Cisco CP-7911G-CH1 System Administrator Manual

Cisco Unified SCCP and SIP SRST 
System Administrator Guide 
(All Versions)



April 23, 2012

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Text Part Number: OL-13143-04

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

Cisco Unified SCCP and SIP SRST System Administrator Guide (All Versions)
© 2008-2012 Cisco Systems, Inc. All rights reserved.

.

IMPLIED, INCLUDING, WITHOUT



CONTENTS

Cisco Unified SCCP and SIP SRST Feature Overview 11

Contents 11

Cisco Unified SCCP SRST 11

Information About SCCP SRST 12
Prerequisites for Configuring Cisco Unified SCCP SRST 14
Restrictions for Configuring Cisco Unified SCCP SRST 17

Cisco Unified SIP SRST 19

Information About SIP SRST 19
Prerequisites for Configuring Cisco Unified SIP SRST 19
Restrictions for Configuring Cisco Unified SIP SRST 19

MGCP Gateways and SRST 23

Support for Cisco Unified IP Phones and Platforms 23

Finding Cisco IOS Software Releases That Support Cisco Unified SRST 23
Cisco Unified IP Phone Support 24
Platform and Memory Support 24
Cisco Unified Communications Manager Compatibility 24
Signal Support 25
Language Support 25
Switch Support 25

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Where to Go Next 25

Additional References 26

Related Documents 27
Standards 29
MIBs 29
RFCs 29
Technical Assistance 29

Obtaining Documentation, Obtaining Support, and Security Guidelines 29

Cisco Unified Survivable Remote Site Telephony Feature Roadmap 31

Contents 31

Documentation Organization 32

Feature Roadmap 33

Information About New Features in Cisco Unified SRST 38

New Features in Cisco Unified SRST Version 9.0 38

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Contents

New Features in Cisco Unified SRST Version 8.8 44
New Features in Cisco Unified SRST Version 8.0 45
New Features in Cisco Unified SRST Version 7.0/4.3 45
New Features in Cisco Unified SRST Version 4.2(1) 46
New Features in Cisco Unified SRST Version 4.1 46
New Features in Cisco Unified SRST Version 4.0 46
New Features in Cisco Unified SRST Version 3.4 48
New Features in Cisco SRST Version 3.3 48
New Features in Cisco SRST Version 3.2 49
New Features in Cisco SRST Version 3.1 52
New Features in Cisco SRST Version 3.0 53
New Features in Cisco SRST Version 2.1 57
New Features in Cisco SRST Version 2.02 59

Where to Go Next 61

Setting Up the Network 63

Contents 63

Information About Setting Up the Network 64

How to Set Up the Network 64

Enabling IP Routing 64
Enabling Cisco Unified SRST on an MGCP Gateway 64
Configuring DHCP for Cisco Unified SRST Phones 70
Specifying Keepalive Intervals 73

Where to Go Next 74

Cisco Unified SIP SRST 4.1 75

Contents 75

Prerequisites for Cisco Unified SIP SRST 4.1 75

Restrictions for Cisco Unified SIP SRST 4.1 76

Information About Cisco Unified SIP SRST 4.1 76

Out-of-Dialog REFER 76
Digit Collection on SIP Phones 77
Caller ID Display 78
Disabling SIP Supplementary Services for Call Forward and Call Transfer 78
Idle Prompt Status 78
Enhanced 911 Services 78

How to Configure Cisco Unified SIP SRST 4.1 Features 79

Enabling KPML for SIP Phones 79
Disabling SIP Supplementary Services for Call Forward and Call Transfer 81

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Configuring Idle Prompt Status for SIP Phones 82

Where to Go Next 83

Setting Up Cisco Unified IP Phones using SCCP 85

Contents 85

Information About Setting Up Cisco Unified IP Phones 85

How to Set Up Cisco Unified IP Phones 86

Configuring Cisco Unified SRST to Support Phone Functions 86
Configuring Cisco Unified 8941 and 8945 SCCP IP Phones 88
Verifying That Cisco Unified SRST Is Enabled 89
Configuring IP Phone Clock, Date, and Time Formats 90
Configuring IP Phone Language Display 92
Configuring Customized System Messages for Cisco Unified IP Phones 94
Configuring a Secondary Dial Tone 95
Configuring Dual-Line Phones 96
Configuring Eight Calls per Button (Octo-Line) 98
Configuring the Maximum Number of Calls 100
Troubleshooting 102

Contents

How to Set Up Cisco IP Communicator for Cisco Unified SRST 102

Verifying Cisco IP Communicator 103
Troubleshooting Cisco IP Communicator 103

Where to Go Next 103

Setting Up Cisco Unified IP Phones using SIP 105

Contents 105

Prerequisites for Configuring the SIP Registrar 105

Restrictions for Configuring the SIP Registrar 105

Information About Configuring the SIP Registrar 106

How to Configure the SIP Registrar 106

Configuring the SIP Registrar 106
Configuring Backup Registrar Service to SIP Phones 108
Configuring Backup Registrar Service to SIP Phones (Using Optional Commands) 112
Verifying SIP Registrar Configuration 115
Verifying Proxy Dial-Peer Configuration 117

Where to Go Next 120

Configuring Call Handling 123

Contents 123

Prerequisites for Configuring SIP SRST Features Using Back-to-Back User Agent Mode 123

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Contents

Restrictions for Configuring SIP SRST Features Using Back-to-Back User Agent Mode 124

Information About Configuring SCCP SRST Call Handling 124

H.323 VoIP Call Preservation Enhancements for WAN Link Failures 124
Toll Fraud Prevention 124

Information About Configuring SIP SRST Features Using Back-to-Back User Agent Mode 125

Cisco Unified SIP SRST and Cisco SIP Communications Manager Express Feature Crossover 125

How to Configure Cisco Unified SCCP SRST 127

Configuring Incoming Calls 127
Configuring Outgoing Calls 147

How to Configure Cisco Unified SIP SRST 164

Configuring SIP Phone Features 164
Configuring SIP-to-SIP Call Forwarding 166
Configuring Call Blocking Based on Time of Day, Day of Week, or Date 168
SIP Call Hold and Resume 172

How to Configure Optional Features 174

Enabling Three-Party G.711 Ad Hoc Conferencing 174
Defining XML API Schema 176

Where to Go Next 176

Configuring Secure SRST for SCCP and SIP 177

Contents 177

Prerequisites for Configuring Secure SRST 177

Restrictions for Configuring Secure SRST 178

Information About Configuring Secure SRST 179

Benefits of Secure SRST 179
Cisco IP Phones Clear-Text Fallback During Non-Secure SRST 179
Signaling Security on Unify SRST - TLS 180
Media Security on Unify SRST - SRTP 182
Establishment of Secure Cisco Unified SRST to the Cisco Unified IP Phone 182
Secure SRST Authentication and Encryption 184

How to Configure Secure Unified SRST 185

185

Preparing the Cisco Unified SRST Router for Secure Communication 186
Configuring Cisco Unified Communications Manager to the Secure Cisco Unified SRST Router 203
Enabling SRST Mode on the Secure Cisco Unified SRST Router 206
Configuring Secure SCCP SRST 207
Configuring Secure SIP Call Signaling and SRTP Media with Cisco SRST 221

Additional References 230

Related Documents 230

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Standards 231
MIBs 231
RFCs 231
Technical Assistance 231

Command Reference 232

Feature Information for Secure SCCP and SIP SRST 233

Where to Go Next 233

Integrating Voice Mail with Cisco Unified SRST 235

Contents 235

Information About Integrating Voice Mail with 
Cisco Unified SCCP SRST 235

How to Integrate Voice Mail with Cisco Unified SCCP and SIP SRST 237

Configuring Direct Access to Voice Mail 237
Configuring Message Buttons 240
Redirecting to Cisco Unified Communications Manager Gateway 243
Configuring Call Forwarding to Voice Mail 243
Configuring Message Waiting Indication 247

Contents

Configuration Examples for SCCP SRST 249

Configuring Local Voice-Mail System (FXO and FXS): Example 249
Configuring Central Location Voice-Mail System (FXO and FXS): Example 250
Configuring Voice-Mail Access over FXO and FXS: Example 251
Configuring Voice-Mail Access over BRI and PRI: Example 251

How to Configure DTMF Relay for SIP Applications and Voice Mail 252

DTMF Relay Using SIP RFC 2833 252
DTMF Relay Using SIP Notify (Nonstandard) 254

Where to Go Next 256

Setting Video Parameters 257

Contents 257

Prerequisites for Setting Video Parameters 257

Restrictions for Setting Video Parameters 258

Information About Setting Video Parameters 258

Matching Endpoint Capabilities 259
Retrieving Video Codec Information 259
Call Fallback to Audio-Only 259
Call Setup for Video Endpoints 259
Flow of the RTP Video Stream 260

How to Set Video Parameters for Cisco Unified SRST 261

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Contents

Configuring Slow Connect Procedures 261
Verifying Cisco Unified SRST 262
Setting Video Parameters for Cisco Unified SRST 268

Troubleshooting Video for Cisco Unified SRST 270

Where to Go Next 270

Monitoring and Maintaining Cisco Unified SRST 271

Appendix A: Configuring Cisco Unified SIP SRST Features Using Redirect Mode 273

Contents 273

Prerequisites for Cisco Unified SIP SRST Features Using Redirect Mode 273

Restrictions for Cisco Unified SIP SRST Features Using Redirect Mode 273

Information About Cisco Unified SIP SRST Features Using Redirect Mode 274

How to Configure Cisco Unified SIP SRST Features Using Redirect Mode 274

Configuring Call Redirect Enhancements to Support Calls Between SIP IP Phones for Cisco Unified SIP
SRST 274

Configuring Sending 300 Multiple Choice Support 277

Configuration Examples for Cisco Unified SIP SRST Features Using Redirect Mode 278

Cisco Unified SIP SRST: Example 279

Where to Go Next 280

Appendix B: Integrating Cisco Unified Communications Manager and Cisco Unified SRST to Use
Cisco Unified SRST as a Multicast MOH Resource 281

Contents 281

Prerequisites for Using Cisco Unified SRST Gateways as a Multicast MOH Resource 282

Restrictions for Using Cisco Unified SRST Gateways as a Multicast MOH Resource 282

Information About Using Cisco Unified SRST Gateways as a Multicast MOH Resource 283

Cisco Unified SRST Gateways and Cisco Unified Communications Manager 283
Codecs, Port Numbers, and IP Addresses 284
Multicast MOH Transmission 286
MOH from a Live Feed 286
MOH from Flash Files 287

How to Use Cisco Unified SRST Gateways as a Multicast MOH Resource 288

Configuring Cisco Unified Communications Manager for Cisco Unified SRST Multicast MOH 288
Configuring Cisco Unified SRST for Multicast MOH from an Audio File 296
Configuring Cisco Unified SRST for MOH from a Live Feed 306

Configurations Examples for Cisco Unified SRST Gateways 311

MOH Routed to Two IP Addresses: Example 311
MOH Live Feed: Example 312

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Feature Information for Cisco Unified SRST as a Multicast MOH Resource 312

Where to Go Next 313

Index

Contents

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Cisco Unified SCCP and SIP SRST System Administrator Guide

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Contents

Cisco Unified SCCP and SIP SRST System Administrator Guide

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Cisco Unified SCCP and SIP SRST Feature
Overview

Revised: February 3, 2011

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

Supported Firmware, Platforms, Memory, and Voice Products.

Cisco Unified SRST 4.0

Contents

• Cisco Unified SCCP SRST, page 11

• Cisco Unified SIP SRST, page 19

• MGCP Gateways and SRST, page 23

• Support for Cisco Unified IP Phones and Platforms, page 23

• Where to Go Next, page 25

• Additional References, page 26

• Obtaining Documentation, Obtaining Support, and Security Guidelines, page 29

Cisco Unified SCCP SRST

• Information About SCCP SRST, page 12

• Prerequisites for Configuring Cisco Unified SCCP SRST, page 14

• Restrictions for Configuring Cisco Unified SCCP SRST, page 17

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Cisco Unified SCCP and SIP SRST System Administrator Guide

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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.

Cisco Unified SCCP and SIP SRST Feature Overview

Note Cisco Unified CM fallback mode telephone service is available only to those Cisco Unified IP

phones that are supported by a Cisco Unified SRST router. Other Cisco Unified IP phones on the
network remain out of service until they re-establish a connection with their primary, secondary,
or tertiary Cisco Unified CM.

Typically, it takes three times the keepalive period for a phone to discover that its connection to Cisco
Unified CM has failed. The default keepalive period is 30 seconds. If the phone has an active standby
connection established with a Cisco Unified SRST router, the fallback process takes 10 to 20 seconds
after connection with Cisco Unified CM is lost. An active standby connection to a Cisco Unified SRST
router exists only if the phone has the location of a single Cisco Unified CM in its Unified
Communications Manager list. Otherwise, the phone activates a standby connection to its secondary
Cisco Unified CM.

Note The time it takes for a Cisco Unified IP Phone to fallback to the SRST router can vary depending

on the phone type. Phones such as the Cisco 7902, Cisco 7905, and Cisco 7912 can take
approximately 2.5 minutes to fallback to SRST mode.

If a Cisco Unified IP phone has multiple Cisco Unified CM in its Cisco Unified CM list, it progresses
through its list of secondary and tertiary Cisco Unified CM before attempting to connect with its local
Cisco Unified SRST router. Therefore, the time that passes before the Cisco Unified IP phone eventually
establishes a connection with the Cisco Unified SRST router increases with each attempt to contact to a
Cisco Unified CM. Assuming that each attempt to connect to a Cisco Unified CM takes about 1 minute,
the Cisco Unified IP phone in question could remain offline for 3 minutes or more following a WAN link
failure.

Cisco Unified SCCP and SIP SRST System Administrator Guide

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Cisco Unified SCCP and SIP SRST Feature Overview

Note During a WAN connection failure, when Cisco Unified SRST is enabled, Cisco Unified IP

phones display a message informing you that they are operating in Cisco Unified CM fallback
mode. For example, the Cisco Unified IP Phone 7960G and Cisco Unified IP Phone 7940G
display a "CM Fallback Service Operating" message, and the Cisco Unified IP Phone 7910
displays a "CM Fallback Service" message when operating in Cisco Unified CM fallback mode.
When the Cisco Unified CM is restored, the message goes away and full Cisco Unified IP phone
functionality is restored.

While in Cisco Unified CM fallback mode, Cisco Unified IP phones periodically attempt to re-establish
a connection with Cisco Unified CM at the central office. Generally, the default time that Cisco Unified
IP phones wait before attempting to re-establish a connection to a remote Cisco Unified CM is 120
seconds. The time can be changed in Cisco Unified CM; see the "Device Pool Configuration Settings"
chapter in the appropriate Cisco Unified CM Administration Guide. A manual reboot can immediately
reconnect Cisco Unified IP phones to Cisco Unified CM.

When a connection is re-established with Cisco Unified CM, Cisco Unified IP phones automatically
cancel their registration with the Cisco Unified SRST Router. However, if a WAN link is unstable, Cisco
Unified IP phones can bounce between Cisco Unified CM and Cisco Unified SRST. A Cisco Unified IP
phone cannot re-establish a connection with the primary Cisco Unified CM at the central office if it is
currently engaged in an active call.

Cisco Unified SCCP SRST



Cisco Unified SRST supports the following call combinations:

• SCCP phone to SCCP phone

• SCCP phone to PSTN/router voice-port

• SCCP phone to WAN VoIP using SIP or H.323

• SIP phone to SIP phone

• SIP phone to PSTN / router voice-port

• SIP phone to Skinny Client Control Protocol (SCCP) phone

• SIP phone to WAN VoIP using SIP

Figure 1 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
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

Unified IP phones use the

router and are able to make and receive off-net calls.








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Cisco Unified SCCP and SIP SRST System Administrator Guide

13

Cisco Unified SCCP SRST

IP IP IP

V

PSTN

IP

network

WAN

disconnected

Central

Cisco Unified

Communications

Manager

Cisco Unified SRST

router

Fax

Telephone Telephone

Cisco IP phones

PCs

146613

Cisco Unified SCCP and SIP SRST Feature Overview

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

Inactive Call Detection document.

Prerequisites for Configuring Cisco Unified SCCP SRST

Before configuring Cisco Unified SRST, you must do the following:

• You have an account on Cisco.com to download software.

To obtain an account on Cisco.com, go to www.cisco.com and click Register at the top of the screen.

• You have purchased a Cisco Unified SRST license.

–

To purchase a license, go to http://www.cisco.com/cgi-bin/tablebuild.pl/ip-key.

–

To activate cme-srst feature license, see the Activating CME-SRST Feature License document.

Media

• Choose an appropriate Cisco Unified SRST version. Each SRST version supports a specific set of

IP phones, memory requirements, features, and DNs. See the

section on page 24 and the “Restrictions for Configuring Cisco Unified SCCP SRST” section on
page 17.

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“Platform and Memory Support”

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Cisco Unified SCCP and SIP SRST Feature Overview

• Choose an appropriate phoneload. SRST only supports certain phoneloads that have been tested with

the various Cisco
see the



Unified Communications Manager versions. For the most up-to-date phoneloads,

Cisco Unified SRST 4.3 Supported Firmware, Platforms, Memory, and Voice Products.

• If you have Cisco Unified Communications Manager already installed, verify that your version of 

Cisco Unified Communications Manager is compatible with your Cisco Unified SRST release. See
the

“Cisco Unified Communications Manager Compatibility” section on page 24.

Installing Cisco Unified Communications Manager

When installing Cisco Unified Communications Manager, consider the following:

• See the installation instructions for your version in the Cisco Unified Communications Manager

Install and Upgrade Guides.

• Integrate Cisco Unified SRST with Cisco Unified Communications Manager. Integration is

performed from Cisco Unified Communications Manager. See the “Integrating Cisco Unified

SCCP SRST with Cisco Unified Communications Manager” section on page 16.

Cisco Unified SCCP SRST

Installing Cisco Unified SCCP SRST

Cisco Unified SRST versions have different installation instructions:

• Installing Cisco Unified SRST V3.0 and Later Versions, page 15

• Installing Cisco Unified SRST V2.0 and V2.1, page 15

• Installing Cisco Unified SRST V1.0, page 15

To update Cisco Unified SRST, follow the installation instructions described in this section.

Installing Cisco Unified SRST V3.0 and Later Versions

Install the Cisco IOS software release image containing the Cisco SRST or Cisco Unified SRST version
that is compatible with your Cisco

Communications Manager Compatibility” section on page 24. Cisco IOS software can be downloaded

from the Cisco Software Center at http://www.cisco.com/public/sw-center/.

Cisco SRST and Cisco Unified SRST can be configured to support continuous multicast output of music­on-hold (MOH) from a flash MOH file in flash memory. For more information, see the

API Schema” section on page 176. 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/.

Unified Communications Manager version. See the “Cisco Unified

“Defining XML

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

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Unified Communications Manager and Cisco Unified SRST versions.

Cisco Unified SCCP and SIP SRST System Administrator Guide

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Cisco Unified SCCP and SIP SRST Feature Overview

Cisco Unified SCCP SRST

Integrating Cisco Unified SCCP SRST with Cisco Unified Communications Manager

There are two procedures for integrating Cisco Unified SRST with Cisco Unified
Communications
Communications

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

Step 1 Create an SRST reference.

a. From any page in Cisco Unified Communications Manager, click System and SRST.

b. On the Find and List SRST References page, click Add a New SRST Reference.

c. On the SRST Reference Configuration page, enter a name in the SRST Reference Name field and

the IP address of the Cisco SRST router in the IP Address field.

d. Click Insert.

Manager. Procedure selection depends on the Cisco Unified
Manager version that you have.

Unified SRST Router.

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

a. From any page in Cisco Unified Communications Manager, click System and Device Pool.

b. On the Device Pool Configuration page, click on the required device pool icon.

c. On the Device Pool Configuration page, choose an SRST reference or “Use Default Gateway” from

the SRST Reference field’s menu.

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

If you have firmware versions that enable Cisco Unified SRST by default, no additional configuration is
required on Cisco
versions disable Cisco

Unified Communications Manager to support Cisco Unified SRST. If your firmware

Unified SRST by default, you must enable Cisco Unified SRST for each phone

configuration.

Step 1 Go to the Cisco Unified Communications Manager Phone Configuration page.

a. From any page in Cisco Unified Communications Manager, click Device and Phone.

b. In the Find and List Phones page, click Find.

c. After a list of phones appears, click on the required device name.

d. The Phone Configuration appears.

Step 2 In the Phone Configuration page, go to the Product Specific Configuration section at the end of the page,

choose Enabled from the Cisco

Step 3 Go to the Phone Configuration page for the next phone and choose Enabled from the Cisco Unified

Unified SRST field’s menu, and click Update.

SRST field’s menu by repeating Step 1 and Step 2.

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Cisco Unified SCCP and SIP SRST Feature Overview

Restrictions for Configuring Cisco Unified SCCP SRST

Table 1 provides a history of restrictions from Cisco SCCP SRST Version 1.0 to the present version of

Cisco Unified SCCP SRST.

Ta b l e 1 Restrictions from Cisco SCCP SRST from the Present Version to Version 1.0

Cisco Unified SCCP SRST

Cisco Unified SRST
Version

Cisco IOS
Release Restrictions

Ver si o n 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 SIP SRST Features” section on page 106.

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17

Cisco Unified SCCP and SIP SRST Feature Overview

Cisco Unified SCCP SRST

Table 1 Restrictions from Cisco SCCP SRST from the Present Version to Version 1.0(continued)

Cisco Unified SRST
Version

Ver si o n 4 . 0

Ver si o n 3 . 4

Ver si o n 3 . 2

Ver si o n 3 . 1

Ver si o n 3 . 0

Ver si o n 2 . 1

Version 2.02

Version 2.01

Ver si o n 2 . 0

Cisco IOS
Release Restrictions

12.4(4)XC

12.4(4)T

12.3(11)T

12.3(7)T

12.2(15)ZJ

12.3(4)T

• All of the restrictions in Cisco SRST Version 1.0.

• Caller-id display on supported Cisco Unified IP phones: SIP phones in fallback

mode displays the name and number of the caller. SCCP phones in fallback mode
display only the caller-id number assigned to the line; the caller-ID name
configuration for SCCP phones is not preserved during SRST fallback.

• Call transfer is supported only on the following:

–

VoIP H.323, VoFR, and VoATM between Cisco gateways running Cisco IOS
Release 12.2(11)T and using the H.323 nonstandard information element

12.2(15)T

12.2(13)T

12.2(11)T

12.2(8)T1

12.2(8)T

12.2(2)XT

–

FXO and FXS loop-start (analog)

–

FXO and FXS ground-start (analog)

–

Ear and mouth (E&M) (analog) and DID (analog)

–

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)

Ver si o n 1 . 0 12.2(2)XB

12.2(2)XG

12.1(5)YD

–

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.

• Does not support first generation Cisco Unified IP phones, such as Cisco IP Phone

30 VIP and Cisco IP Phone 12 SP+.

• 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.

Cisco Unified SCCP and SIP SRST System Administrator Guide

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Cisco Unified SCCP and SIP SRST Feature Overview

Cisco Unified SIP SRST

• Information About SIP SRST, page 19

• Prerequisites for Configuring Cisco Unified SIP SRST, page 19

• Restrictions for Configuring Cisco Unified SIP SRST, page 19

Information About SIP SRST

This guide describes Cisco Unified SRST functionality for SIP networks. Cisco Unified SIP SRST
provides backup to an external SIP proxy server 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

Cisco Unified SIP SRST

• 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 lose connectivity to the network core (because of a WAN outage), they may be unable to
make or receive calls. Cisco Unified SIP SRST functionality on a SIP PSTN gateway provides service
reliability for SIP-based IP phones in the event of a WAN outage. Cisco Unified SIP SRST enables the
SIP IP phones to continue to make and receive calls to and from the PSTN and also to make and receive
calls to and from other SIP IP phones.

To see a branch office Cisco Unifed IP Phones connected to a remote central Cisco Unified CM
Operating in SRST mode, see

Figure 1.

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.

Restrictions for Configuring Cisco Unified SIP SRST

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

Cisco Unified SIP SRST.

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19

Cisco Unified SCCP and SIP SRST Feature Overview

Cisco Unified SIP SRST

Ta b l e 2 Restrictions from Cisco SIP SRST from the Present Version to Version 3.0

Cisco Unified SRST
Version

Cisco IOS
Release Restrictions

Ver si o n 8 . 0 15.1(1)T • SIP phones may be configured on the Cisco Unified CM with an Authenticated

device security mode. The Cisco Unified CM ensures integrity and authentication
for the phone using a TLS connection with NULL-SHA cipher for signaling. If
such an Authenticated SIP phone fails over to the Cisco Unified SRST device, and
if the Cisco Unified CM and SRST device are configured to support secure SIP
SRST, it will register using TCP instead of TLS/TCP, thus disabling the
Authenticated mode until the phone fails back to the Cisco Unified CM.

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Cisco Unified SCCP and SIP SRST Feature Overview

Table 2 Restrictions from Cisco SIP SRST from the Present Version to Version 3.0(continued)

Cisco Unified SIP SRST

Cisco Unified SRST
Version

Cisco IOS
Release Restrictions

Ver si o n 4 . 1 12.4.(15)T • Cisco Unified SRST does not support BLF speed-dial notification, call forward all

synchronization, dial plans, directory services, or music-on-hold (MOH).

• Prior to SIP phone load 8.0, SIP phones maintained dual registration with both

Cisco

Unified Communications Manager and Cisco Unified SRST simultaneously.
In SIP phone load 8.0 and later versions, SIP phones use keepalive to maintain a
connection with Cisco
Cisco

Unified Communications Manager. Every two minutes, a SIP phone sends a
keepalive message to Cisco

Unified SRST during active registration with

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
are registered with Cisco

Unified Communications Manager while the phones

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

• 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.

Unified SRST
fallback mode. Therefore, if the PSAP calls back after the Cisco Unified IP Phones
register back to Cisco
Cisco

Emergency Responder will not have any history of those calls. As a result,

Unified Communications Manager,

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

• Using ELINs for multiple purposes can result in unexpected interactions with

• There are a number of other ways that your configuration of Enhanced 911 Services

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location identification numbers (ELINs) for each emergency response location
(ERL).

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.

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 SIP SRST Features” section on page 106.

Cisco Unified SCCP and SIP SRST System Administrator Guide

21

Cisco Unified SCCP and SIP SRST Feature Overview

Cisco Unified SIP SRST

Table 2 Restrictions from Cisco SIP SRST from the Present Version to Version 3.0(continued)

Cisco Unified SRST
Version

Ver si o n 4 . 0

Ver si o n 3 . 4

Ver si o n 3 . 2

Ver si o n 3 . 1

Ver si o n 3 . 0

Cisco IOS
Release Restrictions

12.4(4)XC

12.4(4)T

12.3(11)T

12.3(7)T

12.2(15)ZJ

12.3(4)T

Not Supported

• MOH is not supported for a call hold invoked from a SIP phone. A caller hears only

silence when placed on hold by a SIP phone.

• As of Cisco IOS Release 12.4(4)T, bridged call appearance, find-me, incoming call

screening, paging, SIP presence, call park, call pickup, and SIP location are not
supported.

• SIP-NAT is not supported.

• Cisco Unity Express is not supported.

• Transcoding is not supported.

Phone Features

• For call waiting to work on the Cisco ATA and Cisco IP Phone 7912 and Cisco

Unified IP Phone 7905G with a 1.0(2) build, the incoming call leg should be
configured with the G.711 codec.

Note Cisco Unified IP Phone 7905G, Cisco Unified IP Phone 7912G, and Cisco

Analog Telephone Adaptor (ATA) 186 are not capable of dual registration; thus
they are not supported and have limited functionality with Cisco Unified SIP
SRST.

General

• Call detail records (CDRs) are only supported by standard IOS RADIUS support;

CDRs are not supported otherwise.

• All calls must use the same codec, either G.729r8 or G.711.

• Calls that have been transferred cannot be transferred a second time.

• URL dialing is not supported. Only number dialing is supported.

• The SIP registrar functionality provided by Cisco Unified SIP SRST provides no

security or authentication services.

• SIP IP phones that do not support dual concurrent registration with both their

primary and their backup SIP proxy or registrar may be unable to receive incoming
calls from the Cisco Unified SIP SRST gateway during a WAN outage. These
phones may take a significant amount of time to discover that their primary SIP
proxy or registrar is unreachable before they initiate a fallback registration to their
backup proxy or registrar (the SIP SRST gateway).

• SIP-phone-to-SIP-trunk support requires Refer and 302/300 Redirection to be

supported by the SIP trunk (Version 3.0).

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Cisco Unified SCCP and SIP SRST Feature Overview

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 alternate commands. With
Cisco

IOS releases after 12.3(14)T, the ccm-manager fallback-mgcp and service commands must be
configured. A complete configuration for these commands is shown in the section the

Unified SRST on an MGCP Gateway” section on page 64.

Note The commands listed above are ineffective unless both commands are configured. For instance, your

configuration will not work if you only configure the ccm-manager fallback-mgcp command.

For more information on the fallback methods for MGCP gateways, see the Configuring MGCP Gateway

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

MGCP Gateways and SRST

“Enabling Cisco

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.

• Finding Cisco IOS Software Releases That Support Cisco Unified SRST, page 23

• Cisco Unified IP Phone Support, page 24

• Platform and Memory Support, page 24

• Cisco Unified Communications Manager Compatibility, page 24

• Signal Support, page 25

• Language Support, page 25

• Switch Support, page 25

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.

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23

Support for Cisco Unified IP Phones and Platforms

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

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

Data Sheet and Cisco VG248 Analog Phone Gateway Version 1.2(1) Release Notes.

Cisco Unified SCCP and SIP SRST Feature Overview

Parameters

Cisco VG248 Analog Phone Gateway

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
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
Cisco

Feature Navigator.

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

IOS software release, see the online release notes or, if supported,

Cisco Unified Communications Manager Compatibility

Cisco Feature Navigator. Cisco Feature

See Cisco Unified Communications Manager Compatibility Matrix.

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Cisco Unified SCCP and SIP SRST Feature Overview

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

Where to Go Next

• 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)

Where to Go Next

The next chapters of this book describe how to configure Cisco Unified SIP SRST. As shown in Tabl e 3,
each chapter takes you through tasks in the order in which they need to be performed. The first task for
configuring Cisco Unified SRST is to ensure that the basic software and hardware in your system are
configured correctly for Cisco Unified SRST.

Ta b l e 3 Cisco Unified SRST Configuration Sequence

Task Where Task Is Described

1. Setting up a Cisco Unified SRST system

to communicate with your network

2. Configuring Version 4.1 features Cisco Unified SIP SRST 4.1, page 75

3. Setting up the basic Cisco Unified SRST

phone configuration using SCCP

Setting Up the Network, page 63

Setting Up Cisco Unified IP Phones using SCCP,
page 85

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25

Additional References

Table 3 Cisco Unified SRST Configuration Sequence(continued)

Task Where Task Is Described

4. Providing a backup to an external SIP

proxy server by supplying basic registrar
services

5. Configuring incoming and outgoing calls Configuring Call Handling, page 123

6. Configuring optional security for SRST Configuring Secure SRST for SCCP and SIP, page 177

7. Setting up voice mail Integrating Voice Mail with Cisco Unified SRST,

8. Setting up video parameters Setting Video Parameters, page 257

9. Monitoring and maintaining Cisco

Unified Survivable Remote
Site

Telephony (SRST)

Additional References

Cisco Unified SCCP and SIP SRST Feature Overview

Setting Up Cisco Unified IP Phones using SIP,
page 105

page 235

Monitoring and Maintaining Cisco Unified SRST,
page 271

The following sections provide additional references related to Cisco Unified SIP SRST:

• Related Documents, page 27

• Standards, page 29

• MIBs, page 29

• RFCs, page 29

• Technical Assistance, page 29

Cisco Unified SCCP and SIP SRST System Administrator Guide

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Cisco Unified SCCP and SIP SRST Feature Overview

Related Documents

Related Topic Documents

Cisco IOS voice product configuration • Cisco IOS Voice Configuration Library

• Cisco IOS Voice Command Reference

• Cisco IOS Debug Command Reference

• Cisco IOS Tcl IVR and VoiceXML Application Guide

• Cisco IOS Survivable Remote Site Telephony Version 3.2 System

Administrator Guide

Configuring SRST and MGCP Fallback • Configuring MGCP Gateway Support for

Cisco Unified Communications Manager

• MGCP Gateway Fallback Transition to Default H.323 Session

Application

• Configuring SRS Telephony and MGCP Fallback

Cisco Unified Communications Manager user
documentation

• Cisco Unified Communications Manager

• Cisco Unified Communications Manager Security Guide

• Cisco Unified Communications Operating System

Administration Guide

Additional References



Cisco Unified IP Phones • Cisco 7900 Series Unified IP Phones End-User Guides

• Cisco IP Phone Authentication and Encryption for

Cisco Communications Manager

• Cisco Unified IP Phone 7970 Series Administration Guide for

Cisco Unified CallManager, Release 5.0 (for models 7970G and
7971G-GE) (SCCP), “Understanding Security Features for

Cisco IP Phones” section.

Cisco Unified SRST commands and specifications • Cisco Unified SRST and Cisco Unified SIP SRST Command

Reference (All Versions)

• Cisco Unified SRST 8.0 Supported Firmware, Platforms,

Memory, and Voice Products

• Cisco Unified SRST 4.3 Supported Firmware, Platforms,

Memory, and Voice Products

Cisco Security Documentation • Media and Signaling Authentication and Encryption Feature for

Cisco IOS MGCP Gateways

• Cisco IOS Certificate Server

• Manual Certificate Enrollment (TFTP and Cut-and-Paste)

• Certification Authority Interoperability Commands

• Certificate Enrollment Enhancements

Cisco SIP SRST V3.4: Cisco IOS SIP Survivable

• Cisco IOS SIP SRST Feature Roadmap

Remote Site Telephony Feature Roadmap

Cisco SIP functionality • Cisco IOS SIP Configuration Guide

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27

Cisco Unified SCCP and SIP SRST Feature Overview

Additional References

Related Topic Documents

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

Command Reference

Command reference information for voice and
telephony commands

• Cisco IOS Voice Command Reference

• Cisco IOS Debug Command Reference

DHCP • Cisco IOS DHCP Server

Media Inactive Call Detection • Media Inactive Call Detection

Phone documentation for Cisco Unified SRST • Cisco Unified IP Phones 7900 Series

• Survivable Remote Site Telephony

Standard Glossary • Cisco IOS Voice Configuration Library Glossary

Standard Preface • Cisco IOS Voice Configuration Library Preface

Cisco Unified SCCP and SIP SRST System Administrator Guide

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Cisco Unified SCCP and SIP SRST Feature Overview

Obtaining Documentation, Obtaining Support, and Security Guidelines

Standards

Standard Title

ITU X. 509 Version 3 Public-Key and Attribute Certificate Frameworks

MIBs

MIB MIBs Link

No new or modified MIBs are supported by this
feature, and support for existing MIBs has not been
modified by this feature.

To locate and download MIBs for selected platforms, Cisco IOS
releases, and feature sets, use Cisco
following URL:

http://www.cisco.com/go/mibs

MIB Locator found at the

RFCs

RFC Title

RFC 2246 The Transport Layer Security (TLS) Protocol Version 1.0

RFC 2543 SIP: Session Initiation Protocol

RFC 3261 SIP: Session Initiation Protocol

RFC 3711 The Secure Real-Time Transport Protocol (SRTP)

Technical Assistance

Description Link

The Cisco Technical Support & Documentation
website contains thousands of pages of searchable
technical content, including links to products,
technologies, solutions, technical tips, and tools.
Registered Cisco.com users can log in from this page to
access even more content.

http://www.cisco.com/techsupport

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

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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documents, see the monthly

Cisco Unified SCCP and SIP SRST System Administrator Guide

29

Obtaining Documentation, Obtaining Support, and Security Guidelines

Cisco Unified SCCP and SIP SRST Feature Overview

Cisco Unified SCCP and SIP SRST System Administrator Guide

30

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Cisco Unified Survivable Remote Site Telephony
Feature Roadmap

Revised: April 23, 2012

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
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.

http://www.cisco.com/go/fn. You must have an account on

Contents

• Documentation Organization, page 32

• Feature Roadmap, page 33

• Information About New Features in Cisco Unified SRST, page 38

• Where to Go Next, page 61

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Cisco Unified Survivable Remote Site Telephony Feature Roadmap

Documentation Organization

Documentation Organization

This document consists of the following chapters or appendixes as shown in Table 1.

Ta b l e 1 Cisco Unified SRST Configuration Sequence

Chapter or Appendix Description

Cisco Unified SCCP and SIP SRST
Feature Overview, page 11

Setting Up the Network, page 63 Describes how to set up a Cisco Unified SRST system to communicate with your

Cisco Unified SIP SRST 4.1,
page 75

Setting Up Cisco Unified IP Phones
using SCCP, page 85

Setting Up Cisco Unified IP Phones
using SIP, page 105

Configuring Call Handling,
page 123

Configuring Secure SRST for SCCP
and SIP, page 177

Integrating Voice Mail with Cisco
Unified SRST, page 235

Gives a brief description of Cisco Unified SRST and provides 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.

network.

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

Describes how to set up the basic Cisco Unified SRST phone configuration.

Describes features available in Version 3.0 that are also necessary for Version 3.4.
Features include instructions on how to provide a backup to an external SIP proxy
server 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.

Describes how to configure incoming and outgoing calls.

Describes the Secure SRST security functionality to the Cisco Unified SRST.

Describes how to set up voice mail.

Setting Video Parameters, page 257 Describes how to set up video parameters.

Monitoring and Maintaining Cisco
Unified SRST, page 271

Appendix A: Configuring Cisco
Unified SIP SRST Features Using
Redirect Mode, page 273

Appendix B: Integrating Cisco
Unified Communications Manager
and Cisco Unified SRST to Use
Cisco Unified SRST as a Multicast
MOH Resource, page 281

Cisco Unified SCCP and SIP SRST System Administrator Guide

32

Provides a list of useful show commands for monitoring and maintaining Cisco Unified
SRST.

Describes features using redirect mode, which applies to version 3.0 only.

Describes how to configure Cisco Unified CM and Cisco Unified SRST to enable
multicast music-on-hold (MOH).

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Cisco Unified Survivable Remote Site Telephony Feature Roadmap

Feature Roadmap

Feature Roadmap

Table 2 provides a feature history summary of Cisco Unified SRST features.

Ta b l e 2 Features by Cisco Unified SRST Software Version

Cisco Unified SRST Cisco IOS Release Enhancements or Modifications

Ver si o n 9 . 0 15.2(2)T • Support for Cisco Unified 6901 and 6911 SIP IP Phones, page 38

• Support for Cisco Unified 6921, 6941, 6945, and 6961 SIP IP Phones,

page 40

• Support for Cisco Unified 8941 and 8945 SIP IP Phones, page 41

• Multiple Calls Per Line, page 42

• Voice and Fax Support on Cisco ATA-187, page 43

Ver si o n 8 . 8 15.2(1)T Support for Cisco Unified 6945, 8941, and 8945 SCCP IP Phones, page 44

Ver si o n 8 . 6 15.1(4)M Support for Cisco Unified 8941 and 8945 SCCP IP Phones were introduced. For

more information, see

Phones, page 88.

Ver si o n 8 . 0 15.1(1)T Beginning with Cisco IP Phone firmware 8.5(3) and Cisco IOS Release 15.1(1)T,

Cisco SRST supports SIP signaling over UDP, TCP, and TLS connections,
providing both RTP and SRTP media connections based on the security settings
of the IP phone. For more information, see the following sections:

• Signaling Security on Unify SRST - TLS, page 180

• Media Security on Unify SRST - SRTP, page 182

Configuring Cisco Unified 8941 and 8945 SCCP IP

• Configuring Secure SIP Call Signaling and SRTP Media with Cisco SRST,

page 221

Version 7.0/4.3 See Cisco Feature

Navigator for

compatibility.

Version 4.2(1) See Cisco Feature

Navigator for

compatibility.

• Configuring Eight Calls per Button (Octo-Line), page 98

• Configuring Consultative Transfer, page 134

Enhanced 911 Services, page 78

The following new features are included:

• 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.

Ver si o n 4 . 1 12.4(15)T • Enabling KPML for SIP Phones, page 79

• Disabling SIP Supplementary Services for Call Forward and Call Transfer,

page 78

• Configuring Idle Prompt Status for SIP Phones, page 82

• Enhanced 911 Services, page 78

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Cisco Unified Survivable Remote Site Telephony Feature Roadmap

Feature Roadmap

Table 2 Features by Cisco Unified SRST Software Version (continued)

Cisco Unified SRST Cisco IOS Release Enhancements or Modifications

Ver si o n 4 . 0 12.4(4)XC • Cisco IP Communicator Support, page 47

• Fax Passthrough using SCCP and ATAs Support, page 47

• H.323 VoIP Call Preservation Enhancements for WAN Link Failures for

SCCP Phones, page 47

• Video Support, page 47

Ver si o n 3 . 4 12.4(4)T • Cisco SIP SRST 3.4, page 48

• Appendix A: Configuring Cisco Unified SIP SRST Features Using Redirect

Mode, page 273

• Configuring Call Handling, page 123 (see Back-to-Back User Agent Mode)

Ver si o n 3 . 3 • Secure SRST, page 48.

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

page 49

• Enhancement to the show ephone Command, page 49

Ver si o n 3 . 2 12.3(11)T • Enhancement to the alias Command, page 50

• Enhancement to the pickup Command, page 50

• Enhancement to the user-locale Command, page 50

• Increased the Number of Cisco Unified IP Phones Supported on the Cisco

3845, page 50

• MOH Live-Feed Support, page 50

• No Timeout for Call Preservation, page 51

• RFC 2833 DTMF Relay Support, page 51

• Translation Profile Support, page 51

Ver si o n 3 . 1 12.3(7)T • Cisco Unified IP Phone 7920 Support, page 52

• Cisco Unified IP Phone 7936 Support, page 52

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Table 2 Features by Cisco Unified SRST Software Version (continued)

Cisco Unified SRST Cisco IOS Release Enhancements or Modifications

Ver si o n 3 . 0 12.2(15)ZJ

12.3(4)T

• Additional Language Options for IP Phone Display, page 53

• Consultative Call Transfer and Forward Using H.450.2 and H.450.3 for SCCP

Phones, page 54

• Customized System Message for Cisco Unified IP Phones, page 54

• Dual-Line Mode, page 54

• E1 R2 Signaling Support, page 54

• European Date Formats, page 56

• Huntstop for Dual-Line Mode, page 56

• Music-on-Hold for Multicast from Flash Files, page 56

• Ringing Timeout Default, page 56

• Secondary Dial Tone, page 56

• Enhancement to the show ephone Command, page 56

• System Log Messages for Phone Registrations, page 57

Feature Roadmap

• Three-Party G.711 Ad Hoc Conferencing, page 57

• Support for Cisco VG248 Analog Phone Gateway 1.2(1) and Higher

Versions, page 57

Ver si o n 2 . 1 • Cisco Unified IP Phone 7902G Support, page 58

• Cisco Unified IP Phone 7912G Support, page 59

• Additional Language Options for IP Phone Display, page 58

• Cisco SRST Aggregation, page 58

• Cisco ATA 186 and ATA 188 Support, page 58

• Cisco Unified IP Phone 7905G Support, page 59

• Cisco Unified IP Phone Expansion Module 7914 Support, page 59

• Enhancement to the dialplan-pattern Command, page 59

Version 2.02 • Cisco Unified IP Phone Conference Station 7935 Support, page 60

• Increase in Directory Numbers, page 60

• Cisco Unity Voice Mail Integration Using In-Band DTMF Signaling Across

the PSTN and BRI/PRI, page 60

• 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

• Support was added for additional connected Cisco IP phones.

1750 was removed.

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• Support was added for additional directory numbers or virtual voice ports on

Cisco

IP phones.

Cisco Unified SCCP and SIP SRST System Administrator Guide

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Cisco Unified Survivable Remote Site Telephony Feature Roadmap

Feature Roadmap

Table 2 Features by Cisco Unified SRST Software Version (continued)

Cisco Unified SRST Cisco IOS Release Enhancements or Modifications

Ver si o n 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
Cisco

MC3810-V3 concentrators.

• Cisco Unified SRST was implemented on the Cisco 1750 and Cisco 1751

3725 and Cisco 3745 routers and the

routers.

• Huntstop support.

• Class of restriction (COR).

• Translation rule support.

• MOH and tone on hold.

• Distinctive ringing.

• Forward to a central voice mail 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

Table 2 Features by Cisco Unified SRST Software Version (continued)

Cisco Unified SRST Cisco IOS Release Enhancements or Modifications

Ver si o n 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

• Dimming of all Cisco Unified IP Phone function keys that are not supported

Unified Communications Manager failure.

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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Cisco Unified Survivable Remote Site Telephony Feature Roadmap

Information About New Features in Cisco Unified SRST

Information About New Features in Cisco Unified SRST

This section contains the following topics:

• New Features in Cisco Unified SRST Version 9.0, page 38

• New Features in Cisco Unified SRST Version 8.8, page 44

• New Features in Cisco Unified SRST Version 8.0, page 45

• New Features in Cisco Unified SRST Version 7.0/4.3, page 45

• New Features in Cisco Unified SRST Version 4.2(1), page 46

• New Features in Cisco Unified SRST Version 4.1, page 46

• New Features in Cisco Unified SRST Version 4.0, page 46

• New Features in Cisco Unified SRST Version 3.4, page 48

• New Features in Cisco SRST Version 3.3, page 48

• New Features in Cisco SRST Version 3.2, page 49

• New Features in Cisco SRST Version 3.1, page 52

• New Features in Cisco SRST Version 3.0, page 53

• New Features in Cisco SRST Version 2.1, page 57

• New Features in Cisco SRST Version 2.02, page 59

New Features in Cisco Unified SRST Version 9.0

Cisco Unified SRST 9.0 supports the following new Cisco Unified SIP IP phones:

• Cisco Unified 6901 and 6911 SIP IP Phones

• Cisco Unified 6921, 6941, 6945, and 6961 SIP IP Phones

• Cisco Unified 8941 and 8945 SIP IP Phones

Cisco Unified SRST 9.0 supports the following new features:

• Multiple Calls Per Line, page 42

• Voice and Fax Support on Cisco ATA-187, page 43

Support for Cisco Unified 6901 and 6911 SIP IP Phones

Table 3 lists all the features supported on the Cisco Unified 6901 and 6911 SIP IP Phones in Cisco

Unified SRST 9.0.

Ta b l e 3 Features Supported on the Cisco Unified 6901 and 6911 SIP IP Phones in

Cisco Unified SRST 9.0

Features 6901 6911

After Hour Not Supported Not Supported

Barge Not Supported Not Supported

Busy-Lamp-Field Monitoring Not Supported Not Supported

Cisco Unified SCCP and SIP SRST System Administrator Guide

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

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Cisco Unified Survivable Remote Site Telephony Feature Roadmap

Information About New Features in Cisco Unified SRST

Table 3 Features Supported on the Cisco Unified 6901 and 6911 SIP IP Phones in

Cisco Unified SRST 9.0 (continued)

Features 6901 6911

Button Layout Not Supported Not Supported

Call Forward All Softkey Not Supported Not Supported

Call Park Not Supported Not Supported

Call Transfer Supported Supported

cBarge Not Supported Not Supported

Directory Service Not Supported Not Supported

Extension Mobility Not Supported Not Supported

Group Pickup Not Supported Not Supported

Hold Supported Supported

Intercom Not Supported Not Supported

KEM Not Supported Not Supported

Meet-Me Conference Not Supported Not Supported

Mobility Not Supported Not Supported

Multicast MoH Not Supported Not Supported

Multicast Paging Not Supported Not Supported

MyPhoneApp Not Supported Not Supported

Pickup Not Supported Not Supported

Privacy Not Supported Not Supported

Programmable Line Key Not Supported Not Supported

Redial Supported Supported

Resume Supported Supported

Shared Lines Not Supported Not Supported

Software Ad-Hoc Conference Supported

1

Supported

1

Speakerphone Supported Supported

Speed Dial Not Supported Not Supported

Video Not Supported Not Supported

1. This feature is controlled by the phone.



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Cisco Unified Survivable Remote Site Telephony Feature Roadmap

Information About New Features in Cisco Unified SRST

Support for Cisco Unified 6921, 6941, 6945, and 6961 SIP IP Phones

Table 4 lists all the features supported on the Cisco Unified 6921, 6941 6945, and 6961 SIP IP Phones

in Cisco Unified SRST 9.0.

Ta b l e 4 Features Supported on the Cisco Unified 6921, 6941, 6945, and 6961 SIP IP Phones in

Cisco Unified SRST 9.0

Features 6921 6941 6945 6961

After Hour Not Supported Not Supported Not Supported Not Supported

Barge Not Supported Not Supported Not Supported Not Supported

Busy-Lamp-Field
Monitoring

Button Layout Not Supported Not Supported Not Supported Not Supported

Call Forward All
Softkey

Call Park Not Supported Not Supported Not Supported Not Supported

Call Transfer Supported Supported Supported Supported

cBarge Not Supported Not Supported Not Supported Not Supported

Directory Service Supported

Extension
Mobility

Group Pickup Not Supported Not Supported Not Supported Not Supported

Hold Supported Supported Supported Supported

Intercom Not Supported Not Supported Not Supported Not Supported

KEM Not Supported Not Supported Not Supported Not Supported

Meet-Me
Conference

Mobility Not Supported Not Supported Not Supported Not Supported

Multicast MoH Not Supported Not Supported Not Supported Not Supported

Multicast Paging Not Supported Not Supported Not Supported Not Supported

MyPhoneApp Not Supported Not Supported Not Supported Not Supported

Pickup Not Supported Not Supported Not Supported Not Supported

Privacy Not Supported Not Supported Not Supported Not Supported

Programmable
Line Key

Redial Supported Supported Supported Supported

Resume Supported Supported Supported Supported

Shared Lines Not Supported Not Supported Not Supported Not Supported

Software Ad-Hoc
Conference

Speakerphone Supported Supported Supported Supported

Not Supported Not Supported Not Supported Not Supported

Supported

1

1

Supported

Supported

1

1

Not Supported Not Supported Not Supported Not Supported

Not Supported Not Supported Not Supported Not Supported

Not Supported Not Supported Not Supported Not Supported

Supported

1

Supported

1

Supported

Supported

Supported

1

1

1

Supported

Supported

Supported

1

1

1

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Cisco Unified Survivable Remote Site Telephony Feature Roadmap

Table 4 Features Supported on the Cisco Unified 6921, 6941, 6945, and 6961 SIP IP Phones in

Cisco Unified SRST 9.0 (continued)

Features 6921 6941 6945 6961

Speed Dial Not Supported Not Supported Not Supported Not Supported

Video Not Supported Not Supported Not Supported Not Supported

1. This feature is controlled by the phone.

Support for Cisco Unified 8941 and 8945 SIP IP Phones

Table 5 lists all the features supported on the Cisco Unified 8941 and 8945 SIP IP Phones in Cisco

Unified SRST 9.0.

Information About New Features in Cisco Unified SRST

Ta b l e 5 Features Supported on the Cisco Unified 8941 and 8945 SIP IP Phones in

Cisco Unified SRST 9.0

Features 8941 8945

After Hour Not Supported Not Supported

Barge Not Supported Not Supported

Busy-Lamp-Field Monitoring Not Supported Not Supported

Button Layout Not Supported Not Supported

Call Forward All Softkey Not Supported Not Supported

Call Park Not Supported Not Supported

Call Transfer Supported Supported

cBarge Not Supported Not Supported

Directory Service Not Supported Not Supported

Extension Mobility Not Supported Not Supported

Group Pickup Not Supported Not Supported

Hold Supported Supported

Intercom Not Supported Not Supported

KEM Not Supported Not Supported

Meet-Me Conference Not Supported Not Supported

Mobility Not Supported Not Supported

Multicast MoH Not Supported Not Supported

Multicast Paging Not Supported Not Supported

MyPhoneApp Not Supported Not Supported

Pickup Not Supported Not Supported

Privacy Not Supported Not Supported

Programmable Line Key Not Supported Not Supported

Redial Supported Supported

Resume Supported Supported



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Information About New Features in Cisco Unified SRST

Cisco Unified Survivable Remote Site Telephony Feature Roadmap

Table 5 Features Supported on the Cisco Unified 8941 and 8945 SIP IP Phones in

Features 8941 8945

Shared Lines Not Supported Not Supported

Software Ad-Hoc Conference Supported

Speakerphone Supported Supported

Speed Dial Not Supported Not Supported

Video Supported Supported

1. This feature is controlled by the phone.

Multiple Calls Per Line

Cisco Unified SRST 9.0 provides support for the Multiple Calls Per Line (MCPL) feature on Cisco
Unified 6921, 6941, 6945, and 6961 SIP IP phones and Cisco Unified 8941 and 8945 SCCP and SIP IP
phones.

Before Cisco Unified SRST 9.0, the maximum number of calls supported for every directory number
(DN) on Cisco Unified 8941 and 8945 SCCP IP phones was restricted to two.

With Cisco Unified SRST 9.0, the MCPL feature overcomes the limitation on the maximum number of
calls per line.

In Cisco Unified SRST 9.0, the MCPL feature is not supported on Cisco Unified 6921, 6941, 6945, and
6961 SCCP IP phones. The maximum number of calls allowed on these phones is two and the maximum
number of calls allowed on octo-line directory numbers on these phones before activating Call Forward
Busy or a busy tone is one.

Cisco Unified SRST 9.0 (continued)

1

Supported



1

Cisco Unified 8941 and 8945 SCCP IP Phones

Before Cisco Unified SRST 9.0, the values for the max-dn and timeouts busy commands were
hardcoded for Cisco Unified 8941 and 8945 SCCP IP phones.

In Cisco Unified SRST 9.0, you can configure the max-dn and timeouts busy commands in
call-manager-fallback configuration mode. Use the max-dn command to set the maximum number of
DNs that can be supported by the router and enable dual-line mode, octo-line mode, or both modes. Use
the timeouts busy command to set the timeout value for call transfers to busy destinations.

For configuration information, see the “Configuring the Maximum Number of Calls” section on

page 100.

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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Cisco Unified Survivable Remote Site Telephony Feature Roadmap

Voice and Fax Support on Cisco ATA-187

Cisco ATA-187 is a SIP-based analog telephone adaptor that turns traditional telephone devices into IP
devices. Cisco ATA-187 can connect with a regular analog FXS phone or fax machine on one end, while
the other end is an IP side that uses SIP for signaling and registers as a Cisco Unfiied 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.

Ta b l e 6 Features Supported on Cisco ATA-187 in Cisco Unified SRST 9.0

Features ATA-187

Ad-Hoc Conference Not Supported

Barge Not Supported

Call Forward All Supported

Call Transfer Supported

Call Waiting Supported

cBarge Not Supported

Hold Supported

Meet-Me Conference Not Supported

Pickup Supported

Redial Supported

Resume Supported

Shared Lines Not Supported

Speed Dial Not Supported

Voice Mail Supported

Information About New Features in Cisco Unified SRST

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For more information on Cisco ATA-187, see Cisco ATA 187 Analog Telephone Adaptor Administration

Guide for SIP.

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43

Cisco Unified Survivable Remote Site Telephony Feature Roadmap

Information About New Features in Cisco Unified SRST

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

Table 7 lists the features supported on Cisco Unified 6945, 8941, and 8945 SCCP IP Phones in Cisco

Unified SRST.

Ta b l e 7 Features Supported on the Cisco Unified 6945, 8941, and 8945 SCCP IP Phones in

Cisco Unified SRST

Features 6945 8941 8945

After Hours Supported Supported Supported

Button Layout Not Supported Not Supported Not Supported

Call Forward Supported Supported Supported

Call Park Not Supported Not Supported Not Supported

Call Transfer Supported Supported Supported

Call Transfer Recall Not Supported Not Supported Not Supported

cBarge Not Supported Not Supported Not Supported

Conferencing

1

Directory Services Supported Supported Supported

Enhanced Busy-Lamp-Field
Monitoring

Extension Mobility Not Supported Not Supported Not Supported

Forced Authorization Code Not Supported Not Supported Not Supported

Hold Supported Supported Supported

Intercom Not Supported Not Supported Not Supported

Live Record Not Supported Not Supported Not Supported

Multicast MOH Supported Supported Supported

Multicast Paging Not Supported Not Supported Not Supported

My Phone Apps Not Supported Not Supported Not Supported

Night Service Not Supported Not Supported Not Supported

Privacy Not Supported Not Supported Not Supported

Programmable Line Keys Not Supported Not Supported Not Supported

Resume Supported Supported Supported

Secure Real-time Transport
Protocol

Supported Supported Supported

Not Supported Not Supported Not Supported

Supported Supported Supported

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Cisco Unified Survivable Remote Site Telephony Feature Roadmap

Table 7 Features Supported on the Cisco Unified 6945, 8941, and 8945 SCCP IP Phones in

Cisco Unified SRST (continued)

Features 6945 8941 8945

Shared Lines Supported Supported Supported

Single Number Reach Not Supported Not Supported Not Supported

Speakerphone Supported Supported Supported

Speed Dial Supported Supported Supported

Transfer to Voicemail Not Supported Not Supported Not Supported

Video Telephony Not Supported Not Supported Not Supported

Whisper Intercom Not Supported Not Supported Not Supported

1. Only software conferencing is supported.

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).

Information About New Features in Cisco Unified SRST

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 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), page 98

• Configuring Consultative Transfer, page 134

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Cisco Unified Survivable Remote Site Telephony Feature Roadmap

Information About New Features in Cisco Unified SRST

New Features in Cisco Unified SRST Version 4.2(1)

Cisco Unified SRST Version 4.2(1) introduces the following new features:

• Enhancements for Enhanced 911 Services, page 78

New Features in Cisco Unified SRST Version 4.1

Cisco Unified SRST Version 4.1 introduces the following new feature:

• Enhanced 911 Services, page 78

New Features in Cisco Unified SRST Version 4.0

Cisco Unified SRST Version 4.0 has introduced the following new features:

• Additional Cisco Unified IP Phone Support, page 46

• Cisco IP Communicator Support, page 47

• Fax Passthrough using SCCP and ATAs Support, page 47

• H.323 VoIP Call Preservation Enhancements for WAN Link Failures for SCCP Phones, page 47

• Video Support, page 47

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

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

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

Voice Products.

Cisco IP

command in Cisco

Cisco Unified SRST 4.3 Supported Firmware, Platforms, Memory, and

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Cisco Unified Survivable Remote Site Telephony Feature Roadmap

Cisco IP Communicator Support

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

Fax Passthrough using SCCP and ATAs Support

Fax passthrough 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
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
and Defaults” chapter in

Guide for SCCP.

Cisco ATA 186 and Cisco ATA 188 Analog Telephone Adaptor Administrator's

Information About New Features in Cisco Unified SRST

information, see the “Parameters

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
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.

Unified Communications Manager,

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
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
audio calls.

For more information, see the “Setting Video Parameters” section on page 257.

Unified IP Phones do

Unified SRST

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Cisco Unified Survivable Remote Site Telephony Feature Roadmap

Information About New Features in Cisco Unified SRST

New Features in Cisco Unified SRST Version 3.4

Cisco SRST V3.4 introduced the new features described in the following section:

• Cisco SIP SRST 3.4, page 48

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 proxy server by providing
basic registrar and back-to-back user agent (B2BUA) services. These services are used by a SIP IP phone
in the event of a WAN connection outage when the SIP phone is unable to communicate with its primary
SIP proxy.

Cisco SIP SRST Version 3.4 can support SIP phones with standard RFC 3261 feature support locally and
across SIP WAN networks. With Cisco SIP SRST Version 3.4, SIP phones can place calls across SIP
networks in the same way as Skinny Client Control Protocol (SCCP) phones. For full information about
SIP SRST, Version 3.4, see

Cisco SIP SRST Version 3.4 System Administrator Guide.

New Features in Cisco SRST Version 3.3

Cisco SRST V3.3 introduced the new features described in the following sections:

• Secure SRST, page 48

• Cisco Unified IP Phone 7970G and Cisco Unified 7971G-GE Support, page 49

• Enhancement to the show ephone Command, page 49

Secure SRST

Secure Cisco IP phones that are located at remote sites and that are attached to gateway routers can
communicate securely with Cisco
link or Cisco
phones becomes nonsecure. To overcome this situation, gateway routers can now function in secure
SRST mode, which activates when the WAN link or Cisco
down. When the WAN link or Cisco
Communications Manager resumes secure call-handling capabilities.

Secure SRST provides new SRST security features such as authentication, integrity, and media
encryption. Authentication provides assurance to one party that another party is whom it claims to be.
Integrity provides assurance that the given data has not been altered between the entities. Encryption
implies confidentiality; that is, that no one can read the data except the intended recipient. These security
features allow privacy for SRST voice calls and protect against voice security violations and identity
theft. For more information see the

Unified Communications Manager goes down, all communication through the remote

Unified Communications Manager using the WAN. But if the WAN

Unified Communications Manager is restored, Cisco Unified

“Configuring Secure SRST for SCCP and SIP” section on page 177.

Unified Communications Manager goes

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Information About New Features in Cisco Unified SRST

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
customizeable 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
for more information.

“Cisco Unified IP Phone Expansion Module 7914 Support” section on page 59

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

ephone

command in Cisco Unified SRST and Cisco Unified SIP SRST Command Reference (All

Unified IP Phone 7971G-GE. For more information, see the show

Ve rs i o ns ) .

New Features in Cisco SRST Version 3.2

Cisco SRST V3.2 introduced the new features described in the following sections:

• Enhancement to the alias Command, page 50

• Enhancement to the cor Command, page 50

• Enhancement to the pickup Command, page 50

• Enhancement to the user-locale Command, page 50

• Increased the Number of Cisco Unified IP Phones Supported on the Cisco 3845, page 50

• MOH Live-Feed Support, page 50

• No Timeout for Call Preservation, page 51

• RFC 2833 DTMF Relay Support, page 51

• Translation Profile Support, page 51

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Information About New Features in Cisco Unified SRST

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

• 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).

Cisco Unified Survivable Remote Site Telephony Feature Roadmap

Unified Communications Manager fallback was increased to 50.

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

For more information, see the user-locale command in the

Cisco Unified SRST and Cisco Unified SIP SRST Command Reference (All Versions).

Unified Communications Manager V4.0 or later 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
SRST mode. If an FXO port is used for a live feed, the port must be supplied with an external third-party
adapter 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

Communications Manager and Cisco Unified SRST to Use Cisco Unified SRST as a Multicast MOH
Resource” section on page 281 for configuration instructions.

IP phones. See the “Appendix B: Integrating Cisco Unified

IP phones in

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No Timeout for Call Preservation

To preserve existing H.323 calls on the branch in the event of an outage, disable the H.225 keepalive
timer by entering the no h225 timeout keepalive command. This feature is supported in Cisco IOS
Releases 12.3(7)T1 and higher versions. See the

section on page 11 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 Voice Mail” section on page 252 for

configuration instructions.

To use voice mail 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
Versions 3.0 and 3.1.

Information About New Features in Cisco Unified SRST

“Cisco Unified SCCP and SIP SRST Feature Overview”

SRST

Translation Profile Support

Cisco SRST 3.2 and later versions support translation profiles. Translation profiles allow you to group
translation rules together and to associate translation rules with the following:

• Called numbers

• Calling numbers

• Redirected called numbers

See the “Enabling Translation Profiles” section on page 140 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).



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Information About New Features in Cisco Unified SRST

New Features in Cisco SRST Version 3.1

Cisco SRST V3.1 introduced the new features described in the following sections:

• Cisco Unified IP Phone 7920 Support, page 52

• Cisco Unified IP Phone 7936 Support, page 52

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

documentation.

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.

Cisco Unified Survivable Remote Site Telephony Feature Roadmap

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.

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New Features in Cisco SRST Version 3.0

Cisco SRST V3.0 introduced the new features described in the following sections:

• Additional Language Options for IP Phone Display, page 53

• Consultative Call Transfer and Forward Using H.450.2 and H.450.3 for SCCP Phones, page 54

• Customized System Message for Cisco Unified IP Phones, page 54

• Dual-Line Mode, page 54

• E1 R2 Signaling Support, page 54

• European Date Formats, page 56

• Huntstop for Dual-Line Mode, page 56

• Music-on-Hold for Multicast from Flash Files, page 56

• Ringing Timeout Default, page 56

• Secondary Dial Tone, page 56

• Enhancement to the show ephone Command, page 56

• System Log Messages for Phone Registrations, page 57

Information About New Features in Cisco Unified SRST

• Three-Party G.711 Ad Hoc Conferencing, page 57

• Support for Cisco VG248 Analog Phone Gateway 1.2(1) and Higher Versions, page 57

Additional Language Options for IP Phone Display

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

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

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Information About New Features in Cisco Unified SRST

Consultative Call Transfer and Forward Using H.450.2 and H.450.3 for SCCP Phones

Cisco SRST V1.0, Cisco SRST V2.0, and Cisco SRST V2.1 allow blind call transfers and blind call
forwarding. Blind calls do not give transferring and forwarding parties the ability to announce or consult
with destination parties. These three versions of Cisco SRST use a Cisco SRST proprietary mechanism
to perform blind transfers. Cisco SRST V3.0 adds the ability to perform call transfers with consultation
using the ITU-T H.450.2 (H.450.2) standard and call forwarding using the ITU-T
standard for H.323 calls.

Cisco SRST V3.0 provides support for IP phones to initiate call transfer and forwarding with H.450.2
and H.450.3 by using the default session application. The built-in H.450.2 and H.450.3 support that is
provided by the default session application applies to call transfers and call forwarding initiated by IP
phones, regardless of the PSTN interface type.

Note All voice gateway routers in the VoIP network must support H.450. For H.450 support, routers with

Cisco SRST must run either Cisco SRST V3.0 and higher versions or Cisco IOS Release 12.2(15)ZJ and
later releases. Routers without Cisco SRST must run either Cisco SRST V2.1 and higher versions or
Cisco IOS Release 12.2(11)YT and later releases. SIP phones does not support this feature.

H.450.3 (H.450.3)

For more information about the default session application, see the Default Session Application

Enhancements document.

For configuration information, see the “Enabling Consultative Call Transfer and Forward Using H.450.2

and H.450.3 with Cisco SRST 3.0” section on page 148.

Customized System Message for Cisco Unified IP Phones

The display message that appears on Cisco Unified IP Phone 7905G, Cisco Unified IP Phone 7940G,
Cisco Unified IP Phone 7960G, and Cisco Unified IP Phone 7910 units when they are in fallback mode
can be customized. The new system message command allows you to edit these display messages on a
per-router basis. The custom system message feature supports English only.

For further information, see the “Configuring Customized System Messages for Cisco Unified IP

Phones” section on page 94.

Dual-Line Mode

A new keyword that was added to the max-dn command allows you to set IP phones to dual-line mode.
Each dual-line IP phone must have one voice port and two channels to handle two independent calls.
This mode enables call waiting, call transfer, and conference functions on a single ephone-dn (ephone
directory number). There is a maximum number of DNs available during Cisco SRST fallback. The

max-dn

For configuration information, see the “Configuring Dual-Line Phones” section on page 96.

command affects all IP phones on a Cisco SRST router.

E1 R2 Signaling Support

Cisco SRST V3.0 supports E1 R2 signaling. R2 signaling is an international signaling standard that is
common to channelized E1 networks; however, there is no single signaling standard for R2. The ITU-T
Q.400-Q.490 recommendation defines R2, but a number of countries and geographic regions implement
R2 in entirely different ways. Cisco Systems addresses this challenge by supporting many localized
implementations of R2 signaling in its Cisco IOS software.

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The Cisco Systems E1 R2 signaling default is ITU, which supports the following countries: Denmark,
Finland, Germany, Russia (ITU variant), Hong Kong (ITU variant), and South Africa (ITU variant). The
expression “ITU variant” means there are multiple R2 signaling types in the specified country, but Cisco
supports the ITU variant.

Cisco Systems also supports specific local variants of E1 R2 signaling in the following regions,
countries, and corporations:

• Argentina

• Australia

• Bolivia

• Brazil

• Bulgaria

• China

• Colombia

• Costa Rica

• East Europe (includes Croatia, Russia, and Slovak Republic)

Information About New Features in Cisco Unified SRST

• 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

• Vie t nam

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Information About New Features in Cisco Unified SRST

European Date Formats

The date format on Cisco IP phone displays can be configured with the following two additional formats:

• yy-mm-dd (year-month-day)

• yy-dd-mm (year-day-month)

For configuration information, see the “Configuring IP Phone Clock, Date, and Time Formats” section

on page 90.

Huntstop for Dual-Line Mode

A new keyword was added to the huntstop command. The channel keyword causes hunting to skip the
secondary channel in dual-line configuration if the primary line is busy or does not answer.

For configuration information, see the “Configuring Dial-Peer and Channel Hunting” section on

page 144.

Music-on-Hold for Multicast from Flash Files

Cisco Unified Survivable Remote Site Telephony Feature Roadmap

Cisco SRST can be configured to support continuous multicast output of MOH from a flash MOH file
in flash memory.

For more information, see the “Defining XML API Schema” section on page 176.

Ringing Timeout Default

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

“Configuring the Ringing Timeout Default” section on page 146.

Secondary Dial Tone

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

The secondary dial tone is created through the secondary dialtone command. For more information, see
the

“Configuring a Secondary Dial Tone” section on page 95.

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).

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Information About New Features in Cisco Unified SRST

System Log Messages for Phone Registrations

Diagnostic messages are added to the system log whenever a phone registers or unregisters from Cisco
Unified

SRST.

Three-Party G.711 Ad Hoc Conferencing

Cisco SRST supports three-party ad hoc conferencing using the G.711 coding technique. For
conferencing to be available, an IP phone must have a minimum of two lines connected to one or more
buttons.

For more information, see the “Enabling Three-Party G.711 Ad Hoc Conferencing” section on page 174.

Support for Cisco VG248 Analog Phone Gateway 1.2(1) and Higher Versions

The Cisco VG248 Analog Phone Gateway is a mixed-environment solution, enabled by Cisco AVVID
(Architecture for Voice, Video and Integrated Data), that allows organizations to support their legacy
analog devices while taking advantage of the new opportunities afforded through the use of IP telephony.
The Cisco VG248 is a high-density gateway for using analog phones, fax machines, modems, voice-mail
systems, and speakerphones within an enterprise voice system based on Cisco Unified CM.

During Cisco Unified CM fallback, Cisco SRST considers the Cisco VG248 to be a group of Cisco
Unified IP Phones. Cisco Unified SRST counts each of the 48 ports on the Cisco VG248 as a separate
Cisco Unified IP Phone. Support for Cisco VG248 Version 1.2(1) and higher versions is also available
in Cisco Unified SRST Version 2.1.

For more information, see Cisco VG248 Analog Phone Gateway Data Sheet and 

Cisco VG248 Analog Phone Gateway Version 1.2(1) Release Notes.

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, page 58

• Cisco SRST Aggregation, page 58

• Cisco ATA 186 and ATA 188 Support, page 58

• Cisco Unified IP Phone 7902G Support, page 58

• Cisco Unified IP Phone 7905G Support, page 59

• Cisco Unified IP Phone 7912G Support, page 59

• Cisco Unified IP Phone Expansion Module 7914 Support, page 59

• Enhancement to the dialplan-pattern Command, page 59

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

documentation.

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Information About New Features in Cisco Unified SRST

Additional Language Options for IP Phone Display

Displays for the Cisco Unified IP Phone 7940G and Cisco Unified IP Phone 7960G can be configured
with ISO-3166 codes for the following countries:

• France

• Germany

• Italy

• Portugal

• Spain

• United States

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

For configuration information, see the “Configuring IP Phone Language Display” section on page 92.

Cisco Unified Survivable Remote Site Telephony Feature Roadmap

Cisco SRST Aggregation

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

Cisco ATA 186 and ATA 188 Support

The Cisco ATA analog telephone adaptors are handset-to-Ethernet adaptors that allow regular analog
telephones to operate on IP-based telephony networks. Cisco ATAs support two voice ports, each with
an independent telephone number. The Cisco ATA 188 also has an RJ-45 10/100BASE-T data port.
Cisco

SRST supports Cisco ATA 186 and Cisco ATA 188 using Skinny Client Control Protocol (SCCP)
for voice calls only.

Cisco Unified IP Phone 7902G Support

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

The Cisco Unified IP Phone 7902G is a single-line IP phone with fixed feature keys that provide
one-touch access to the redial, transfer, conference, and voice-mail access features. Consistent with other
Cisco

IP phones, the Cisco Unified IP Phone 7902G supports inline power, which allows the phone to
receive power over the LAN. This capability gives the network administrator centralized power control
and thus greater network availability.

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

No configuration is necessary.

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 telephone traffic. Four dynamic soft keys
provide access to call features and functions. The graphic display shows calling information and allows
access to features.

The Cisco Unified IP Phone 7912G supports an integrated Ethernet switch, providing LAN connectivity
to a local PC. In addition, the Cisco Unified IP Phone 7912G supports inline power, which allows the
phone to receive power over the LAN. This capability gives the network administrator centralized power
control and thus greater network availability. The combination of inline power and Ethernet switch
support reduces cabling needs to a single wire to the desktop.

Information About New Features in Cisco Unified SRST

Cisco Unified IP Phone Expansion Module 7914 Support

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

Enhancement to the dialplan-pattern Command

A new keyword was added to the dialplan-pattern command. The extension-pattern keyword sets an
extension number’s leading digit pattern when it is different from the E.164 telephone number’s leading
digits defined in the pattern variable. This enhancement allows manipulation of IP
extension number prefix digits. See the dialplan-pattern command in

Unified SIP SRST Command Reference (All Versions).

New Features in Cisco SRST Version 2.02

Cisco SRST Version 2.02 introduced the new features described in the following sections:

• Cisco Unified IP Phone Conference Station 7935 Support, page 60

• Increase in Directory Numbers, page 60

• Cisco Unity Voice Mail Integration Using In-Band DTMF Signaling Across the PSTN and BRI/PRI,

page 60

phone abbreviated

Cisco Unified SRST and Cisco

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Cisco Unified Survivable Remote Site Telephony Feature Roadmap

Information About New Features in Cisco Unified SRST

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
no further administration is necessary. The Cisco
Cisco Unified CM for connection services and receives the appropriate endpoint phone number and any
software enhancements or personalized settings, which are preloaded within Cisco Unified CM.

The Cisco Unified IP Phone 7935 provides three soft keys and menu navigation keys that guide a user
through call features and functions. The Cisco Unified IP Phone 7935 also features a pixel-based LCD
display. The display provides features such as date and time, calling party name, calling party number,
digits dialed, and feature and line status. No configuration is necessary.

Increase in Directory Numbers

Table 8 shows the increases in directory numbers.

Ta b l e 8 Increases in Directory Numbers in Cisco IOS Release 12.2(11)T

7935 to an Ethernet switch port,

7935 dynamically registers to 

Increase in Maximum Directory Number

Cisco Router Maximum Phones

From To

Cisco 1751 24 96 120

Cisco 1760 24 96 120

Cisco 2600XM 24 96 120

Cisco 2691 72 216 288

Cisco 3640 72 216 288

Cisco 3660 240 720 960

Cisco 3725 144 432 576

Cisco 3745 240 720 960

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

Cisco Unity Voice Mail and other voice-mail systems can be integrated with Cisco SRST. Voice-mail
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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Where to Go Next

Proceed to the “Setting Up the Network” section on page 63.

Where to Go Next

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Where to Go Next

Cisco Unified Survivable Remote Site Telephony Feature Roadmap

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Contents

Setting Up the Network

Revised: February 3, 2011

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
Manager fallback.

Unified Communications

• Information About Setting Up the Network, page 64

• How to Set Up the Network, page 64

• Where to Go Next, page 74

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Information About Setting Up the Network

Information About Setting Up the Network

When the WAN link fails, the Cisco Unified IP Phones detect that they are no longer receiving keepalive
packets from Cisco Unified CM. The Cisco Unified IP Phones then register with the router. The Cisco
Unified SRST software is automatically activated and builds a local database of all Cisco Unified IP
Phones attached to it (up to its configured maximum). The IP phones are configured to query the router
as a backup call-processing source when the central Cisco Unified CM does not acknowledge keepalive
packets. The Cisco Unified SRST router now performs call setup and processing, call maintenance, and
call termination.

Cisco Unified Communications Manager uses DHCP to provide Cisco Unified IP Phones with the IP
address of Cisco
typically provided either by the SRST router itself or through the Cisco Unified SRST router using
DHCP relay. Configuring DHCP is one of two main tasks in setting up network communication. The
other task is configuring the Cisco Unified SRST router to receive messages from the Cisco
through the specified IP addresses. Keepalive intervals are also set at this time.

Unified Communications Manager. In a remote branch office, DHCP service is

How to Set Up the Network

Setting Up the Network

IP phones

This section contains the following tasks:

• Enabling IP Routing, page 64 (Required)

• Enabling Cisco Unified SRST on an MGCP Gateway (Required)

• Configuring DHCP for Cisco Unified SRST Phones, page 70 (Required)

• Specifying Keepalive Intervals, page 73 (Optional)

Enabling IP Routing

To initiate SRST service, you need to enable IP routing command and configure an interface that you
want to use or bind. For information about enabling IP routing, see

Enabling Cisco Unified SRST on an MGCP Gateway

To use SRST as your fallback mode with an MGCP gateway, SRST and MGCP fallback must both be
configured on the same gateway. The configuration below allows SRST to assume control over the voice
port and over call processing on the MGCP gateway. Due to command changes that were made in
Cisco

IOS Release 12.3(14)T, use the configuration task that corresponds with the Cisco IOS Release
you have installed.

Note The commands described in the configuration below are ineffective unless both commands are

configured. For instance, your configuration will not work if you only configure the ccm-manager
fallback-mgcp command.

Configuring IP Addressing.

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How to Set Up the Network

Note When an MGCP-controlled PRI goes into SRST mode, do not make or save configuration changes to the

NVRAM on the router. If configuration changes are made and saved in SRST mode, the
MGCP-controlled PRI fails when normal MGCP operation is restored.

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

Perform this task to enable SRST on a MGCP Gateway if you are using a software release prior to
Cisco

IOS Release 12.3(14)T.

SUMMARY STEPS

1. enable

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

Example:

Router> enable

Step 2

configure terminal

Example:

Router# configure terminal

Step 3

ccm-manager fallback-mgcp

Example:

Router(config)# ccm-manager fallback-mgcp

Enables privileged EXEC mode.

• Enter your password when prompted.

Enters global configuration mode.

Enables the gateway fallback feature and allows an MGCP
voice gateway to provide call processing services through
SRST or other configured applications when
Cisco

Unified Communications Manager is unavailable.

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How to Set Up the Network

Command or Action Purpose

Step 4

call application alternate [application-name]
or

service [alternate | default] service-name
location

Example:

Router(config)# call application alternate
or
Router(config)# service default

Setting Up the Network

The call application alternate command specifies that
the default voice application takes over if the MGCP
application is not available. The application-name
argument is optional and indicates the name of the specific
voice application to use if the application in the dial peer
fails. If a specific application name is not entered, the
gateway uses the DEFAULT application.

Or

The service command loads and configures a specific,
standalone application on a dial peer. The keywords and
arguments are as follows:

• alternate (Optional). Alternate service to use if the

service that is configured on the dial peer fails.

• default (Optional). Specifies that the default service

(“DEFAULT”) on the dial peer is used if the alternate
service fails.

• service-name: Name that identifies the voice

application.

• location: Directory and filename of the Tcl script or

VoiceXML document in URL format. For example,
flash memory (flash:filename), a TFTP
(tftp://../filename), or an HTTP server
(http://../filename) are valid locations.

Step 5

exit

Exits global configuration mode and returns to privileged
EXEC mode.

Example:

Router(config)# exit

Configuring SRST on an MGCP Gateway Using Cisco IOS Release 12.3(14)T or Later Releases

Perform this task to enable SRST on an MGCP Gateway if you are using Cisco IOS Release 12.3(14)T
or later version.

Restrictions

Effective with Cisco IOS Release 12.3(14)T, the call application alternate command is replaced by the
service command. The service command can be used in all releases after Cisco IOS Release 12.3(14)T.

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Setting Up the Network

SUMMARY STEPS

DETAILED STEPS

Command or Action Purpose

Step 1

enable

Example:

Router> enable

Step 2

configure terminal

How to Set Up the Network

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

Enables privileged EXEC mode.

• Enter your password when prompted.

Enters global configuration mode.

Step 3

Step 4

Step 5

Example:

Router# configure terminal

ccm-manager fallback-mgcp

Example:

Router(config)# ccm-manager fallback-mgcp

application [application-name]

Example:

Router(config) application app-xfer

global

Example:

Router(config)# global

Enables the gateway fallback feature and allows an MGCP
voice gateway to provide call processing services through
SRST or other configured applications when
Cisco

Unified Communications Manager is unavailable.

The application-name argument is optional and indicates
the name of the specific voice application to use if the
application in the dial peer fails. If a specific application
name is not entered, the gateway uses the DEFAULT
application.

Enters global configuration mode.

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How to Set Up the Network

Command or Action Purpose

Step 6

service [alternate | default] service-name
location

Example:

Router(config) service myapp
https://myserver/myfile.vxml

Step 7

exit

Setting Up the Network

Loads and configures a specific, standalone application on
a dial peer.

• alternate (Optional). Alternate service to use if the

service that is configured on the dial peer fails.

• default (Optional). Specifies that the default service

(“DEFAULT”) on the dial peer is used if the alternate
service fails.

• service-name: Name that identifies the voice

application.

• location: Directory and filename of the Tcl script or

VoiceXML document in URL format. For example,
flash memory (flash:filename), a TFTP
(tftp://../filename), or an HTTP server
(http://../filename) are valid locations.

Exits global configuration mode and returns to privileged
EXEC mode.

Example:

Router(config)# exit

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

The following is an example of configuring SRST on an MGCP Gateway if you are using Cisco IOS
Release 12.3(14)T or later release:

isdn switch-type primary-net5
!
! 
ccm-manager fallback-mgcp
ccm-manager mgcp
ccm-manager config
mta receive maximum-recipients 0
!
controller E1 1/0
pri-group timeslots 1-12,16 service mgcp
!
controller E1 1/1
! 



! 
! 
interface Ethernet0/0
ip address 10.48.80.9 255.255.255.0
half-duplex
!
interface Serial1/0:15
no ip address
no logging event link-status
isdn switch-type primary-net5
isdn incoming-voice voice
isdn bind-l3 ccm-manager
no cdp enable
! 


! 

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!
call rsvp-sync
!
call application alternate DEFAULT



!--- For Cisco IOS® Software Release 12.3(14)T or later,
this command was replaced by the service command
in global application configuration mode.
application
global
service alternate Default




!
voice-port 1/0:15
!
mgcp
mgcp dtmf-relay voip codec all mode cisco
mgcp package-capability rtp-package
mgcp sdp simple
!
mgcp profile default
!
!
!
dial-peer cor custom
!
!
!
dial-peer voice 10 pots
application mgcpapp
incoming called-number
destination-pattern 9T
direct-inward-dial
port 1/0:15



!
!
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

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How to Set Up the Network

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
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
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
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
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:

Cisco IOS DHCP Server document and see your Cisco Unified

Communications Manager TFTP server address provided by the

Setting Up the Network

IP phone and

Unified IP Phones may expire. This may

• Defining a Single DHCP IP Address Pool, page 70:Use this method if the Cisco Unified SRST

router is a DHCP server and if you can use a single shared address pool for all your DHCP clients.

• Defining a Separate DHCP IP Address Pool for Each Cisco Unified IP Phone, page 71: Use this

method if the Cisco Unified SRST router is a DHCP server and you need separate pools for
non-IP-phone DHCP clients.

• Defining the DHCP Relay Server, page 72: Use this method if the Cisco Unified SRST router is not

a DHCP server and you want to relay DHCP requests from IP phones to a DHCP server on a different
router.

Defining a Single DHCP IP Address Pool

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

SUMMARY STEPS

1. ip dhcp pool pool-name

2. network ip-address [mask | prefix-length]

3. option 150 ip ip-address

4. default-router ip-address

5. exit

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DETAILED STEPS

Command or Action Purpose

Step 1

ip dhcp pool pool-name

Example:

Router(config)# ip dhcp pool mypool

Step 2

network ip-address [mask | prefix-length]

Example:

Router(config-dhcp)# network 10.0.0.0 255.255.0.0

Step 3

option 150 ip ip-address

Example:

Router(config-dhcp)# option 150 ip 10.0.22.1

Step 4

default-router ip-address

Example:

Router(config-dhcp)# default-router 10.0.0.1

Step 5

exit

How to Set Up the Network

Creates a name for the DHCP server address pool
and enters DHCP pool configuration mode.

Specifies the IP address of the DHCP address pool
and the optional mask or number of bits in the
address prefix, preceded by a forward slash.

Specifies the TFTP server address from which the
Cisco

IP phone downloads the image configuration
file. This needs to be the IP address of Cisco Unified
CM.

Specifies the router to which the 
Cisco Unified IP phones are connected directly.

• 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.

Exits DHCP pool configuration mode.

Example:

Router(config-dhcp)# exit

Defining a Separate DHCP IP Address Pool for Each Cisco Unified IP Phone

This task creates a name for the DHCP server address pool and specifies IP addresses. This method
requires you to make an entry for every Cisco Unified IP phone.

SUMMARY STEPS

1. ip dhcp pool pool-name

2. host ip-address subnet-mask

3. option 150 ip ip-address

4. default-router ip-address

5. exit

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How to Set Up the Network

DETAILED STEPS

Command or Action Purpose

Step 1

ip dhcp pool pool-name

Example:

Router(config)# ip dhcp pool pool2

Step 2

host ip-address subnet-mask

Example:

Router(config-dhcp)# host 10.0.0.0 255.255.0.0

Step 3

option 150 ip ip-address

Example:

Router(config-dhcp)# option 150 ip 10.0.22.1

Step 4

default-router ip-address

Example:

Router(config-dhcp)# default-router 10.0.0.1

Step 5

exit

Setting Up the Network

Creates a name for the DHCP server address pool
and enters DHCP pool configuration mode.

Specifies the IP address that you want the phone to
use.

Specifies the TFTP server address from which the
Cisco

IP phone downloads the image

configuration file. This needs to be the IP address
of Cisco Unified CM.

Specifies the router to which the Cisco
Unified

• This router should be the Cisco Unified SRST

IP phones are connected directly.

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.

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

The Cisco IOS DHCP server feature is enabled on routers by default. If the DHCP server is not enabled
on your Cisco Unified SRST router, use the following steps to enable it.

SUMMARY STEPS

1. service dhcp

2. interface type number

3. ip helper-address ip-address

4. exit

Cisco IOS DHCP Server document.

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Setting Up the Network

DETAILED STEPS

Command or Action Purpose

Step 1

service dhcp

Example:

Router(config)# service dhcp

Step 2

interface type number

Example:

Router(config)# interface serial 0

Step 3

ip helper-address ip-address

Example:

Router(config-if)# ip helper-address 10.0.22.1

Step 4

exit

How to Set Up the Network

Enables the Cisco IOS DHCP Server feature on
the router.

Enters interface configuration mode for the
specified interface. See

Cisco IOS Interface and

Hardware Component Command Reference,
Release 12.3T for more information.

Specifies the helper address for any unrecognized
broadcast for TFTP server and Domain Name
System (DNS) requests. For each server, a
separate ip helper-address command is required
if the servers are on different hosts. You can also
configure multiple TFTP server targets by using
the ip helper-address command for multiple
servers.

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

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Where to Go Next

DETAILED STEPS

Command or Action Purpose

Step 1

Step 2

Step 3

call-manager-fallback

Example:

Router(config)# call-manager-fallback

keepalive seconds

Example:

Router(config-cm-fallback)# keepalive 60

exit

Example:

Router(config-cm-fallback)# exit

Setting Up the Network

Enters call-manager-fallback configuration mode.

Sets the time interval, in seconds, between keepalive
messages that are sent to the router by Cisco Unified
Phones.

• seconds: Range is 10 to 65535. Default is 30.

Exits call-manager-fallback configuration mode.

IP

Examples

The following example sets a keepalive interval of 45 seconds:

call-manager-fallback
keepalive 45

Where to Go Next

The next step is setting up the phone and getting a dial tone. For instructions, see the

“Cisco Unified SIP SRST 4.1” section on page 75.

For additional information, see the “Additional References” section on page 26 in the “Cisco Unified

SCCP and SIP SRST Feature Overview” chapter.

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Cisco Unified SIP SRST 4.1

Revised: February 3, 2011

This chapter describes the features and provides the configuration information for Cisco Unified
SIP

SRST 4.1:

• Out-of-Dialog REFER(OOD-R)

• Digit Collection on SIP Phones

• Caller ID Display

• Disabling SIP Supplementary Services for Call Forward and Call Transfer

• Idle Prompt Status

Note With Cisco IOS Release 12.4(15)T, the number of SIP phones supported on each platform is now

equivalent to the number of SCCP phones supported. For example, 3845 now supports 720 phones
regardless of whether these are SIP or SCCP.

Contents

• Prerequisites for Cisco Unified SIP SRST 4.1, page 75

• Restrictions for Cisco Unified SIP SRST 4.1, page 76

• Information About Cisco Unified SIP SRST 4.1, page 76

• How to Configure Cisco Unified SIP SRST 4.1 Features, page 79

• Where to Go Next, page 83

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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Restrictions for Cisco Unified SIP SRST 4.1

Restrictions for Cisco Unified SIP SRST 4.1

• Cisco Unified SRST does not support BLF speed-dial notification, call forward all synchronization,

dial plans, directory services, or music-on-hold (MOH).

• Prior to SIP phone load 8.0, SIP phones maintained dual registration with both

Cisco

Unified Communications Manager and Cisco Unified SRST simultaneously. In SIP phone
load 8.0 and later versions, SIP phones use keepalive to maintain a connection with
Cisco

Unified SRST during active registration with Cisco Unified Communications Manager.
Every 2 minutes, a SIP phone sends a keepalive message to Cisco
Cisco

Unified SRST responds to this keepalive with a 404 message. This process repeats until
fallback to Cisco
two minutes to Cisco
Cisco

Unified SRST. Cisco Unified SRST continues to support dual registration for SIP phone loads
older than 8.0.

Unified SRST occurs. After fallback, SIP phones send a keepalive message every

Unified Communications Manager while the phones are registered with

Information About Cisco Unified SIP SRST 4.1

Cisco Unified SIP SRST 4.1

Unified SRST.

• Out-of-Dialog REFER, page 76

• Digit Collection on SIP Phones, page 77

• Caller ID Display, page 78

• Disabling SIP Supplementary Services for Call Forward and Call Transfer, page 78

• Idle Prompt Status, page 78

• Enhanced 911 Services, page 78

Out-of-Dialog REFER

Out-of-dialog REFER (OOD-R) enables remote applications to establish calls by sending a REFER
message to Cisco Unified SRST without an initial INVITE. After the REFER is sent, the remainder of
the call setup is independent of the application and the media stream does not flow through the
application. The application using OOD-R triggers a call setup request that specifies the Referee address
in the Request-URI and the Refer-Target in the Refer-To header. The SIP messaging used to
communicate with Cisco Unified SRST is independent of the end-user device protocol, which can be
H.323, plain old telephone service (POTS), SCCP, or SIP. Click-to-dial is an example of an application
that can be created using OOD-R.

A click-to-dial application enables users to combine multiple steps into one click for a call setup. For
example, a user can click a web-based directory application from his or her PC to look up a telephone
number, off-hook the desktop phone, and dial the called number. The application initiates the call setup
without the user having to out-dial from his or her own phone. The directory application sends a REFER
message to Cisco Unified SRST, which sets up the call between both parties based on this REFER.

For more information about OOD-R, see Out-of-Dialog REFER from the Cisco Unified Communications

Manager Express System Administrator Guide.

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Cisco Unified SIP SRST 4.1

Digit Collection on SIP Phones

Digit strings dialed by phone users must be collected and matched against predefined patterns to place
calls to the destination corresponding to the user's input. Previously, SIP phones in a
Cisco

Unified SRST system required users to press the DIAL soft key or # key, or wait for the

interdigit-timeout to trigger call processing. This could cause delays in processing the call.

Two new methods of collecting and matching digits are supported for SIP phones depending on the
model of the phone:

• KPML Digit Collection, page 77

• SIP Dial Plans, page 77

KPML Digit Collection

The Key Press Markup Language (KPML) uses SIP SUBSCRIBE and NOTIFY methods to report user
input digit by digit. Each digit dialed by the phone user generates its own signaling message to
Cisco

Unified SRST, which performs pattern recognition by matching a destination pattern to a dial peer
as it collects the dialed digits. This process of relaying each digit immediately is similar to the process
used by SCCP phones. It eliminates the need for the user to press the Dial soft key or wait for the
interdigit timeout before the digits are sent to the Cisco

KPML is supported on Cisco Unified IP Phones 7911G, 7941G, 7941GE, 7961G, 7961GE, 7970G, and
7971GE. For configuration information, see the

Information About Cisco Unified SIP SRST 4.1

Unified SRST for processing.

“Enabling KPML for SIP Phones” section on page 79.

SIP Dial Plans

A dial plan is a set of dial patterns that SIP phones use to determine when digit collection is complete
after a user goes off-hook and dials a destination number. Dial plans enable SIP phones to perform local
digit collection and recognize dial patterns as user input is collected. After a pattern is recognized, the
SIP phone sends an INVITE message to Cisco
the user's input. All of the digits entered by the user are presented as a block to Cisco
processing. Because digit collection is done by the phone, dial plans reduce signaling messages overhead
compared to KPML digit collection.

SIP dial plans eliminate the need for a user to press the Dial soft key or # key or to wait for the interdigit
timeout to trigger an outgoing INVITE. You configure a SIP dial plan and associate the dial plan with a
SIP phone. The dial plan is downloaded to the phone in the configuration file.

You can configure SIP dial plans and associate them with the following SIP phones:

• Cisco Unified IP Phone 7911G, 7941G, 7941GE, 7961G, 7961GE, 7970G, and 7971GE: These

phones use dial plans and support KPML. If both a dial plan and KPML are enabled, the dial plan
has priority.

If a matching dial plan is not found and KPML is disabled, the user must wait for the interdigit
timeout before the SIP NOTIFY message is sent to Cisco
these phones do not have a Dial soft key to indicate the end of dialing, except when on-hook dialing
is used.

• Cisco Unified IP Phone 7905, 7912, 7940, and 7960: These phones use dial plans and do not support

KPML. If you do not configure a SIP dial plan for these phones, or if the dialed digits do not match
a dial plan, the user must press the Dial soft key or wait for the interdigit timeout before digits are
sent to Cisco

Unified SRST for processing.

Unified SRST to initiate the call to the number matching

Unified SRST for

Unified SRST. Unlike other SIP phones,

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Information About Cisco Unified SIP SRST 4.1

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

The Cisco Unified SRST supports SIP dial plans if they are provisioned in
Cisco

Unified Communications Manager. You cannot configure dial plans in Cisco Unified SRST.

Caller ID Display

The name and number of the caller is included in the Caller ID display on the 
Cisco Unified IP Phone 7911G, 7941G, 7941GE, 7961G, 7961GE, 7970G, and 7971GE. Other SIP
phones display only the number of the caller. Also, the caller ID information is updated on the
destination phone when there is a change in the caller ID of the originating party such as with call
forwarding or call transfer. No new configuration is required to support these enhancements.

Cisco Unified SIP SRST 4.1

plan is a separate XML file that is pointed to from the normal configuration file.

Disabling SIP Supplementary Services for Call Forward and Call Transfer

If a destination gateway does not support supplementary services, you can disable REFER messages for
call transfers and redirect responses for call forwarding from being sent by Cisco

Disabling supplementary services is supported if all endpoints use SCCP or all endpoints use SIP. It is
not supported for a mix of SCCP and SIP endpoints.

Unified SRST.

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
Manager. This message informs the user that the phone is operating in fallback mode and that not all
features are available. The default message that displays “CM Fallback Service Operating” is taken from
the phone dictionary file. You can customize the message by using the system message command on the
Cisco

Unified SRST router. Cisco Unified SRST updates the idle prompt message when a SIP phone
registers or when you modify the message through the configuration. The message displays until a phone
switches back to the Cisco

The idle prompt status message is supported for the Cisco Unified IP Phone 7911G, 7941G, 7941GE,
7961G, 7961GE, 7970G, and 7971GE with Cisco
earlier than Cisco

Unified SRST 4.1, the phones display the default message from the dictionary file.

Unified Communications Manager.

Unified SRST 4.1 and later versions. For versions

Unified Communications

Enhanced 911 Services

Enhanced 911 Services for Cisco Unified SRST enables 911 operators to:

• Immediately pinpoint the location of the 911 caller based on the calling number

• Callback the 911 caller if a disconnect occurs

Before this feature was introduced, Cisco Unified SRST supported only outbound calls to 911. With
basic 911 functionality, calls were simply routed to a Public Safety Answering Point (PSAP). The 911
operator at the PSAP would then have to verbally gather the emergency information and location from

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the caller, before dispatching a response team from the ambulance service, fire department, or police
department. Calls could not be routed to different PSAPs, based on the specific geographic areas that
they cover.

With Enhanced 911 Services, 911 calls are selectively routed to the closest PSAP based on the caller’s
location. In addition, the caller’s phone number and address automatically display on a terminal at the
PSAP. Therefore, the PSAP can quickly dispatch emergency help, even if the caller is unable to
communicate the location. Also, if the caller disconnects prematurely, the PSAP has the information it
needs to contact the 911 caller.

See Configuring Enhanced 911 Services from Cisco Unified Communications Manager Express System

Administrator Guide for more information.

How to Configure Cisco Unified SIP SRST 4.1 Features

This section contains the following tasks:

• Enabling KPML for SIP Phones, page 79

• Disabling SIP Supplementary Services for Call Forward and Call Transfer, page 81

• Configuring Idle Prompt Status for SIP Phones, page 82

Enabling KPML for SIP Phones

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

Restrictions

• This feature is supported only on Cisco Unified IP Phone 7911G, 7941G, 7941GE, 7961G, 7961GE,

7970G, and 7971GE.

• A dial plan assigned to a phone has priority over KPML.

SUMMARY STEPS

1. enable

2. configure terminal

3. voice register pool pool-tag

4. digit collect kpml

5. end

6. show voice register dial-peer

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DETAILED STEPS

Command or Action Purpose

Step 1

enable

Example:

Router> enable

Step 2

configure terminal

Example:

Router# configure terminal

Step 3

voice register pool pool-tag

Example:

Router(config)# voice register pool 4

Step 4

digit collect kpml

Example:

Router(config-register-pool)# digit collect
kpml

Step 5

end

Cisco Unified SIP SRST 4.1

Enables privileged EXEC mode.

• Enter your password if prompted.

Enters global configuration mode.

Enters voice register pool configuration mode to set
phone-specific parameters for a SIP phone.

• 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.

Enables KPML digit collection for the SIP phone.

Note This command is enabled by default for supported

phones in Cisco
Cisco

Unified SRST.

Unified CME and

Exits to privileged EXEC mode.

Example:

Router(config-register-pool)# end

Step 6

show voice register dial-peers

Example:

Router# show voice register dial-peers

What to Do Next

Displays details of all dynamically created VoIP dial peers
associated with the Cisco

Unified CME SIP register

including the defined digit collection method.

After changing the KPML configuration in Cisco Unified SRST, you do not need to create new
configuration profiles and restart the phones. Enabling or disabling KPML is effective immediately in
Cisco

Unified SRST.

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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
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

Unified SRST. You can disable these

Step 1

Step 2

Step 3

Command or Action Purpose

enable

Enables privileged EXEC mode.

• Enter your password if prompted.

Example:

Router> enable

configure terminal

Enters global configuration mode.

Example:

Router# configure terminal

voice service voip

or

dial-peer voice tag voip

Enters voice-service configuration mode to set global
parameters for VoIP features.

or

Enters dial peer configuration mode to set parameters for a

Example:

Router(config)# voice service voip

specific dial peer.

or

Router(config)# dial-peer voice 99 voip

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Command or Action Purpose

Step 4

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

Example:

Router(conf-voi-serv)# no supplementary-service
sip refer

or

Router(config-dial-peer)# no
supplementary-service sip refer

Step 5

end

Example:

Router(config-voi-serv)# end

or

Router(config-dial-peer)# end

Cisco Unified SIP SRST 4.1

Disables SIP call forwarding or call transfer supplementary
services globally or for a dial peer.

• moved-temporarily: SIP redirect response for call

forwarding.

• refer: SIP REFER message for call transfers.

• 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.

Exits to privileged EXEC mode.

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

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
immediately in Cisco Unified SRST.

Prerequisites

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

Unified SRST.

Unified SRST. Modifying the status message takes effect

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DETAILED STEPS

Command or Action Purpose

Step 1

enable

Example:

Router> enable

Step 2

configure terminal

Example:

Router# configure terminal

Step 3

voice register global

Example:

Router(config)# voice register global

Step 4

system message string

Example:

Router(config-register-global)# system message
fallback active

Step 5

end

Where to Go Next

Enables privileged EXEC mode.

• Enter your password if prompted.

Enters global configuration mode.

Enters voice register global configuration mode to set
global parameters for all supported SIP phones in a
Cisco

Unified CME environment.

Defines a status message that displays on SIP phones
registered to Cisco

• string: Up to 32 alphanumeric characters. Default is

Unified SRST.

“CM Fallback Service Operating.”

Exits to privileged EXEC mode.

Example:

Router(config-register-global)# end

Step 6

show voice register global

Example:

Router# show voice register global

Where to Go Next

The next step is configuring Cisco Unified IP phones using SCCP. For instructions, see the “Setting Up

Cisco Unified IP Phones using SCCP” section on page 85.

For additional information, see the “Additional References” section on page 26 in the “Cisco Unified

SCCP and SIP SRST Feature Overview” chapter.

Displays all global configuration parameters associated
with SIP phones.

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Where to Go Next

Cisco Unified SIP SRST 4.1

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Revised: April 23, 2012

This chapter describes how to set up the displays and features that callers will see and use on Cisco
Unified IP Phones during Cisco Unified CM fallback.

Note Ciso Unified IP Phones discussed in this chapter are just examples. For a complete list of IP phones, see

Compatibility Information.

Contents

• Information About Setting Up Cisco Unified IP Phones, page 85

• How to Set Up Cisco Unified IP Phones, page 86

• How to Set Up Cisco IP Communicator for Cisco Unified SRST, page 102

• Where to Go Next, page 103

Information About Setting Up Cisco Unified IP Phones

Cisco Unified IP Phone configuration is limited for Cisco Unified SRST because IP phones retain nearly
all Cisco Unified CM settings during Cisco Unified CM fallback. You can configure the date format,
time format, language, and system messages that appear on Cisco Unified IP Phones during
Cisco

Unified Communications Manager fallback. All four of these settings have defaults, and the
available language options depend on the IP phones and Cisco Unified CM version in use. Also available
for configuration is a secondary dial tone, which can be generated when a phone user dials a predefined
PSTN access prefix and can be terminated when additional digits are dialed. Dual-line phone
configuration is required for dual-line phone operation during Cisco Unified CM fallback.

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How to Set Up Cisco Unified IP Phones

This section contains the following tasks:

• Configuring Cisco Unified SRST to Support Phone Functions, page 86 (Required)

• Configuring Cisco Unified 8941 and 8945 SCCP IP Phones, page 88 (Required)

• Verifying That Cisco Unified SRST Is Enabled, page 89 (Optional)

• Configuring IP Phone Clock, Date, and Time Formats, page 90 (Optional)

• Configuring IP Phone Language Display, page 92 (Optional)

• Configuring Customized System Messages for Cisco Unified IP Phones, page 94 (Optional)

• Configuring a Secondary Dial Tone, page 95 (Optional)

• Configuring Dual-Line Phones, page 96 (Required Under Certain Conditions)

• Configuring Eight Calls per Button (Octo-Line), page 98 (Optional)

• Configuring the Maximum Number of Calls, page 100 (Optional)

• Troubleshooting, page 102 (Optional)

Setting Up Cisco Unified IP Phones using SCCP

Configuring Cisco Unified SRST to Support Phone Functions

Tip When the Cisco Unified SRST is enabled, Cisco Unified IP Phones do not have to be reconfigured while

in Cisco
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-dn max-directory-numbers [dual-line] [preference preference-order]

4. max-ephones max-phones

5. limit-dn phone-type max-lines

6. exit

Unified Communications Manager fallback mode because phones retain the same configuration

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DETAILED STEPS

Command or Action Purpose

Step 1

call-manager-fallback

Example:

Router(config)# call-manager-fallback

Step 2

ip source-address ip-address [port port]
[any-match | strict-match]

Example:

Router(config-cm-fallback)# ip source-address

10.6.21.4 port 2002 strict-match

Step 3

max-dn max-directory-numbers [dual-line]
[preference preference-order]

Example:

Router(config-cm-fallback)# max-dn 15 dual-line
preference 1

How to Set Up Cisco Unified IP Phones

Enters call-manager-fallback configuration mode.

Enables the router to receive messages from the Cisco IP
phones through the specified IP addresses and provides
for strict IP address verification. The default port number
is 2000.

Sets the maximum number of directory numbers (DNs)
or virtual voice ports that can be supported by the router
and activates the dual-line mode.

• 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.

Step 4

max-ephones max-phones

Example:

Router(config-cm-fallback)# max-ephones 24

• dual-line (Optional). Allows IP phones in

Cisco

Unified Communications Manager fallback

mode to have a virtual voice port with two channels.

• preference preference-order (Optional). Sets the

global preference for creating the VoIP dial peers for
all directory numbers that are associated with the
primary number. Range is from 0 to 10. Default is 0,
which is the highest preference.

The alias command also has a preference keyword
that sets alias command preference values. Setting
the alias command preference keyword allows the
default preference set with the max-dn command to
be overridden. See the

“Configuring Call Rerouting”
section on page 129 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.

Configures the maximum number of Cisco IP phones
that can be supported by the router. The default is 0. The
maximum number is platform dependent. See

Compatibility 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.

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Command or Action Purpose

Step 5

limit-dn phone-type max-lines

Example:

Router(config-cm-fallback)# limit-dn 7945 2

Setting Up Cisco Unified IP Phones using SCCP

(Optional) Limits the directory number lines on
Cisco

IP phones during Cisco Unified CM fallback.

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).

The setting for maximum lines is from 1 to 6. The default
number of maximum directory lines is set to 6. If there is
any active phone with the last line number greater than
this limit, warning information is displayed for phone
reset.

Step 6

exit

Example:

Router(config-cm-fallback)# exit

Exits call-manager-fallback configuration mode.

Configuring Cisco Unified 8941 and 8945 SCCP IP Phones

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

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

SUMMARY STEPS

1. enable

2. configure terminal

3. ephone-type phone-type

4. device-id number

5. device-type phone-type

6. end

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DETAILED STEPS

Command or Action Purpose

Step 1

enable

Example:

Router> enable

Step 2

configure terminal

Example:

Router# configure terminal

Step 3

ephone-type phone-type

Example:

Router(config)# ephone-type 8941

Step 4

device-id number

Example:

Router(config-ephone-type)# device-id 586

Step 5

device-type phone-type

Example:

Router(config-ephone-type)# device-type 8941

Step 6

end

How to Set Up Cisco Unified IP Phones

Enables privileged EXEC mode.

• Enter your password if prompted.

Enters global configuration mode.

Enters phone type to configure.

• 8941

• 8945

Specifies the device ID for the phone type.

• 8941—586

• 8945—585

Specifies the device type for the phone.

• 8941

• 8945

Exits 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:

Step 1 Enter the show running-config command to verify the configuration.

Step 2 Enter the show call-manager-fallback all command to verify that the Cisco Unified SRST feature is

enabled.

Step 3 Use the Settings display on the Cisco IP phones in your network to verify that the default router IP

address on the phones matches the IP address of the Cisco Unified SRST router.

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Step 4 To temporarily block the TCP port 2000 Skinny Client Control Protocol (SCCP) connection for one of

the Cisco

IP phones to force the Cisco IP phone to lose its connection to the 
Cisco Unified Communications Manager and register with the Cisco Unified SRST router, perform the
following steps:

a. Use the appropriate IP access-list command to temporarily disconnect a Cisco Unified IP Phone

from the Cisco Unified

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
fallback mode. The Cisco
Operating” message, and the Cisco
operating in Cisco Unified
Unified
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

d. Enter the no form of the appropriate access-list command to restore normal service for the phone.

Setting Up Cisco Unified IP Phones using SCCP

Communications Manager.

Communications Manager

IP Phone 7960 and Cisco IP Phone 7940 display a “CM Fallback Service

IP Phone 7910 displays a “CM Fallback Service” message when

Communications Manager fallback mode. When the Cisco

Communications Manager is restored, the message goes away and full Cisco IP phone

Unified SRST router.

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

SUMMARY STEPS

timezone from Cisco
Time (UTC) time from the SRST router during SRST registration. When in SRST mode, the phones take
the timezone and the UTC time, and apply a timezone offset to produce the correct time display.

Cisco IP Phone 7960 IP phones and other similar SCCP phones such as the Cisco IP Phone 7940, get
their display clock information from the local time of the SRST router during SRST registration. If the
Cisco

Unified SRST router is configured to use the Network Time Protocol (NTP) to automatically sync

the Cisco

Unified SRST router time from an NTP time server, only UTC time is delivered to the router.
This is because the NTP server could be physically located anywhere in the world, in any timezone. As
it is important to display the correct local time, use the clock timezone command to adjust or offset the
Cisco

Unified SRST router time.

The date and time formats that appear on the displays of all Cisco Unified IP Phones in Cisco Unified
CM fallback mode are selected using the date-format and time-format commands as configured below:

1. clock timezone zone 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

Unified Communications Manager. They also receive the Coordinated Universal

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DETAILED STEPS

Command or Action Purpose

Step 1

clock timezone zone hours-offset
[minutes-offset]

Example:

Router(config)# clock timezone PST -8

Step 2

call-manager-fallback

Example:

Router(config)# call-manager-fallback

Step 3

date-format {mm-dd-yy | dd-mm-yy | yy-dd-mm |
yy-mm-dd}

How to Set Up Cisco Unified IP Phones

Sets the time zone for display purposes.

• zone: Name of the time zone to be displayed when

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.

Enters call-manager-fallback configuration mode.

Sets the date format for IP phone display. The choices are
mm-dd-yy, dd-mm-yy, yy-dd-mm, and yy-mm-dd, where

Step 4

Step 5

Example

Example:

Router(config-cm-fallback)# date-format
yy-dd-mm

time-format {12 | 24}

Example:

Router(config-cm-fallback)# time-format 24

exit

Example:

Router(config-cm-fallback)# exit

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

• dd: day

• mm: month

• yy: year

The default is set to mm-dd-yy.

Sets the time display format on all Cisco Unified IP Phones
registered with the router. The default is set to a 12-hour
clock.

Exits call-manager-fallback configuration mode.

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Configuring IP Phone Language Display

During Cisco Unified CM fallback, the language displays shown on 
Cisco Unified IP Phones default to the ISO-3166 country code of US (United States). The
Cisco Unified IP Phone 7940 and Cisco Unified IP Phone 7960 can be configured for different languages
(character sets and spelling conventions) using the user-locale command.

Note This configuration option is available in Cisco SRST V2.1 and later versions running under 

Cisco Unified CM V3.2 and later versions. Systems with software prior to 
Cisco Unified SRST V2.1 and Cisco Unified CM V3.2 can use the default country, United States (US),
only.

SUMMARY STEPS

1. call-manager-fallback

2. user-locale country-code

3. exit

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DETAILED STEPS

Command or Action Purpose

Step 1

call-manager-fallback

Example:

Router(config)# call-manager-fallback

Step 2

user-locale country-code

Example:

Router(config-cm-fallback)# user-locale ES

How to Set Up Cisco Unified IP Phones

Enters call-manager-fallback configuration mode.

Selects a language by country for displays on the Cisco IP
Phone 7940 and Cisco IP Phone 7960.

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).

Step 3

exit

Example:

Router(config-cm-fallback)# exit

Examples

• NL: Dutch.

• NO: Norwegian.

• PT: Portuguese.

• RU: Russian.

• SE: Swedish.

• US: United States English (default).

Exits call-manager-fallback configuration mode.

The following example offers a configuration for the Portugal user locale:

call-manager-fallback
user-locale PT

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How to Set Up Cisco Unified IP Phones

Configuring Customized System Messages for Cisco Unified IP Phones

Use the system message command to customize the system message displayed on all Cisco Unified IP
Phones during Cisco Unified CM fallback.

One of two keywords, primary and secondary, must be included in the command. The primary
keyword is for IP phones that can support static text messages during fallback. The default display
message for primary IP phones in fallback mode is “CM Fallback Service Operating.”

The secondary keyword is for Cisco Unified IP Phones that do not support static text messages and have
a limited display space. Secondary IP phones flash messages during fallback. The default display
message for secondary IP phones in fallback mode is “CM Fallback Service.”

Changes to the display message will occur immediately after configuration or at the end of each call.

Note The normal in-service static text message is controlled by Cisco Unified Communications Manager.

SUMMARY STEPS

1. call-manager-fallback

2. system message {primary primary-string | secondary secondary-string}

DETAILED STEPS

Command or Action Purpose

Step 1

call-manager-fallback

Example:

Router(config)# call-manager-fallback

Step 2

system message {primary primary-string |
secondary secondary-string}

Example:

Router(config-cm-fallback)# system message
primary Custom Message

Step 3

exit

3. exit

Enters call-manager-fallback configuration mode.

Declares the text for the system display message on IP
phones in fallback mode.

• primary primary-string: For Cisco Unified IP Phones

that can support static text messages during fallback,
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
approximately 20

Unified IP Phone 7910. A string of

characters is allowed.

Exits call-manager-fallback configuration mode.

Example:

Router(config-cm-fallback)# exit

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Examples

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

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

Configuring a Secondary Dial Tone

A secondary dial tone can be generated when a phone user dials a predefined PSTN access prefix and
can be terminated when additional digits are dialed. An example is when a secondary dial tone is heard
after the number 9 is dialed to reach an outside line.

SUMMARY STEPS

1. call-manager-fallback

2. secondary-dialtone digit-string

How to Set Up Cisco Unified IP Phones

DETAILED STEPS

Command or Action Purpose

Step 1

call-manager-fallback

Example:

Router(config)# call-manager-fallback

Step 2

secondary-dialtone digit-string

Example:

Router(config-cm-fallback)# secondary-dialtone
9

Step 3

exit

Example:

Router(config-cm-fallback)# exit

Examples

3. exit

Enters call-manager-fallback configuration mode.

Activates a secondary dial tone when a digit string is dialed.

Exits call-manager-fallback configuration mode.

The following example sets the number 8 to trigger a secondary dial tone:

call-manager-fallback
secondary-dialtone 8

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Configuring Dual-Line Phones

Dual-line phone configuration is required for dual-line phone operation during Cisco Unified CM
fallback, see the

Cisco SRST 3.0” section on page 148.

Dual-line IP phones are supported during Cisco Unified CM fallback using the max-dn command.
Dual-line IP phones have one voice port with two channels to handle two independent calls. This
capability enables call waiting, call transfer, and conference functions on a phone-line button.

In dual-line mode, each IP phone and its associated line button can support one or two calls. Selection
of one of two calls on the same line is made using the blue Navigation button located below the phone
display. When one of the dual-line channels is used on a specific phone, other phones that share the
ephone-dn will be unable to use the secondary channel. The secondary channel will be reserved for use
with the primary dual-line channel.

It is recommended that hunting be disabled to the second channel. For more information, see the

“Configuring Dial-Peer and Channel Hunting” section on page 144.

SUMMARY STEPS

“Enabling Consultative Call Transfer and Forward Using H.450.2 and H.450.3 with

Setting Up Cisco Unified IP Phones using SCCP

1. call-manager-fallback

2. max-dn max-directory-numbers [dual-line] [preference preference-order]

3. exit

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DETAILED STEPS

Command or Action Purpose

Step 1

call-manager-fallback

Example:

Router(config)# call-manager-fallback

Step 2

max-dn max-directory-numbers [dual-line]
[preference preference-order]

Example:

Router(config-cm-fallback)# max-dn 15 dual-line
preference 1

How to Set Up Cisco Unified IP Phones

Enters call-manager-fallback configuration mode.

Sets the maximum number of directory numbers (DNs) or
virtual voice ports that can be supported by the router and
activates dual-line mode.

• 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.

Step 3

exit

Example:

Router(config-cm-fallback)# exit

Examples

The alias command also has a preference keyword that
sets alias command preference values. Setting the alias
command preference keyword allows the default
preference set with the max-dn command to be
overridden. See the

“Configuring Call Rerouting”
section on page 129 for more information on using the

max-dn command with the alias command.

Exits call-manager-fallback configuration mode.

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

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Configuring Eight Calls per Button (Octo-Line)

The octo-line feature supports up to eight active calls, both incoming and outgoing, on a single button.
Eight incoming calls to an octo-line directory number ring simultaneously. After an incoming call is
answered, the ringing stops and the remaining seven incoming calls hear a call waiting tone.

After an incoming call on an octo-line directory number is answered, the answering phone is in the
connected state. Other phones that share the directory number are in the remoteMultiline state. A
subsequent incoming call sends the call waiting tone to the phone connected to the call, and sends the
ringing tone to the other phones that are in the remoteMultiline state. All phones sharing the directory
number can pick up any of the incoming unanswered calls.

When multiple incoming calls ring on an octo-line directory number that is shared among multiple
phones, the ringing tone stops on the phone that answers the call, and the call waiting tone is heard for
other unanswered calls. The multiple instances of the ringing calls is displayed on other ephones sharing
the directory number. After a connected call on an octo-line directory number is put on-hold, any phone
that shares this directory number can pick up the held call. If a phone is in the process of transferring a
call or creating a conference, other phones that share the octo-line directory number cannot steal the call.

As new calls come in on an octo-line, the system searches for the next available idle line using the
huntstop chan tag command, where tag is a number from 1 to 8. An idle channel is selected from the
lowest number to the highest. When the highest number of allowed calls is received, the system stops
hunting for available channels. Use this command to limit the number of incoming calls on an octo-line
directory number and reserve channels for outgoing calls or features such as call transfer or conference
calls.

Setting Up Cisco Unified IP Phones using SCCP

Prerequisites

Restrictions

SUMMARY STEPS

With the new feature, you can:

• Configure only dual-line mode

• Configure only octo-line mode

• Configure dual-line mode and octo-line mode

• Cisco Unified SRST 7.0/4.3

• Cisco Unified CM 6.0

• Cisco IOS Release 12.4(15)XZ

Octo-line directory numbers are not supported by the Cisco Unified IP Phone 7902, 7920, or 7931, or
by analog phones connected to Cisco

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 channel 1-8

ATA o r C i s co VG224.

6. end

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DETAILED STEPS

Command or Action Purpose

Step 1

enable

Example:

Router> enable

Step 2

configure terminal

Example:

Router# configure terminal

Step 3

call-manager-fallback

Example:

Router(config)# call-manager-fallback

Step 4

max-dn max-no-of-directories [dual-line |
octo-line] [number octo-line]

Example:

Router(config-cm-fallback)# max-dn 15 dual-line
6 octo-line

How to Set Up Cisco Unified IP Phones

Enables privileged EXEC mode.

• Enter your password if prompted.

Enters global configuration mode.

Enters call-manager-fallback configuration mode.

Sets the maximum number of DNs or virtual voice ports
that can be supported by the router and activates dual-line
mode, octo-line mode, or both modes.

• max-no-of-directories: Maximum number of directory

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.

Step 5

Step 6

huntstop channel 1-8

Example:

Router(config-cm-fallback)# huntstop channel 4

end

Example:

Router(config)# end

• 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.

Enables channel huntstop on an octo-line, which keeps a
call from hunting to the next channel of a directory number
if the last allowed channel is busy or does not answer.

• 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.

Returns to privileged EXEC mode.

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Examples

In the following example, octo-line mode is enabled, there are 8 octo-line directory numbers, there are
a maximum of 23 directory numbers, and a maximum of 6 channels are available for incoming calls:

!
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.

Prerequisites

• Cisco Unified SRST 9.0 and later versions.

• Correct firmware, 9.2(1) or a later version, is installed.

Setting Up Cisco Unified IP Phones using SCCP

SUMMARY STEPS

DETAILED STEPS

Command or Action Purpose

Step 1

enable

Example:

Router> enable

Step 2

configure terminal

Example:

Router# configure terminal

Step 3

call-manager-fallback

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

Enables privileged EXEC mode.

• Enter your password if prompted.

Enters global configuration mode.

Enables Cisco Unified SRST support and enters
call-manager-fallback configuration mode.

Example:

Router(config)# call-manager-fallback

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