Specifications

Cisco IOS Survivable Remote Site Telephony Fea ture Roadmap

Feature Roadmap

Cisco IOS Survivable Remote Site Telephony Version 3.4 System Administrator Guide

For more information, see the “Configuring MOH from Flash Files” section on page 94.

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 72.

Secondary Dial Tone

A secondary dial tone is available for Cisco 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 50.

Enhancement to the show ephone Command

The show ephone command has been enhanced to display the following:

• The configuration and status of additional phones (new keywords: 7905, 7914, 7935, ATA)

• The 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 the Cisco IOS Survivable Remote Site

Telephony (SRST) Command Reference (All Versions).

System Log Messages for Phone Registrations

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

Cisco 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 92.

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

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 CallManager.