Specifications

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Cisco Unified Survivable Remote Site Telephony Version 4.0 System Administrator Guide
Appendix A: Preparing Cisco Unified SRST
Support for SIP
Cisco Unified Survivable Remote Site Telephony (SRST) supports incoming and outgoing Session
Initiation Protocol (SIP) calls to and from IP phones and router voice gateway voice ports, but does not
support direct attachment of SIP phones to Cisco Unified SRST. SIP may be used in situations where the
SRST router is separate from the PSTN gateway and the SRST and PSTN gateways are linked together
using SIP (instead of H.323).
Special configurations to support SIP calls are described in this appendix. For more information about
SIP, see the Cisco IOS SIP Configuration Guide.
Note Prior to version 4.0, the name of this product was Cisco SRST.
Note The Cisco IOS Voice Configuration Library includes a standard library preface, glossary, and feature and
troubleshooting documents and is located at
http://www.cisco.com/en/US/products/ps6441/prod_configuration_guide09186a0080565f8a.html
Contents
• DTMF Relay for SIP Applications and Voice Mail, page 181
DTMF Relay for SIP Applications and Voice Mail
DTMF relay for SIP applications can be used in two voice-mail situations:
• DTMF Relay Using SIP RFC 2833, page 182
• DTMF Relay Using SIP Notify (Nonstandard), page 183