Specifications

Cisco Unified Survivable Remote Site Telephony Feature Roadmap
Feature Roadmap
12
Cisco Unified Survivable Remote Site Telephony Version 4.0 System Administrator Guide
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. For more
information, see Cisco IOS Survivable Remote Site Telephony (SRST) 3.2 Specifications for Cisco IOS
Software Release 12.3(11)T.
MOH Live-Feed Support
Cisco SRST has been enhanced with the new moh-live command. The moh-live command provides
live-feed MOH streams from an audio device connected to an E&M or FXO port to Cisco IP phones in
SRST mode. If an FXO port is used for a live feed, the port must be supplied with an external third-party
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 IP phones. See Configuring SRST MOH Live-Feed Support for
configuration instructions.
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. See the “Cisco Unified SRST Description” section on page 23 for more
information.
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
Appendix A: Preparing Cisco Unified SRST Support for SIP, page 181 for configuration instructions.
To use voice mail on a SIP network that connects to a Cisco Unity Express (CUE) system, use a
nonstandard SIP Notify format. To configure the Notify format, use the sip-notify keyword with the
dtmf-relay command. Using the sip-notify keyword may be required for backward compatibility with
Cisco SRST Versions 3.0 and 3.1.
Translation Profile Support
Cisco SRST 3.2 and later versions support translation profiles. Translation profiles allow you to group
translation rules together and to associate translation rules with the following:
• Called numbers
• Calling numbers
• Redirected called numbers
See the “Enabling Translation Profiles” section on page 74 for more configuration information. For more
information on thetranslation-profile, command see the Cisco Unified Survivable Remote Site
Telephony (SRST) Command Reference (All Versions).