

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Command or Action Purpose

IP:10.1.1.40 32718 7970 keepalive 391 max_line 8
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

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
Example:
applicable for calls between two POTS dial peers.
CallID CID ccVdb Port DSP/Ch Called # Codec
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

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Command or Action Purpose

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
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 09:16:03.188: TLS Handshake retcode
OPSSLReadWouldBlockErr
*Jun 29 09:16:04.188: TLS Handshake retcode
OPSSLReadWouldBlockErr
*Jun 29 09:16:05.188: TLS Handshake retcode
OPSSLReadWouldBlockErr
*Jun 29 09:16:06.188: TLS Handshake retcode
OPSSLReadWouldBlockErr
*Jun 29 09:16:07.188: TLS Handshake retcode
OPSSLReadWouldBlockErr
*Jun 29 09:16:08.188: CRYPTO_PKI_OPSSL - Verifying
1

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Command or Action Purpose

Certs
*Jun 29 09:16:08.212: TLS Handshake completes

Step 5 debug ephone state Use this command to review call setup between two secure
Cisco Unified IP Phones. The debug ephone state trace
Example:
shows the generation and distribution of encryption and
Router# debug ephone state decryption keys between the two phones.
*Jan 11 18:33:09.231:%SYS-5-CONFIG_I:Configured
from
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
*Jan 11 18:33:15.031:dn_tone_control DN=2 chan 1
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
*Jan 11 18:33:16.039:ephone-2[2]:SetCallState line
1
DN 2(-1) chan 1 ref 6 TsProceed
*Jan 11 18:33:16.039:ephone-2[2]:SetCallState line
1
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
*Jan 11 18:33:16.039:ephone-2[2]:Call Info DN 2
line
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
*Jan 11 18:33:16.047:ephone-3[3]:SetCallState line
1
DN 4(4) chan 1 ref 7 TsRingIn
*Jan 11 18:33:16.047:ephone-3[3]::callingNumber
6000
*Jan 11 18:33:16.047:ephone-3[3]::callingParty 6000
*Jan 11 18:33:16.047:ephone-3[3]:Call Info DN 4
line
1 ref 7 call state 7 called 6001 calling 6000
origcalled
*Jan 11 18:33:16.047:ephone-3[3]:Call Info DN 4

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Command or Action Purpose

line
1 ref 7 called 6001 calling 6000 origcalled 6001
calltype 1
*Jan 11 18:33:16.047:ephone-3[3]:Call Info for chan
1
*Jan 11 18:33:16.047:ephone-3[3]:Original Called
Name 6001
*Jan 11 18:33:16.047:ephone-3[3]:6000 calling
*Jan 11 18:33:16.047:ephone-3[3]:6001
*Jan 11 18:33:16.047:ephone-3[3]:Ringer Inside Ring
On
*Jan 11 18:33:16.051:dn_tone_control DN=2 chan 1
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
*Jan 11 18:33:20.831:ephone-3[3]:SetCallState line
1
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
*Jan 11 18:33:20.831:ephone-3[3]:SetCallState line
1
DN 4(-1) chan 1 ref 7 TsConnected
*Jan 11 18:33:20.831:defer_start for DN 2 chan 1
at
CONNECTED
*Jan 11 18:33:20.831:ephone-2[2]:SetCallState line
1
DN 2(-1) chan 1 ref 6 TsConnected
*Jan 11 18:33:20.835:ephone-3[3]::callingNumber
6000
*Jan 11 18:33:20.835:ephone-3[3]::callingParty 6000
*Jan 11 18:33:20.835:ephone-3[3]:Call Info DN 4
line
1 ref 7 call state 4 called 6001 calling 6000
origcalled
*Jan 11 18:33:20.835:ephone-3[3]:Call Info DN 4
line
1 ref 7 called 6001 calling 6000 origcalled 6001
calltype 1
*Jan 11 18:33:20.835:ephone-3[3]:Call Info for chan
1
*Jan 11 18:33:20.835:ephone-3[3]:Original Called
Name 6001
*Jan 11 18:33:20.835:ephone-3[3]:6000 calling
*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

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Configuration Examples for Secure SCCP SRST

Command or Action Purpose

!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.
*Jan 11 18:33:21.087:dn_tone_control DN=2 chan 1
tonetype=0:DtSilence onoff=0 pid=232
*Jan 11 18:33:21.087:DN 4 chan 1 Voice_Mode
*Jan 11 18:33:21.091:DN 2 chan 1 End Voice_Mode
*Jan 11 18:33:21.091:DN 2 chan 1 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
*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
*Jan 11 18:33:21.851:ephone-2[2]::callingParty 6000
*Jan 11 18:33:21.851:ephone-2[2]:Call Info DN 2
line
1 ref 6 call state 4 called 6001 calling 6000
origcalled
*Jan 11 18:33:21.851:ephone-2[2]:Call Info DN 2
line
1 ref 6 called 6001 calling 6000 origcalled 6001
calltype 2
*Jan 11 18:33:21.851:ephone-2[2]:Call Info for chan
1
*Jan 11 18:33:21.851:ephone-2[2]:Original Called
Name 6001
*Jan 11 18:33:21.851:ephone-2[2]:6000 calling
*Jan 11 18:33:21.851:ephone-2[2]:6001

Configuration Examples for Secure SCCP SRST

This section provides the following configuration examples:

Note IP addresses and hostnames in examples are fictitious.

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 Configuring Secure SRST for SCCP and SIP
Secure SCCP SRST: Example

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.
Router# show running-config
.
.
.
! 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
!
! 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
revocation-check none
rsakeypair srstcaserver
!
! Define CTL/7970 trustpoint.
crypto pki trustpoint 7970
enrollment terminal
revocation-check none
!
crypto pki trustpoint PEM
enrollment terminal
revocation-check none
!
! Define CAPF/7960 trustpoint.
crypto pki trustpoint 7960
enrollment terminal
revocation-check none
!
! 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

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Secure SCCP SRST: Example

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
DEDBAAD7 3780136E B112A6
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
DEDBAAD7 3780136E B112A6
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

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Secure SCCP SRST: Example

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
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.

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Secure SCCP SRST: Example

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
match address 116
!
!
interface FastEthernet0/0
ip address 10.1.1.22 255.255.255.0
duplex auto
speed auto
crypto map rtp
!
interface FastEthernet0/1
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

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Control Plane Policing: Example

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
!
mgcp profile default
!
!
dial-peer voice 81235 pots
application mgcpapp
destination-pattern 81235
port 1/1/0
forward-digits all
!
dial-peer voice 81234 pots
application mgcpapp
destination-pattern 81234
port 1/0/0
!
dial-peer voice 999100 pots
application mgcpapp
port 1/0/0
!
dial-peer voice 999110 pots
application mgcpapp
port 1/1/0
!
!
! Enable credentials service on the gateway.
credentials
ip source-address 10.1.1.22 port 2445
trustpoint srstca
!
!
! Enable SRST mode.
call-manager-fallback
transport-tcp-tls
secondary-dialtone 9
transfer-system full-consult
ip source-address 10.1.1.22 port 2000
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.
Router# show running-config
.
.
.

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Configuring Secure SIP Call Signaling and SRTP Media with Cisco SRST

! 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
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
• 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 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.

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

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Configuring Cisco Unified Communications Manager

• 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.

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
2. configure terminal
3. voice service voip
4. srtp
5. allow-connections sip to h323
6. allow-connections sip to sip

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Configuring SIP options for Secure SIP SRST

7. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
Example: • Enter your password if prompted.
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 voice service voip Enters voice service configuration mode.

Example:
Router(config)# voice service voip

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:
Router(config-voi-serv)# allow-connections sip to
h323

Step 6 allow-connections sip to sip Allows connections from SIP endpoints to SIP endpoints.
Example:
Router(config-voi-serv)# allow-connections sip to
sip

Step 7 end Ends the current configuration session and returns to

privileged EXEC mode.
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
2. configure terminal
3. voice service voip
4. sip
5. url sip | sips
6. srtp negotiate cisco

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Configuring SIP SRST Security Policy

7. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
Example: • Enter your password if prompted.
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 voice service voip Enters voice service configuration mode.

Example:
Router(config)# voice service voip

Step 4 sip Enters SIP configuration mode.

Example:
Router(config-voi-serv)# sip

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:
Router(conf-serv-sip)# url sips To configure device-default mode, use the sip keyword to
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

Step 7 end Ends the current configuration session and returns to

privileged EXEC mode.
Example:
Router(conf-serv-sip)# end

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
1. voice register global
2. security-policy secure
3. end

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Configuring SIP User Agent for Secure SIP SRST

DETAILED STEPS

Command or Action Purpose

Step 1 voice register global Enters voice register global configuration mode.
Example:
Router(config)# voice register global

Step 2 security-policy secure Configures SIP registration security policy so that only
SIP/TLS/TCP connections are allowed. For device-default
Example:
mode, use the no security-policy command. Device-default
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.

Step 3 end Ends the current configuration session and returns to

privileged EXEC mode.
Example:
Router(config-register-global)# end

Configuring SIP User Agent for Secure SIP SRST

This section explains how the strict-cipher limits the allowed encryption algorithms.
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.
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 8.41.20.20 255.255.0.0trustpoint srst-trunk1
under sip-ua config mode.

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

DETAILED STEPS

Command or Action Purpose

Step 1 sip-ua Enters SIP user-agent configuration mode.
Example:

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Configuring SIP User Agent for Secure SIP SRST

Command or Action Purpose

Router(config)# sip-ua

Step 2 registrar ipv4: destination-address expires seconds Enables the gateway to register E.164 telephone numbers
with primary and secondary external SIP registrars.
Example:
destination-address is the IP address of the primary SIP
Router(config-sip-ua)# registrar registrar server.
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[strict-cipher] identifies the trustpoint string keyword and argument used
during the TLS handshake. The trustpointstring keyword
Example:
and argument refer to the gateway’s certificate generated
Router(config-sip-ua)# crypto signaling default as part of the enrollment process, using Cisco IOS
trustpoint 3745-SRST strict-cipher
public-key infrastructure (PKI) commands. The
strict-cipher keyword 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 |subnet mask The trustpoint label refers to the CUBE’s certificate that is
}trustpoint trustpoint-name generated with the Cisco IOS PKI commands as part of the
enrollment process.
Example:
Router(config-sip-ua)# crypto signaling Keywords and arguments are as follows:
remote-addr 8.41.20.20 255.255.0.0 trustpoint
srst-trunk1 • remote-addr ip address—Associates an IP address to
a trustpoint.
• trustpoint trustpoint-name—Refers to the SIP
gateways certificate generated as part of the enrollment
process using Cisco IOS PKI commands

Step 6 end Ends the current configuration session and returns to

privileged EXEC mode.
Example:
Router(config-sip-ua)# end

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

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Verifying the Configuration

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.
Router# show sip-ua status registrar
Line destination expires(sec) contact
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.

Router# show voice register global

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
network-locale[2] US
network-locale[3] US
network-locale[4] US
user-locale[0] US (This is the default user locale for this box)
user-locale[1] US
user-locale[2] US
user-locale[3] US
user-locale[4] US
Router#

Configuration Example for Cisco Unified SIP SRST

Current configuration : 15343 bytes
!
! Last configuration change at 05:34:06 UTC Tue Jun 13 2017

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Configuration Example for Cisco Unified SIP SRST

! NVRAM config last updated at 11:57:03 UTC Thu Jun 8 2017

!
version 16.7
service timestamps debug datetime msec
service timestamps log datetime msec
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
enrollment url http://10.0.0.1:80
serial-number
revocation-check none
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
revocation-check none
!
crypto pki trustpoint CAPF-3a66269a
enrollment terminal
revocation-check none
!
crypto pki trustpoint Cisco_Root_CA_2048
enrollment terminal
revocation-check none
!
!
crypto pki certificate chain TRUSTPT-SRST-CA-2

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Configuration Example for Cisco Unified SIP SRST

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
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

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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
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

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Configuration Example for Cisco Unified SIP SRST

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
allow-connections sip to sip
srtp
no supplementary-service sip refer
supplementary-service media-renegotiate
no supplementary-service sip handle-replaces
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
registrar server expires max 120 min 60
!
voice register global
default mode
no allow-hash-in-dn
security-policy secure
max-dn 50
max-pool 40
!
voice register pool 1
id network 10.0.0.1 mask 255.255.0.0
dtmf-relay rtp-nte
codec g711ulaw
!
voice hunt-group 1 sequential
final 89898

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Configuration Example for Cisco Unified SIP SRST

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
!
interface GigabitEthernet0/0/0
ip address 10.0.0.1 255.255.0.0
negotiation auto
!
interface GigabitEthernet0/0/1
no ip address
negotiation auto
!
interface GigabitEthernet0/0/2
ip address 10.0.0.1 255.0.0.0
negotiation auto
!
interface GigabitEthernet0/0/3
no ip address
negotiation auto
!
interface Service-Engine0/1/0
shutdown
!
interface Service-Engine0/2/0
shutdown
!
interface Service-Engine0/3/0
!
interface Service-Engine1/0/0
!
interface GigabitEthernet0
vrf forwarding Mgmt-intf
no ip address
negotiation auto
!
ip forward-protocol nd
ip http server
no ip http secure-server
ip route 0.0.0.0 0.0.0.0 10.0.0.1
!

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Configuration Example for SIP OAuth

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
!
mgcp profile default
!
sip-ua
crypto signaling default trustpoint TRUSTPT-SRST-CA-2
!
!
credentials
ip source-address 10.0.0.1 port 2445
trustpoint TRUSTPT-SRST-CA-2
!
!
call-manager-fallback
max-conferences 8 gain -6
transfer-system full-consult
max-ephones 50
max-dn 50
call-park system application
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
!
end

Configuration Example for SIP OAuth

The following is a configuration example to enable SIP OAuth in Secure SRST.

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Configuration Example for SIP OAuth

Router(config)#voice register pool 1

Router(config-register-pool)#?
voice register pool configuration commands:
after-hour After-hours call blocking
alias Associate alias pattern
application Define application
ata-ivr-pwd Define ATA IVR Password using 0-9 digit
busy-trigger-per-button Define the number of calls that triggers call
forward busy per line on the sip phone
call-forward Define E.164 telephone number for call forward
codec select the preferred codec to be used for SIP phone
conference Adhoc hardware conference configuration
conference-pattern Customized conference-pattern configuration
cor Class of Restriction on dial-peer for this dn
default Set a command to its defaults
dialplan-pattern Define E.164 telephone number prefix
digit Enable digit collect command
dtmf-relay Transport DTMF digits across IP link
emergency Emergency Assistance
exclude Exclude Local Services
exit Exit from voice register pool configuration mode
feature-button Define programmable line key
id define phone or device id
incoming Incoming called number
logout-profile enable extension mobility
lpcor Voice registry pool lpcor setup
max voice register pool max commands
media Media mode setting for SIP extension
night-service Define night-service bell
no Negate a command or set its defaults
number Define E.164 telephone number
overlap-signal Configure Overlap Signaling support
paging-dn set audio paging dn group for phone (use ephone
paging-dn number)
phone-mode Phone mode configuration in voice register pool
pin Define 4-8 digit personal identification number
preference Configure the preference for the voip dial-peers to
be created
presence enable call list feature
provision-tag define phone provision_tag
proxy Define SIP proxy for this pool
registration-timer keepalive registration expires timer
session-server define controlling session-server
sip-oauth Enable sip-oauth on Pool
tone Generate tones
transfer Transfer related configuration
transfer-pattern Customized transfer-pattern configuration
translate-outgoing Translation rule
translation-profile Translation profile
vad Enable vad on dial-peer and phone
voice-class Set voice class parameters

Router(config-register-pool)#sip_oauth ?
<cr> <cr>
Router(config-register-pool)#sip_oauth
Router(config-register-pool)#end

The following is a configuration example to disable SIP OAuth in Secure SRST.

Router(config)#voice register pool 1
Router(config-register-pool)#no sip-oauth
Router(config-register-pool)#end

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Configuration Examples for SHA-2 Cipher Suites

Configuration Examples for SHA-2 Cipher Suites

The following is a configuration example to enable SHA-2 cipher suite in SRST.
Router(config)#call-manager-fallback
Router(config-cm-fallback)#transport-tcp-tls ?
v1.0 Enable TLS Version 1.0
v1.1 Enable TLS Version 1.1
v1.2 Enable TLS Version 1.2

Router(config-cm-fallback)#transport-tcp-tls v1.2 ?
sha2 Allow SHA2 ciphers only

Router(config-cm-fallback)#transport-tcp-tls v1.2 sha2

Additional References
The following sections provide references related to this feature.

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 for Release 8.0(2) Documentation Guide 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, Memory, and Voice Products

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 (SRST) Reference Telephony (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.

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 Configuring Secure SRST for SCCP and SIP
MIBs

MIBs
MIB MIBs Link

No new or modified MIBs are supported by this To locate and download MIBs for selected platforms,
feature, and support for existing MIBs has not Cisco IOS releases, and feature sets, use Cisco MIB
been modified by this feature. Locator found at the 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.

Technical Assistance
Description Link

The Cisco Support website provides extensive online resources, including http://www.cisco.com/techsupport
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 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

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 Configuring Secure SRST for SCCP and SIP
Feature Information for Secure SCCP and SIP SRST

Feature Information for Secure SCCP and SIP SRST

The Feature Information for Secure SIP Call Signaling and SRTP Media with Cisco SRST table 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 The Feature Information for Secure SIP Call Signaling and SRTP Media with Cisco SRST 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.

Feature Name Releases Feature Information

Secure SIP Call Signaling 15.0(1)XA Adds Session Initiation Protocol/Transport Layer
and SRTP Media with Cisco Security/Transmission Control Protocol (SIP/TLS/TCP)
SRST 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.

SHA2-Cipher-Only Mode for Cisco IOS XE Restricts Secure SIP SRST and Secure SCCP SRST to
Unified Secure SRST Cupertino 17.8.1a only using TLS 1.2 Cipher Suites.

SIP OAuth Client Cisco IOS XE Introduced support for SIP OAuth authentication for
Registration for Unified Cupertino 17.8.1a Secure SRST.
Secure SRST

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 CHAPTER 11
Configuring SIP Trunking on Unified SRST
This chapter describes how to configure SIP trunking on Cisco Unified Survivable Remote Site Telephony
(Unified SRST).
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.
• Unified SRST and Unified Border Element Co-location, on page 341
• Feature Information for Configuring SIP Trunking on Cisco Unified SRST, on page 356

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
• Call Forward
• Call Transfer

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Unified SRST and Unified Border Element Co-location

• 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, on page
343.
The Figure shows a co-located deployment of Unified SRST with Cisco Unified Border Element.

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Configuration Recommendations for Unified SRST and Unified Border Element Co-location

Figure 4: Co-located Deployment of Unifed SRST and Cisco Unified Border Elelement

Configuration Recommendations for Unified SRST and Unified Border Element

Co-location

Note The recommended configurations have considered single SIP trunk dial-peer, acting as both inbound
and outbound dial-peer to handle calls to and from the Service Provider. Similarly, a single dial-peer,
acting as both inbound and outbound dial-peer to handle calls to and from the Communication Manager.

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, on page 346.
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:

1. Dial-peer configuration

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2. Tenant configuration
3. Global configuration

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

dial-peer voice 101 voip

description **CUCM/PBX**
voice-class sip options-keepalive profile 101

• The relevant CLI commands for configuring dial-peer groups are:

voice class dpgdial-peer-group-id (Creates a dial-peer group).
destination dpgdial-peer-group-id(Specifies the dial-peer group from which the outbound dial-peer(s)
is chosen).
• Avoid configuring dial-peer groups on the Service Provider SIP trunk dial-peer.
• Configure the destination pattern (.T) on the Unified Communications Manager dial-peer.
• It is mandatory to configurevoice class tenant on the Service Provider SIP trunk dial-peer router. A
configuration with voice class tenant on the Unified Communications Manager dial-peer is also validated,
though it is not mandatory.
• Configure the CLI command destination dpgdial-peer-group-id (destination dpg 101) on the Unified
Communications Manager dial-peer. This dpg configuration has Service Provider SIP trunk dial-peer
information. 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), on the Service Provider SIP trunk dial-peer. Match the
incoming call from SIP trunk using the address information From URI.
voice class uri 201 sip
host dns:sip-trunk.sample

Under dial-peer:
incoming uri from 201

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• Configure the CLI command transport tcp tls v1.2 undersip-ua configuration mode, not voice class
tenant.
• 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-cryptocrypto-tag command. A sample
configuration is as follows:
voice class srtp-crypto 1
crypto 1 AES_CM_128_HMAC_SHA1_32
crypto 2 AES_CM_128_HMAC_SHA1_80

• 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]
[minsec]].
registrar server expires max120min60
• 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.
• Redundancy is not supported for the co-located deployment of Unified SRST and Unified Border Element.
• Virtual interfaces are not supported for the co-located deployment of Unified SRST and Unified Border
Element.
• 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

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Restrictions

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.
• 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,
on page 341.

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.
dial-peer voice 201 voip
description **SIP-TRUNK.SAMPLE**
session protocol sipv2
session target sip-server
session transport tcp tls
destination e164-pattern-map 201
incoming uri from 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

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Examples

ip qos dscp cs6 signaling

clid strip name
no vad

The following example provides the show running-config command output for the co-located deployment of
Unified SRST and Unified Border Element:
Building configuration...
Current configuration : 15564 bytes
!
! 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 timestamps debug datetime msec
service timestamps log datetime msec
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
!
!
!

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Examples

!
!
!
!
!
!
!
subscriber templating
!
!
!
!
!
!
!
multilink bundle-name authenticated
!
!
!
!
!
!
trunk group 1
xsvc
!
!
crypto pki trustpoint sipgw1
enrollment url http://8.41.20.1:80
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

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9F015EAC A667F30B 505281AA 29EB10A3 F1C75A99 2A224653 F3B985DD F17BC8DD

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
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
supplementary-service media-renegotiate
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
registrar server expires max 240 min 60
!
!
voice class uri 101 sip
host ipv4:10.64.86.136
!
voice class uri 201 sip
host dns:sip-trunk.sample
!
voice class uri 301 sip
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"
!
voice class sip-profiles 201
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"

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rule 8 response ANY sip-header Contact modify "<sips:(.*)>" "<sip:\1;transport=tls>"

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," ""
!
voice class sip-profiles 101
rule 1 request INVITE sip-header Supported modify "100rel," ""
!
voice class sip-profiles 102
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
!
!
voice class e164-pattern-map 201
e164 11[02]
e164 11[68]T
e164 11[025]
e164 +T
e164 0T
e164 2...
!
!
voice class e164-pattern-map 301
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
!
voice class tenant 1
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 expires 60000
timers register 100
timers buffer-invite 1000

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timers dns registrar-cache ttl

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
no pass-thru content custom-sdp
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
!
voice class tenant 2
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 expires 900000
timers register 100
timers buffer-invite 10000
timers dns registrar-cache ttl
sip-server dns:sip-trunk.sample:5060
connection-reuse
asserted-id pai
bind control source-interface GigabitEthernet0/0/1
bind media source-interface GigabitEthernet0/0/1
no pass-thru content custom-sdp
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
!
voice class tenant 3
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 expires 900000
timers register 500
timers buffer-invite 1000
timers dns registrar-cache ttl
sip-server dns:sipp.sample
connection-reuse
asserted-id pai
bind control source-interface GigabitEthernet0/0/1
bind media source-interface GigabitEthernet0/0/1
no pass-thru content custom-sdp
conn-reuse
sip-profiles 3000
outbound-proxy dns:sipp.sample:6600
privacy-policy passthru
call-route p-called-party-id
midcall-signaling preserve-codec
!
voice class tenant 4
timers expires 60000
timers buffer-invite 10000
connection-reuse

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asserted-id pai
bind control source-interface GigabitEthernet0/0/0
bind media source-interface GigabitEthernet0/0/0
no pass-thru content custom-sdp
privacy-policy passthru
call-route p-called-party-id
midcall-signaling preserve-codec
!
voice class srtp-crypto 1
crypto 1 AES_CM_128_HMAC_SHA1_32
crypto 2 AES_CM_128_HMAC_SHA1_80
!
!
!
voice register global
default mode
no allow-hash-in-dn
max-dn 40
max-pool 40
!
voice register pool 1
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
!
!
voice hunt-group 2 sequential
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
!
!

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!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
template 1
!
!
!
!
!
interface GigabitEthernet0/0/0
ip address 8.39.23.13 255.255.0.0
ip nat inside
media-type rj45
negotiation auto
!
interface GigabitEthernet0/0/1
ip address 10.64.86.64 255.255.0.0
ip nat outside
negotiation auto
!
interface Service-Engine0/1/0
!
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
no ip http secure-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

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tftp-server flash:kern88xx.12-0-1MN-113.sbn
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
!
mgcp profile default
!
!
!
!
dial-peer voice 201 voip
description **SIP-TRUNK.SAMPLE**
session protocol sipv2
session target sip-server
session transport tcp tls
destination e164-pattern-map 201
incoming uri from 201
voice-class codec 1
voice-class sip url sips
voice-class sip profiles 201
voice-class sip tenant 1
voice-class sip srtp-crypto 1
dtmf-relay rtp-nte
srtp
fax-relay ecm disable

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Examples

fax rate 14400

clid strip name
no vad
!
dial-peer voice 301 voip
description **SIP-TRUNK.SAMPLE**
session protocol sipv2
session target sip-server
session transport tcp
destination e164-pattern-map 301
incoming uri from 201
voice-class codec 1
voice-class sip url sip
voice-class sip profiles 201
voice-class sip tenant 2
dtmf-relay rtp-nte
srtp fallback
fax-relay ecm disable
fax rate 14400
clid strip name
no vad
!
dial-peer voice 401 voip
description **SIP-TRUNK.SAMPLE**
destination-pattern 4...
session protocol sipv2
session target sip-server
session transport udp
incoming uri from 301
voice-class codec 1
voice-class sip url sip
voice-class sip profiles 201
voice-class sip tenant 3
dtmf-relay rtp-nte
fax-relay ecm disable
fax rate 14400
clid strip name
no vad
!
dial-peer voice 101 voip
description **CUCM/PBX**
destination-pattern .T
session protocol sipv2
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 tenant 4
voice-class sip srtp-crypto 1
voice-class sip options-keepalive profile 101
dtmf-relay rtp-nte
srtp fallback
fax-relay ecm disable
fax rate 14400
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711alaw
no vad
!
!
presence
!

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Feature Information for Configuring SIP Trunking on Cisco Unified SRST

gateway
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

Feature Information for Configuring SIP Trunking on Cisco

Unified SRST
Not all commands may be available in your Cisco IOS Software release. For release information about specific
command, see the command reference documentation.
Use Cisco Feature Navigator to find information about the platform support and software image support.
Cisco Feature Navigator enables you to determine which Cisco IOS and Cisco Catalyst operating system
software images support a specific software release, feature set, or platform. To access Cisco Feature Navigator,
go to http://www.cisco.com/go/cfn . You do not need an account on Cisco.com.

Note The 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.

The following table lists the release history for this feature.

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Feature Information for Configuring SIP Trunking on Cisco Unified SRST

Feature Name Releases Feature Information

Cisco Unified SRST and Cisco Cisco IOS XE Fuji Added Support for co-location of Cisco Unified
Unified Border Element 16.7.1 SRST and Cisco Unified Border Element on
Co-location Cisco 4000 Series Integrated Services Router.

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 CHAPTER 12
Integrating Voice Mail 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.
• Information About Integrating Voicemail with Cisco Unified SRST, on page 359
• How to Integrate Voicemail with Cisco Unified SCCP and SIP SRST, on page 360
• Configuration Examples for Unified SRST, on page 374
• How to Configure DTMF Relay for SIP Applications and Voicemail, on page 377

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 5 - Cisco Unified Communications Manager
Fallback with BRI or PRI). 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 6 - Cisco
Unified Communications Manager Fallback with PSTN).
From Unified SRST Release 12.0 onwards, Unified SRST supports voicemail on IPv6 protocols for SIP IP
phones.

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How to Integrate Voicemail with Cisco Unified SCCP and SIP SRST

Figure 5: Cisco Unified Communications Manager Fallback with BRI or PRI

Figure 6: Cisco Unified Communications Manager Fallback with PSTN

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.

How to Integrate Voicemail with Cisco Unified SCCP and SIP

SRST
This section contains the following tasks:

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Configuring Direct Access to Voicemail

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.

Table 3: 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 (.) Indicates that the preceding digit occurred zero or more times; similar to the
wildcard usage.

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.

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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. Do the following to configure a video codec:
• video codec codec
6. exit

DETAILED STEPS

Command or Action Purpose

Step 1 dial-peer voice tag {pots |voatm | vofr | voip} (FXO or FXS and BRI or PRI) Defines a particular dial
peer, specifies the method of voice encapsulation, and enters
Example:
dial-peer configuration mode. The dial-peer command
Router(config)# dial-peer voice 1002 pots provides different syntax for individual routers. This
example is 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
Example:
plan) to be used for a dial peer.
Router(config-dial-peer)# destination-pattern 1100T
• +: (Optional) Character that indicates an E.164
standard number.
• string: See Table Valid Entries for the String Argument
in the destination-pattern command.
• 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
|slot/port:ds0-group-no} a specific voice port on Cisco routers.
Example:

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Command or Action Purpose

Router(config-dial-peer)# port 1/1/1 • slot-number: Number of the slot in the router in which
the voice interface card (VIC) is installed. Valid entries
are 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.

Step 4 forward-digits {num-digit |all | extra} (Optional for FXO or FXS) Specifies which digits to
forward for voice calls.
Example:
Router(config-dial-peer)# forward-digits all • num-digit: The number of digits to be forwarded. If
the 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 theno 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 Do the following to configure a video codec: Configures a video codec at the dial peer level.
• video codec codec
Example:
For Video Codec
Device(config-dial-peer)# video codec h261

Step 6 exit (FXO or FXS and BRI or PRI) Exits dial-peer configuration
mode.
Example:
Router(config-dial-peer)# exit

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Examples

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
dial-peer voice 1102 pots
destination-pattern 1101
port 1/1/1
forward-digits all
dial-peer voice 1103 pots
destination-pattern 1101
port 1/1/1
forward-digits all
dial-peer voice 1104 pots
destination-pattern 1101
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
interface Serial2/0:23
no ip address
no logging event link-status
isdn switch-type primary-net5
isdn incoming-voice voice
isdn T309-enable
no cdp enable
voice-port 2/0:23
dial-peer voice 1102 pots
destination-pattern 1101T
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.

SUMMARY STEPS
1. call-manager-fallback
2. voicemail phone-number

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3. call-forward busy directory-number

4. call-forward noan directory-number timeout seconds
5. exit
6. voice register pool tag
7. call-forward b2bua busy directory-number
8. call-forward b2bua noan directory-numbertimeout seconds
9. exit

DETAILED STEPS

Command or Action Purpose

Step 1 call-manager-fallback Enters call-manager-fallback configuration mode.
Example:
Router(config)# call-manager-fallback

Step 2 voicemail phone-number Configures the telephone number that is dialed when the
message button on a Cisco Unified SCCP IP Phone is
Example:
pressed.
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.
Example:
Router(config-cm-fallback)# call-forward busy directory-number : Selected directory number representing
2000 a fully qualified E.164 number. This number can contain
“.” wildcard characters that correspond to the right-justified
digits in the directory number extension.

Step 4 call-forward noan directory-number timeout seconds Configures call forwarding to another number when no
answer is received from the Cisco SCCP IP phone.
Example:
Router(config-cm-fallback)# call-forward noan directory-number : Selected directory number representing
2000 timeout 10 a fully qualified E.164 number. This number can contain
“.” 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.

Step 5 exit Exits call-manager-fallback configuration mode.

Example:
Router(config-cm-fallback)# exit

Step 6 voice register pool tag Enters voice register pool configuration mode.
Example:
Router(config)# voice register pool 1

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Command or Action Purpose

Step 7 call-forward b2bua busy directory-number Configures call forwarding to another number when the
Cisco SIP IP phone is busy.
Example:
Router(config-register-pool)# call-forward directory-number : Selected directory number representing
b2bua busy 2000 a fully qualified E.164 number. This number can contain
“.” wildcard characters that correspond to the right-justified
digits in the directory number extension.

Step 8 call-forward b2bua noan directory-numbertimeout Configures call forwarding to another number when no
seconds answer is received from the Cisco SIP IP phone.
Example: directory-number : Selected directory number representing
Router(config-register-pool)# call-forward noan a fully qualified E.164 number. This number can contain
2000 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.

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).
call-manager-fallback
voicemail 1101
call-forward busy 1101
call-forward noan 1101 timeout 3
voice register pool 1
call-forward b2bua busy 1101
call-forward b2bua noan 1101 timeout 3

Redirecting to Cisco Unified Communications Manager Gateway

Before you begin

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

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CM to declare the last redirected number and the originally dialed number to and from configured devices
and applications.

SUMMARY STEPS
1. From any page in Cisco Unified CM, click Device and Gateway
2. From the Find and List Gateways page, click Find.
3. From the Find and List Gateways page, choose a device name.
4. From the Gateway Configuration page, check Redirecting Number IE Delivery - Outgoing.

DETAILED STEPS

Command or Action Purpose

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 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 the below figure, this is accomplished
through the voicemail and pattern commands.

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Figure 7: How Voicemail Dial Sequence 1101#6000#2 Is Configured in Cisco Unified SRST

The # cgn #2, # cdn #2, and # fdn #2 portions of the pattern commands shown in 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 the following figure, there are three types of tokens that correspond to three possible call states
during voicemail forwarding.
Figure 8: How Numbers Are Extracted from Tokens

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).
• An external trunk call reaches an unanswered extension and the call is forwarded to voicemail ( pattern
trunk-to-ext no-answer command).

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Prerequisites

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).
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.

Configuring Call Forwarding to Voicemail

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}] [tag3 {CGN | CDN |
FDN}] [last-tag]
4. pattern ext-to-ext no-answertag1 {CGN | CDN | FDN} [tag2 {CGN | CDN | FDN}] [tag3 {CGN |
CDN | FDN}] [last-tag]
5. pattern trunk-to-ext busytag1{CGN | CDN | FDN} [tag2 {CGN | CDN | FDN}] [tag3 {CGN | CDN
| FDN}] [last-tag]
6. pattern trunk-to-ext no-answertag1 {CGN | CDN | FDN} [tag2 {CGN | CDN | FDN}] [tag3 {CGN |
CDN | FDN}] [last-tag]

DETAILED STEPS

Command or Action Purpose

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 {CGN Configures the DTMF digit pattern forwarding necessary
|CDN | FDN}] [tag3 {CGN |CDN | FDN}] [last-tag] to activate the voicemail system when the user presses the
messages button on the phone.
Example:
Router(config-vm-int)# pattern direct 2 CGN * • tag1: Alphanumeric string fewer than four DTMF
digits in length. The alphanumeric string consists of a
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.

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Command or Action Purpose

• tag2and 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 | FDN} [tag2 Configures the DTMF digit pattern forwarding necessary
{CGN | CDN | FDN}] [tag3 {CGN | CDN | FDN}] to activate the voicemail system once an internal extension
[last-tag] attempts to connect to a busy extension and the call is
forwarded to voicemail.
Example:
Router(config-vm-int)# pattern ext-to-ext busy 7
FDN * CGN *

Step 4 pattern ext-to-ext no-answertag1 {CGN | CDN | FDN} Configures the DTMF digit pattern forwarding necessary
[tag2 {CGN | CDN | FDN}] [tag3 {CGN | CDN | FDN}] to activate the voicemail system once an internal extension
[last-tag] fails to connect to an extension and the call is forwarded to
voicemail. For argument and keyword information, see
Example:
Step 1.
Router(config-vm-int)# pattern ext-to-ext no-answer
5 FDN * CGN *

Step 5 pattern trunk-to-ext busytag1{CGN | CDN | FDN} [tag2 Configures the DTMF digit pattern forwarding necessary
{CGN | CDN | FDN}] [tag3 {CGN | CDN | FDN}] to activate the voicemail system once an external trunk call
[last-tag] reaches a busy extension and the call is forwarded to
voicemail.
Example:
Router(config-vm-int)# pattern trunk-to-ext busy
6 FDN * CGN *

Step 6 pattern trunk-to-ext no-answertag1 {CGN | CDN | FDN} Configures the DTMF digit pattern forwarding necessary
[tag2 {CGN | CDN | FDN}] [tag3 {CGN | CDN | FDN}] to activate the voicemail system when an external trunk call
[last-tag] reaches an unanswered extension and the call is forwarded
to voicemail.
Example:
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

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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.
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
1. call-manager-fallback
2. configure terminal
3. mwi relay
4. mwi reg-e164
5. exit
6. sip-ua
7. mwi-server {ipv4:destination-address | dns:host-name}[expires seconds[port port][transport ] {tcp |
udp}] [unsolicited]]
8. exit

DETAILED STEPS

Command or Action Purpose

Step 1 call-manager-fallback Enters call-manager-fallback configuration mode.
Example:
Router(config)# call-manager-fallback

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Command or Action Purpose

Step 2 configure terminal Enters global configuration mode.
Example:
Router# configure terminal

Step 3 mwi relay Enables the Unified SRST router to relay MWI information
to remote Cisco IP phones.
Example:
Router(config-cm-fallback)# mwi relay

Step 4 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

Step 5 exit Exits call-manager-fallback configuration mode.

Example:
Router(config-cm-fallback)# exit

Step 6 sip-ua Enters SIP user-agent configuration mode.

Example:
Router(config)# sip-ua

Step 7 mwi-server {ipv4:destination-address | Configures voicemail server settings on a voice gateway or

dns:host-name}[expires seconds[port port][transport ] user agent. The IP address and port for the SIP-based MWI
{tcp | udp}] [unsolicited]] server should be in the same LAN as the voicemail server.
The MWI server is a Cisco Unified SRST router. Keywords
Example:
and arguments are as follows:
Router(config-sip-ua)# mwi-server ipv4:10.0.2.254
• 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.

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Command or Action Purpose

Step 8 exit Exits SIP user-agent configuration mode.
Example:
Router(config-sip-ua)# exit

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
2. configure terminal
3. sip-ua
4. mwi-server {ipv4:destination-address | dns:host-name}[unsolicited]
5. exit
6. voice register global
7. mwi unsolicited
8. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
Example: • Enter your password if prompted.
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 sip-ua Enters Session Initiation Protocol (SIP) user agent (ua)
configuration mode for configuring the user agent.
Example:
Router(config)# sip-ua

Step 4 mwi-server {ipv4:destination-address | Configures voicemail server settings on a voice gateway or

dns:host-name}[unsolicited] user agent. Keywords and arguments are as follows:
Example: • ipv4:destination-address: IP address of the voicemail
For g711alaw Codec server.

Router(config-sip-ua)# mwi-server ipv4:10.0.2.254 • dns:host-name: The argument should contain the

unsolicited complete hostname to be associated with the target
Or address; for example, dns:test.cisco.com.

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Command or Action Purpose

Router(config-sip-ua)# mwi-server • unsolicited: Requires the voicemail server to send a
dns:server.yourcompany.com unsolicited
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 5 exit Exits SIP user-agent configuration mode.

Example:
Router(config-sip-ua)# exit

Step 6 voice register global Enters voice register global configuration mode to set
parameters for all supported SIP phones in SIP SRST mode.
Example:
Router(config)# voice register global

Step 7 mwi unsolicited Enables all SIP phones to receive MWI notification.
Example:
Router(config-register-global)# mwi unsolicited

Step 8 end Exits to privileged EXEC mode.

Example:
Router(config-register-global)# end

Configuration Examples for Unified SRST

This section provides the following configuration examples:

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
!
dial-peer voice 101 pots
destination-pattern 14011
port 3/0/0
!
dial-peer voice 102 pots
preference 1
destination-pattern 14011
port 3/0/1
!
dial-peer voice 103 pots
preference 2
destination-pattern 14011
port 3/1/0
!
dial-peer voice 104 pots

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Configuring Central Location Voicemail System (FXO and FXS): Example

destination-pattern 14011
port 3/1/1
!
! Cisco Unified SRST configuration
!
call-manager-fallback
max-ephones 24
max-dn 144
ip source-address 1.4.214.104 port 2000
voicemail 14011
call-forward busy 14011
call-forward noan 14011 timeout 3
! 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
!
dial-peer voice 101 pots
destination-pattern 14011
port 3/0/0
!
! Cisco Unified SRST configuration
!
call-manager-fallback
max-ephones 24
max-dn 144
ip source-address 1.4.214.104 port 2000
voicemail 14011
call-forward busy 14011
call-forward noan 14011 timeout 3
!
! 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 *

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Configuring Voicemail Access over FXO and FXS: Example

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.
voice-port 1/1/1
timing digit 250
timing inter-digit 250
dial-peer voice 1102 pots
destination-pattern 1101T
port 1/1/1
call-manager-fallback
timeouts interdigit 5
ip source-address 1.6.0.199 port 2000
max-ephones 24
max-dn 24
transfer-pattern 3...
voicemail 1101
call-forward busy 1101
call-forward noan 1101 timeout 3
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
interface Serial2/0:23
no ip address
no logging event link-status
isdn switch-type primary-net5
isdn incoming-voice voice
isdn T309-enable
no cdp enable
voice-port 2/0:23
dial-peer voice 1102 pots
destination-pattern 1101T
direct-inward-dial
port 2/0:23
call-manager-fallback
timeouts interdigit 5
ip source-address 1.6.0.199 port 2000
max-ephones 24
max-dn 24
transfer-pattern 3...
voicemail 1101
call-forward busy 1101

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Message Waiting Indication for SIP SRST: Example

call-forward noan 1101 timeout 3

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
For SIP SRST forwarding call to voicemail configuration, see the Configuring Call Handling section.

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 for SIP applications can be used in two voicemail situations:

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:
• When SIP is used to connect a Cisco Unified SRST system to a remote SIP-based IVR or voicemail
application, such as Cisco Unity.

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• 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 voicetagvoip
2. dtmf-relay rtp-nte
3. dtmf-relay rtp-nte
4. exit
5. sip-ua
6. notify telephone-event max-durationtime
7. exit

DETAILED STEPS

Command or Action Purpose

Step 1 dial-peer voicetagvoip Enters dial-peer configuration mode.
Example:
Router(config)# dial-peer voice 2 voip

Step 2 dtmf-relay rtp-nte Enters global configuration mode.

Example:
Router(config-dial-peer)# dtmf-relay rtp-nte

Step 3 dtmf-relay rtp-nte Forwards DTMF tones by using Real-Time Transport

Protocol (RTP) with the Named Telephone Event (NTE)
Example:
payload type.
Router(config)# voice register global

Step 4 exit Exits dial-peer configuration mode.

Example:
Router(config-dial-peer)# exit

Step 5 sip-ua Enables SIP user-agent configuration mode.

Example:
Router(config)# sip-ua

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Command or Action Purpose

Step 6 notify telephone-event max-durationtime Configures the maximum time interval allowed between
two consecutive NOTIFY messages for a single DTMF
Example:
event.
Router(config-sip-ua)# notify telephone-event
max-duration 2000 • max-durationtime : Time interval between
consecutive NOTIFY messages for a single DTMF
event, in milliseconds. Range is from 500 to 3000.
Default is 2000.

Step 7 exit Exits SIP user-agent configuration mode.

Example:
Router(config-sip-ua)# exit

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:
dial-peer voice 123 voip
destination-pattern [12]...
monitor probe icmp-ping
session protocol sipv2
session target ipv4:10.8.17.42
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
1. dial-peer voicetagvoip
2. dtmf-relay sip-notify
3. exit
4. sip-ua
5. notify telephone-event max-durationtime
6. exit

DETAILED STEPS

Command or Action Purpose

Step 1 dial-peer voicetagvoip Enters dial-peer configuration mode.
Example:
Router(config)# dial-peer voice 2 voip

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Command or Action Purpose

Step 2 dtmf-relay sip-notify Forwards DTMF tones using SIP NOTIFY messages.
Example:
Router(config-dial-peer)# dtmf-relay sip-notify

Step 3 exit Exits dial-peer configuration mode.

Example:
Router(config-dial-peer)# exit

Step 4 sip-ua Enables SIP user-agent configuration mode.

Example:
Router(config)# sip-ua

Step 5 notify telephone-event max-durationtime Configures the maximum time interval allowed between
two consecutive NOTIFY messages for a single DTMF
Example:
event.
Router(config-sip-ua)# notify telephone-event
max-duration 2000 • max-durationtime : Time interval between
consecutive NOTIFY messages for a single DTMF
event, in milliseconds. Range is from 500 to 3000.
Default is 2000.

Step 6 exit Exits SIP user-agent configuration mode.

Example:
Router(config-sip-ua)# exit

Troubleshooting Tips
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
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

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Address types supported:IP4

Transport types supported:RTP/AVP udptl

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 CHAPTER 13
Setting Video Parameters
This chapter describes how to set video parameters for a Cisco Unified Survivable Remote Site Telephony
(SRST) Router.
• Prerequisites for Setting Video Parameters, on page 383
• Restrictions for Setting Video Parameters, on page 384
• Information About Setting Video Parameters, on page 384

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 Communications Manager 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 you install the correct video firmware 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)
Cisco Unified IP Phone 7960G version 6.0(4)
Cisco Unified IP Phone 7970G version 6.0(2)
• 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.

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 Setting Video Parameters
Restrictions for Setting Video Parameters

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 does not support 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 an incoming 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 Communications Manager. 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 Communications Manager. 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
the Cisco Unified SRST audio calls.
To set video parameters, refer the following concepts:

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Matching Endpoint Capabilities

Matching Endpoint Capabilities

Cisco Unified SRST stores Endpoint capabilities during the phone registration. These capabilities are used to
match with other endpoints during the call setup. Endpoints can update at any time; however, the router
recognizes endpoint capability changes only during the call setup. If you add a video feature to a phone, the
information about it is updated in the router’s internal data structure. However, the information does not take
effect until the next call. If a video feature is revoked, the router continues to view the video capability until
the call stops. However, there is no video stream that is exchanged between the two endpoints.

Note The endpoint capability match is executed every time when an incoming 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 the call setup.

Call Fallback to and Audio-Only Endpoint

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.
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.

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Call Setup Between Two Local SCCP Endpoints

A call-type flag is set during the call setup on the basis of the endpoint and capability match. After call setup,
the call -type flag is used to determine whether an extra video-media path is required. Call signaling is managed
by the Cisco Unified Communications Manager Express 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 the media setup. During the media setup, a message is
sent to establish the video-media path. If the endpoint responds, the video-media path is established and
invokes a start-multimedia-transmission function.

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 stop 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 the 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 Communications Manager Express 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, stops 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.

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
============================

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How to Set Video Parameters for Cisco Unified SRST

How to Set Video Parameters for Cisco Unified SRST

When you enable the Cisco Unified SRST, do not reconfigure the Cisco Unified IP phones for video capabilities.
All ephones retain the same configuration used with Cisco Unified Communications Manager. However, you
can set video parameters for Cisco Unified SRST.
The following are the task for setting Video parameters for Cisco Unified SRST:

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
2. configure terminal
3. voice service voip
4. h323
5. call start slow

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
Example: • Enter your password if prompted.
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 voice service voip Enters voice-service configuration mode.

Example:
Router(config)# voice service voip

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

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Verifying Cisco Unified SRST

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

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
Example: • Enter your password if prompted.
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
Example:
your network while in fallback mode.
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.

Example
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
ip source-address 10.1.1.1 port 2000
max-video-bit-rate 384(kbps)
max-ephones 52
max-dn 110
max-conferences 16 gain -6
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

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date-format dd-mm-yy
timezone 0 Greenwich Standard Time
call-forward busy 6001
call-forward noan 6001 timeout 8
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:
7910: 34
7935: 34
7936: 34
7940: 34
7960: 34
7970: 34
Log (table parameters):
max-size: 150
retain-timer: 15
transfer-system full-consult
local directory service: enabled.
ephone-dn 1
number 1001
name 1001
description 1001
label 1001
preference 0 secondary 9
huntstop
call-forward busy 6001
call-forward noan 6001 timeout 8
call-waiting beep
ephone-dn 2
number 1002
name 1002
description 1002
preference 0 secondary 9
huntstop
call-forward busy 6001
call-forward noan 6001 timeout 8
call-waiting beep
ephone-dn 3
preference 0 secondary 9
huntstop
call-waiting beep
ephone-dn 4
preference 0 secondary 9
huntstop
call-waiting beep
ephone-dn 5
preference 0 secondary 9
huntstop
call-waiting beep
ephone-dn 6
preference 0 secondary 9
huntstop
call-waiting beep
ephone-dn 7
preference 0 secondary 9
huntstop
call-waiting beep
ephone-dn 8
preference 0 secondary 9

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huntstop
call-waiting beep
ephone-dn 9
preference 0 secondary 9
huntstop
call-waiting beep
ephone-dn 10
preference 0 secondary 9
huntstop
call-waiting beep
ephone-dn 11
preference 0 secondary 9
huntstop
call-waiting beep
ephone-dn 12
preference 0 secondary 9
huntstop
call-waiting beep
ephone-dn 13
preference 0 secondary 9
huntstop
call-waiting beep
ephone-dn 14
preference 0 secondary 9
huntstop
call-waiting beep
ephone-dn 15
preference 0 secondary 9
huntstop
call-waiting beep
ephone-dn 16
preference 0 secondary 9
huntstop
call-waiting beep
ephone-dn 17
preference 0 secondary 9
huntstop
call-waiting beep
ephone-dn 18
preference 0 secondary 9
huntstop
call-waiting beep
ephone-dn 19
preference 0 secondary 9
huntstop
call-waiting beep
ephone-dn 20
preference 0 secondary 9
huntstop
call-waiting beep
Number of Configured ephones 0 (Registered 2)
voice-port 50/0/1
station-id number 1001
station-id name 1001
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

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!
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
!
dial-peer voice 20055 pots
destination-pattern 1001
huntstop
call-forward busy 6001
call-forward noan 6001
progress_ind setup enable 3
port 50/0/1
dial-peer voice 20056 pots
destination-pattern 1002
huntstop
call-forward busy 6001
call-forward noan 6001
progress_ind setup enable 3
port 50/0/2
dial-peer voice 20057 pots
huntstop
progress_ind setup enable 3
port 50/0/3
dial-peer voice 20058 pots
huntstop
progress_ind setup enable 3
port 50/0/4
dial-peer voice 20059 pots
huntstop
progress_ind setup enable 3
port 50/0/5
dial-peer voice 20060 pots
huntstop
progress_ind setup enable 3
port 50/0/6
dial-peer voice 20061 pots

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huntstop
progress_ind setup enable 3
port 50/0/7
dial-peer voice 20062 pots
huntstop
progress_ind setup enable 3
port 50/0/8
dial-peer voice 20063 pots
huntstop
progress_ind setup enable 3
port 50/0/9
dial-peer voice 20064 pots
huntstop
progress_ind setup enable 3
port 50/0/10
dial-peer voice 20065 pots
huntstop
progress_ind setup enable 3
port 50/0/11
dial-peer voice 20066 pots
huntstop
progress_ind setup enable 3
port 50/0/12
dial-peer voice 20067 pots
huntstop
progress_ind setup enable 3
port 50/0/13
dial-peer voice 20068 pots
huntstop
progress_ind setup enable 3
port 50/0/14
dial-peer voice 20069 pots
huntstop
progress_ind setup enable 3
port 50/0/15
dial-peer voice 20070 pots
huntstop
progress_ind setup enable 3
port 50/0/16
dial-peer voice 20071 pots
huntstop
progress_ind setup enable 3
port 50/0/17
dial-peer voice 20072 pots
huntstop
progress_ind setup enable 3
port 50/0/18
dial-peer voice 20073 pots
huntstop
progress_ind setup enable 3
port 50/0/19
dial-peer voice 20074 pots
huntstop
progress_ind setup enable 3
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

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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 SEPFFDD00000002.cnf.xml
tftp-server system:/its/XMLDefault7960.cnf.xml alias SEPFFDD00000003.cnf.xml
tftp-server system:/its/XMLDefault7960.cnf.xml alias SEPFFDD00000004.cnf.xml
tftp-server system:/its/XMLDefault7960.cnf.xml alias SEPFFDD00000005.cnf.xml
tftp-server system:/its/XMLDefault7960.cnf.xml alias SEPFFDD00000006.cnf.xml
tftp-server system:/its/XMLDefault7960.cnf.xml alias SEPFFDD00000007.cnf.xml
tftp-server system:/its/XMLDefault7960.cnf.xml alias SEPFFDD00000008.cnf.xml
tftp-server system:/its/XMLDefault7960.cnf.xml alias SEPFFDD00000009.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
tftp-server system:/its/XMLDefault7960.cnf.xml alias SEPFFDD00000010.cnf.xml
tftp-server system:/its/XMLDefault7960.cnf.xml alias SEPFFDD00000011.cnf.xml
tftp-server system:/its/XMLDefault7960.cnf.xml alias SEPFFDD00000012.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.

SUMMARY STEPS
1. enable
2. configure terminal
3. dcall-manager-fallback
4. video
5. maximum bit-rate value

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
Example: • Enter your password if prompted.
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 dcall-manager-fallback Enters call-manager-fallback configuration mode.

Example:
Router(config)# call-manager-fallback

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Command or Action Purpose

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

Example
The following example shows the configuration for video with Cisco Unified SRST:
call-manager-fallback
video
maximum bit-rate 384
max-conferences 2 gain -6
transfer-system full-consult
ip source-address 10.0.1.1 port 2000
max-ephones 52
max-dn 110
dialplan-pattern 1 4084442... extension-length 4
transfer-pattern .T
keepalive 45
voicemail 6001
call-forward pattern .T
call-forward busy 6001
call-forward noan 6001 timeout 3
moh music-on-hold.au
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.
1. For SCCP endpoint troubleshooting, use the following debug commands:
• Debug cch323 video: Enables the 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
are sent or received.
• Debug h245 asn1: Displays ASN.1 contents of H.245 messages that are sent or received.
• Debug VoIP CCAPI inout: Displays the execution path through the call-control-application
programming interface (CPI).

2. For ephone troubleshooting, use the following debug commands:

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• 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.

3. For basic video-to-video call checking, use the following show commands:
• Show call active video: Displays call information for SCCP video CallsInProgress.
• Show ephone off hook: 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.

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 CHAPTER 14
Monitoring and Maintaining Cisco Unified SRST
• Monitoring and Maintaining Cisco Unified SRST, on page 397

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
Cisco Unified SRST Router.

Router# show call-manager-fallback dial-peer Displays the output of the dial peers of the Cisco
Unified SRST Router.

Router# show call-manager-fallback ephone-dn Displays Cisco Unified IP Phone destination numbers
when in Cisco Unified Communications 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 ephonephone 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.

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Monitoring and Maintaining Cisco Unified SRST

Command Purpose

Router# show ephone summary Displays a summary of all Cisco Unified IP Phones.

Router# show ephone Displays Unified IP Phone status for a specific phone
telephone-numberphone-number number.

Router# show ephone unregistered Displays Unified IP Phone status for all unregistered
phones.

Router# show ephone-dntag 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.

Router# show running-config Display 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 information.

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 information.

Router# show voice register pool tag Displays specific SIP phone voice register Pool detail
information.

Router# show voice register dial-peers Displays SIP-SRST created dial peer.

Router# show voice register dn all Displays all config voice register directory number
detail information.

Router# show voice register dn tag Displays specific voice register directory number
detail information.

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 APPENDIX A
Appendix A: Configuring Cisco Unified SIP SRST
Features Using Redirect Mode
This chapter describes Cisco Unified Session Initiation Protocol (SIP) Survivable Remote Site Telephony
(SRST) features using redirect mode.

Note This chapter applies to version 3.0 only.

• Prerequisites for Cisco Unified SIP SRST Features Using Redirect Mode, on page 399
• Restrictions for Cisco Unified SIP SRST Features Using Redirect Mode, on page 399
• Information About Cisco Unified SIP SRST Features Using Redirect Mode, on page 400
• How to Configure Cisco Unified SIP SRST Features Using Redirect Mode, on page 400
• Configuration Examples for Cisco Unified SIP SRST Features Using Redirect Mode, on page 404

Prerequisites for Cisco Unified SIP SRST Features Using

Redirect Mode
Complete the prerequisites documented in the Cisco Unified SCCP and SIP SRST Feature Overview, on page
41 chapter.

Restrictions for Cisco Unified SIP SRST Features Using Redirect

Mode
See the restrictions documented in the Cisco Unified SRST Feature Overview chapter.
See the restrictions documented in the Cisco Unified SCCP and SIP SRST Feature Overview, on page 41
chapter.

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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 if 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.
• Other 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
The call redirect enhancement supports calls from a local SIP phone to another local SIP phone through the
Cisco IOS voice gateway. Before 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 Audio and Video Codecs at the Dial Peer Level

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

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Configuring Call Redirect Enhancements to Support Calls On a Specific VoIP Dial Peer

2. configure terminal
3. voice service voip
4. redirect ip2ip
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
Example: • Enter your password if prompted.
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 voice service voip Enters voice service configuration mode.

Example:
Router(config)# voice service voip

Step 4 redirect ip2ip Configures a video codec at the dial peer level. Redirects
SIP phone calls to SIP phone calls globally on a gateway
Example:
using the Cisco IOS voice gateway.
Router(config-voi-srv)# redirect ip2ip

Step 5 end Returns to privileged EXEC mode.

Example:
Router(config-voi-srv)# end

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.

Before you begin

The redirect ip2ip command must be configured on an inbound dial peer of the gateway.

SUMMARY STEPS
1. enable
2. configure terminal
3. dial-peer voice tag voip

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Configuring Sending 300 Multiple Choice Support

4. application application-name
5. redirect ip2ip
6. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
Example: • Enter your password if prompted.
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 dial-peer voice tag voip Enters dial-peer configuration mode.

Example: • tag : A number that uniquely identifies the dial peer
Router(config)# dial-peer voice 25 voip (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.

Step 4 application application-name Enables a specific application on a dial peer.

Example: • For SIP, the default Tool Command Language (Tcl)
Router(config-dial-peer)# application session application (from the Cisco IOS image) is session and
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

Step 6 end Returns to privileged EXEC mode.

Example:
Router(config-dial-peer)# end

Configuring Sending 300 Multiple Choice Support

Before 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.

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Appendix A: Configuring Cisco Unified SIP SRST Features Using Redirect Mode

SUMMARY STEPS
1. enable
2. configure terminal
3. voice service voip
4. sip
5. redirect contact order [best-match | longestmatch]
6. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
Example: • Enter your password if prompted.
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 voice service voip Enters voice service configuration mode.

Example:
Router(config)# voice service voip

Step 4 sip Enters SIP configuration mode.

Example:
Router(config-voi-srv)# sip

Step 5 redirect contact order [best-match | longestmatch] Sets the order of contacts in the 300 Multiple Choice
message. The keywords are defined as follows:
Example:
Router(conf-serv-sip)# redirect contact order • best-match : Uses the current system configuration
best-match to set the order of contacts.
• longestmatch : 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.

Step 6 end Returns to privileged EXEC mode.

Example:
Router(config-serv-sip)# end

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Configuration Examples for Cisco Unified SIP SRST Features Using Redirect Mode

Configuration Examples for Cisco Unified SIP SRST Features

Using Redirect Mode
This section provides the following configuration example:

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
registrar server expires max 600 min 60
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.
!
voice register pool 1
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
!
voice register pool 2
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
!
voice register pool 3
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
!
voice register pool 4
id network 10.2.161.0 mask 255.255.255.0
number 1 94... preference 1

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Appendix A: Configuring Cisco Unified SIP SRST Features Using Redirect Mode

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
Rule 0 94 91
!
! Sets up proxy monitoring.
!
call fallback active
!
dial-peer cor custom
name 95
name 94
name 91
!
! Configures COR values to be applied to the voice register pool.
!
dial-peer cor list call95
member 95
!
dial-peer cor list call94
member 94
!
dial-peer cor list call91
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
!
dial-peer voice 91500 pots
corlist incoming call91
corlist outgoing call91
destination-pattern 91500
port 1/0/0
!

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Appendix A: Configuring Cisco Unified SIP SRST Features Using Redirect Mode

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 APPENDIX B
Appendix 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.
• Prerequisites for Using Cisco Unified SRST Gateways as a Multicast MOH Resource, on page 407
• Restrictions for Using Cisco Unified SRST Gateways as a Multicast MOH Resource, on page 408
• Information About Using Cisco Unified SRST Gateways as a Multicast MOH Resource, on page 408
• How to Use Cisco Unified SRST Gateways as a Multicast MOH Resource, on page 412
• Configurations Examples for Cisco Unified SRST Gateways, on page 431
• Feature Information for Cisco Unified SRST as a Multicast MOH Resource, on page 432
• Where to Go Next, on page 433

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.

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Restrictions for Using Cisco Unified SRST Gateways as a Multicast MOH Resource

• 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

Multicast MOH Resource
• 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.

Information About Using Cisco Unified SRST Gateways as a

Multicast MOH Resource
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

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

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Appendix B: Integrating Cisco Unified Communications Manager and Cisco Unified SRST to Use Cisco Unified SRST as a Multicast MOH Resource

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.
The following figure 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.

Figure 9: Multicast MOH from Cisco Unified SRST Flash Memory

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Codecs, Port Numbers, and IP Addresses

Codecs, Port Numbers, and IP Addresses

Cisco Unified SRST multicast MOH supports G.711 only. 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, 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.
Figure 10: IP Address and Port Usage for G.711 and G.729 Configuration

The figure 1 and figure 2 shows all branches using Cisco Unified SRST multicasting MOH.The 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.

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Multicast MOH Transmission

Figure 11: MOH Sources for Cisco Unified SRST and Other Unified SRST IP Phones Using MOH

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 the 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 the 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.

MOH from a Live Feed

MOH live feed is an SRST feature that provides MOH streams to IP phones from an audio device connected
to a local E&M (ISR G2) or FXO (ISR G2/G3) port, or from a remote gateway. Live audio is fed continuously
from a fixed source to the MOH playout buffer instead of being read from a flash file.
Live feed audio can also be streamed via multicast to compatible devices. For more information, see Configuring
Cisco Unified SRST for MOH from a Live Feed section.

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.

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How to Use Cisco Unified SRST Gateways as a Multicast MOH Resource

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.
The following table provides a summary of options for 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

Live feed The multicast audio stream has minimal delay for Configuring Cisco Unified SRST for
local IP phones. The MOH stream for PSTN callers MOH from a Live Feed
is delayed by a few seconds. If the live feed audio
input fails, callers on hold hear silence.

Live feed and The live feed stream has a few seconds of delay Configuring Cisco Unified SRST for
flash memory for both PSTN and local IP phone callers. The Multicast MOH from an Audio File
flash MOH acts as backup for the live-feed MoH.
and
We recommend this option if you want live-feed
Configuring Cisco Unified SRST for
because it provides guaranteed MOH if the
MOH from a Live Feed
live-feed input is not found or fails.

How to Use Cisco Unified SRST Gateways as a Multicast MOH

Resource
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.
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
The following sections describe the Cisco Unified CM configuration tasks for Cisco Unified SRST multicast
MOH:
• Configuring the MOH Audio Source to Enable Multicasting
• Enabling Multicast on the Cisco Unified Communications Manager MOH Server and Configuring Port
Numbers and IP Addresses
• Creating an MRG and an MRGL, Enabling MOH Multicast, and Configuring Gateways
• Creating a Region for the MOH Server
• Verifying Cisco Unified Communications Manager Multicast MOH

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Appendix B: Integrating Cisco Unified Communications Manager and Cisco Unified SRST to Use Cisco Unified SRST as a Multicast MOH Resource

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. The 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.
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.

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Configuring the MOH Audio Source to Enable Multicasting

Figure 12: Unified Communications Manager Screens for Configuring Multicast MOH

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.

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.

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Enabling Multicast on the Cisco Unified Communications Manager MOH Server and Configuring Port Numbers and IP Addresses

Note These instructions assume that an MOH audio source file was already created.

SUMMARY STEPS
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.
2. Double-click the required audio source listed in the MOH Audio Sources column.
3. In the MOH Audio Source Configuration window, check Allow Multicasting.
4. Click Update.

DETAILED STEPS

Command or Action Purpose

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 and the Multicast MOH
Transmission section), 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.

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Table 4: Example of the Differences Between Incrementing Multicast on IP Address and Incrementing Multicast on Port Number

Audio Source Codec Increment Multicast on IP Address Increment Multicast on Port

Number

Destination IP Destination Port Destination IP Destination Port

Address Address

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 intouse. 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.

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Use this procedure to enable multicast and configure port numbers and IP addresses.

SUMMARY STEPS
1. Enable multicast MOH for Cisco Unified CM
2. Set the base IP address and port number.
3. Select whether Cisco Unified CM increments port numbers or IP addresses.
4. Enter a maximum number of hops.
5. Use Cisco IOS commands to stop Cisco Unified CM signals from crossing the WAN and reaching Cisco
Unified SRST gateways.

DETAILED STEPS

Command or Action Purpose

Step 1 Enable multicast MOH for Cisco Unified CM
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 In the MOH Server Configuration window, in the Increment
or IP addresses. 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 settingbecause
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 If all of your branches run Cisco Unified SRST, skip this
from crossing the WAN and reaching Cisco Unified SRST step. If your system configuration includes routers that do
gateways. not run Cisco Unified SRST and use a different codec, enter

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Creating an MRG and an MRGL, Enabling MOH Multicast, and Configuring Gateways

Command or Action Purpose

the following Cisco IOS commands starting from global
configuration mode on the central site router:

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.
Use the following procedure to create an MRG and MRGL, to enable MOH multicast, and to configure
gateways.

SUMMARY STEPS
1. Create an MRG with a multicast MOH media resource.
2. Create an MRGL that contains the newly created MRG.
3. Add the MRGL to the required IP phones.
4. Add the MRGL to the required gateway.

DETAILED STEPS

Command or Action Purpose

Step 1 Create an MRG with a multicast MOH media resource.
Step 2 Create an MRGL that contains the newly created MRG.
Step 3 Add the MRGL to the required IP phones.
Step 4 Add the MRGL to the required gateway.

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. 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.

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Verifying Cisco Unified Communications Manager Multicast MOH

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.

SUMMARY STEPS
1. Create an MOH server region.
2. Create other regions as needed for different codecs.

DETAILED STEPS

Command or Action Purpose

Step 1 Create an MOH server region.
Step 2 Create other regions as needed for different codecs.

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.

SUMMARY STEPS
1. Verify that Cisco Unified CM system’s multicast MOH is heard on a remote gateway.
2. Verify that the Cisco Unified CM system’s MOH is multicast, not unicast.

DETAILED STEPS

Command or Action Purpose

Step 1 Verify that Cisco Unified CM system’s multicast MOH is
heard on a remote gateway.

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Configuring Cisco Unified SRST for Multicast MOH from an Audio File

Command or Action Purpose

Step 2 Verify that the Cisco Unified CM system’s MOH is
multicast, not unicast.

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.

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.

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Enabling Multicast MOH on the Cisco Unified SRST Gateway

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
8. call-manager-fallback
9. ip source-address ip-address [ port port
10. max-ephones max-phones
11. max-dn max-directory-number
12. moh filename
13. multicasting-enabled
14. multicast moh multicast-addressport port [ route ip-address-list ]
15. exit

DETAILED STEPS

Command or Action Purpose

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 theinterface
configuration mode.
Example:
Router(config)# interface loopback 1 number —Loopback interface number. The range is from
0 to 2147483647.

Step 3 ip address ip-address mask Sets a primary IP address for an interface.

Example: • ip-address—IP address.
Router(config-if)# ip address 10.1.1.1
255.255.255.255
• mask—Mask for the associated IP subnet.

Step 4 exit Exits interface configuration mode.

Example:
Router(config-if)# exit

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Command or Action Purpose

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
Example:
interface type and enters interface configuration mode.
Router(config)# interface fastethernet 0/0

Step 6 ip address ip-address mask (Optional if the route keyword is not used in the multicast
moh command. See Step 9 and Step 13.) Sets a primary
Example:
IP address for an interface.
Router(config-if)# ip-address 172.21.51.143
255.255.255.192

Step 7 exit (Optional if the route keyword is not used in the multicast
moh command. See Step 9 and Step 13.) Exits interface
Example:
configuration mode.
Router(config-if)# exit

Step 8 call-manager-fallback Enters call-manager-fallback configuration mode.

Example:
Router(config)# call-manager-fallback

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
Example:
messages from Cisco Unified IP phones through the
Router(config-cm-fallback)# ip source-address specified IP addresses and ports.
172.21.51.143 port 2000
• 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:
Router(config-cm-fallback)# max-dn 1 max-directory-number —Maximum number 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.

Example: filename—Filename of the music file. The music file must
Router(config-cm-fallback)# moh music-on-hold.au reside in flash memory.

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Verifying Basic Cisco Unified SRST Multicast MOH Streaming

Command or Action Purpose

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.

Step 14 multicast moh multicast-addressport port [ route Enables multicast of MOH from a branch office flash MOH
ip-address-list ] file to IP phones in the branch office.
Example: • multicast-addressport port —Declares the IP address
Router(config-cm-fallback)# multicast moh and port number of MOH packets that are to be
239.1.1.1 multicast. The multicast IP address and port must
port 16386 route 239.1.1.2 239.1.1.3 239.1.1.4 match the IP address and the port number that Cisco
239.1.1.5
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. 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.

Step 15 exit Exits call-manager-fallback configuration mode.

Example:
Router(config-cm-fallback)# exit

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

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Verifying Cisco Unified SRST MOH to PSTN

DETAILED STEPS

Command or Action Purpose

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
Example:
multicasting MOH out of Loopback 0 and Fast Ethernet
Router# debug ephone moh 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
Example:
fastethernet output displays as 50 packets/sec and 80 kbps
Router# show interfaces fastethernet 0/0 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
!

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. 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.
2. show ccm-manager music-on-hold
3. debug h245 asn
4. show call active voice

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DETAILED STEPS

Command or Action Purpose

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
Example:
or an earlier version.
Router# show ccm-manager music-on-hold
Current active multicast sessions : 1 Note that the show ccm-manager music-on-hold
Multicast RTP port Packets Call Codec Incoming command displays information about PSTN connections
Address number in/out id Interface
=======================================================
on hold only. It does not display information about multicast
239.1.1.1 16384 326/326 42 G.711ulaw streams going to IP phones on hold. The following is an
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.

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
Example:
command output to verify the H.323 handshaking between
Router# debug h245 asn Cisco Unified Communications Manager and the Cisco
*Mar 1 04:20:19.227: H245 MSC INCOMING PDU ::=
value MultimediaSystemControlMessage ::= response Unified SRST gateway. When a PSTN caller is placed on
: hold, Cisco Unified Communications Manager sends an
openLogicalChannelAck : H.245 closeLogicalChannel, followed by an
{ openLogicalChannel. Verify that the final
forwardLogicalChannelNumber 6
forwardMultiplexAckParameters openLogicalChannelAck from Cisco Unified
h2250LogicalChannelAckParameters : Communications Manager to the Cisco Unified SRST
{ gateway contains the expected multicast IP address and port
sessionID 1 number. In the following example, the IP address is
mediaChannel unicastAddress : iPAddress :
{ EF010101 (239.1.1.1) and the port number is 16384.
network 'EF010101'H
tsapIdentifier 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:
Communications Manager and the Cisco Unified SRST
Router# show call active voice | include gateway.
RemoteMedia

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Verifying Cisco Unified SRST Multicast MOH to IP Phones

Command or Action Purpose

RemoteMediaIPAddress=239.1.1.1 The IP address and port number displayed must match the
RemoteMediaPort=16384
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.

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.

SUMMARY STEPS
1. Verify that an IP phone caller hears MOH when placed on hold by an IP phone caller.
2. Check the MOHMulticastResourceActive and MOHUnicastResourceActive counters.

DETAILED STEPS

Command or Action Purpose

Step 1 Verify that an IP phone caller hears MOH when placed on Use an IP phone to call a second IP phone, and put the
hold by an IP phone caller. second caller on hold. The second caller should hear MOH.

Step 2 Check the MOHMulticastResourceActive and Use the Performance window to check the
MOHUnicastResourceActive counters. MOHMulticastResourceActive and
MOHUnicastResourceActive counters under the Cisco
MOH Device performance object. See Step 2 in the
Verifying Cisco Unified Communications Manager
Multicast MOH section. For Cisco Unified SRST
multicasting MOH to work, the multicast counter must
increment.

Troubleshooting Tips
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

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.

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Prerequisites

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 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.
The following sections describe the configuration tasks for Cisco Unified SRST MOH live feed:

Prerequisites
Cisco Unified SRST for multicast MOH, as described in the Configuring Cisco Unified SRST for Multicast
MOH from an Audio File section, 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.

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Setting Up the Voice Port on the Cisco Unified SRST Gateway

• 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.

SUMMARY STEPS
1. voice-port port
2. input gain decibels
3. auto-cut-through
4. operation 4-wire
5. signal immediate
6. no shutdown
7. exit

DETAILED STEPS

Command or Action Purpose

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
Example:
argument for your router, see Cisco IOS Survivable Remote
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
Example:
integers from –6 to 14.
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
Example:
you use this command with E&M ports.
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
Example:
command for E&M ports.
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
Example:
and to send address information as DTMF digits.
Router(config-voiceport)# signal immediate

Step 6 no shutdown Activates the voice port.

Example:
Router(config-voiceport)# no shutdown

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Setting Up the Directory Numbers on the Cisco Unified SRST Gateway

Command or Action Purpose

Step 7 exit Exits voice-port configuration mode.
Example:
Router(config-voiceport)# exit

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 tagpots
2. destination-pattern string
3. port port
4. exit

DETAILED STEPS

Command or Action Purpose

Step 1 dial-peer voice tagpots 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
Example:
E.164 telephone number to be used for a dial peer.
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
Example:
Unified SRST Gateway section.
Router(config-dial-peer)# port 1/1/0

Step 4 exit Exits dial-peer configuration mode.

Example:
Router(config-dial-peer)# exit

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
1. call-manager-fallback
2. max-dn max-directory-number

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Appendix B: Integrating Cisco Unified Communications Manager and Cisco Unified SRST to Use Cisco Unified SRST as a Multicast MOH Resource

3. multicast moh multicast-addressportport [ route ip-address-list ]

4. moh-live dn-number calling-numberout-calloutcall-number
5. exit

DETAILED STEPS

Command or Action Purpose

Step 1 call-manager-fallback Enters call-manager-fallback configuration mode.
Example:
Router(config)# call-manager-fallback

Step 2 max-dn max-directory-number Sets the maximum possible number of virtual voice ports
that can be supported by a router.
Example:
Router(config-cm-fallback)# max-dn 1 • max-directory-number—Maximum number of
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-addressportport [ route Enables multicast of MOH from a branch office flash MOH
ip-address-list ] file to IP phones in the branch office.
Example: Note This command must be used to source live feed
Router(config-cm-fallback)# multicast moh MOH to multicast Cisco Unified CM mode. It
239.1.1.1 port 16386 route 239.1.1.2 239.1.1.3 is not required in strict SRST mode.
239.1.1.4 239.1.1.5

Example: • multicast-address and port port —Declares the IP

Router(config-cm-fallback)# multicast moh
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. 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.

Step 4 moh-live dn-number Specifies that this telephone number is to be used for an
calling-numberout-calloutcall-number outgoing call that is to be the source for an MOH stream.

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Verifying Cisco Unified SRST MOH Live Feed

Command or Action Purpose

Example: • dn-number calling-number — Sets the MOH
Router(config-cm-fallback)# moh-live dn-number telephone number. The calling-number argument is a
3333 out-call 7777 sequence of digits that represent a telephone number.
• 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 the Setting
Up the Voice Port on the Cisco Unified SRST Gateway
section.

Step 5 exit Exits call-manager-fallback configuration mode.

Example:
Router(config-cm-fallback)# exit

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.

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

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

interface FastEthernet0/0
ip address 172.21.51.143 255.255.255.192

call-manager-fallback
ip source-address 172.21.51.143 port 2000
max-ephones 1
max-dn 1
moh music-on-hold.au
multicast moh 239.1.1.1 port 16384 route 172.21.51.143 10.1.1.1

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MOH Live Feed: Example

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 the Configuring
the MOH Audio Source to Enable Multicasting section. 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
!
dial-peer voice 7777 pots
destination-pattern 7777
port 1/0/0
!
!
call-manager-fallback
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
The Feature Information for Cisco Unified SRST as a Multicast MOH Resource table 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.
Use Cisco Feature Navigator to find information about platform support and software image support. 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 http://www.cisco.com/go/cfn. An
account on Cisco.com is not required.

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Where to Go Next

Note The Feature Information for Cisco Unified SRST as a Multicast MOH Resource table 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 5: Feature Information for Cisco Unified SRST as a Multicast MOH Resource

Feature Name Releases Feature Information

Cisco Unified SRST as a Multicast MOH Resource 3.0 The MOH-live feature was added.

Where to Go Next
For additional information, see the Related Documents section in the Cisco Unified SCCP and SIP SRST
Feature Overview, on page 41 chapter.

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Where to Go Next

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