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Table Of Contents
Cisco Unified Survivable Remote Site Telephony Feature Roadmap
Information About New Features in Cisco Unified SRST V4.0
Information About New Features in Cisco SRST V3.4
Information About New Features in Cisco SRST V3.3
Information About New Features in Cisco SRST V3.2
Information About New Features in Cisco SRST V3.1
Information About New Features in Cisco SRST V3.0
Information About Features That Were New in Cisco SRST V2.1
Information About Features That Were New in Cisco SRST V2.02
Cisco Unified Survivable Remote Site Telephony Feature Roadmap
This chapter contains a list of Cisco Unified Survivable Remote Site Telephony (Cisco Unified SRST) features and the location of feature documentation.
Note
Prior to version 4.0, the name of this product was Cisco SRST.
Use Cisco Feature Navigator to find information about platform support and Cisco IOS software image support. Access Cisco Feature Navigator at http://www.cisco.com/go/fn. You must have an account on Cisco.com. If you do not have an account or have forgotten your username or password, click Cancel at the login dialog box and follow the instructions that appear.
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Contents
Documentation Organization
This document consists of the following chapters or appendixes as shown in Table 1.
Feature Roadmap
Table 2 provides a feature history summary of Cisco Unified SRST features.
Table 2 Cisco Unified SRST Features by Cisco IOS Release
Version 4.0
12.4(4)XC
12.4(9)T•
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Version 3.4
12.4(4)T
Version 3.3
12.3(14)T
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Version 3.2
12.3(11)T
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Version 3.1
12.3(7)T
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Version 3.0
12.3(4)T
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12.2(15)ZJ
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Version 2.1
12.2(15)T1
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12.2(15)T
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12.2(11)YT
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Version 2.02
12.2(13)T
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Version 2.01
12.2(11)T
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Version 2.0
12.2(8)T1
Cisco Unified SRST was implemented on the Cisco 2600XM and Cisco 2691 routers.
12.2(8)T
Cisco Unified SRST was integrated into Cisco IOS Release 12.2(8)T and implemented on the Cisco 3725 and Cisco 3745 routers and the Cisco MC3810-V3 concentrators.
12.2(2)XT
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Version 1.0
12.1(5)YD1
Support was added for 144 Cisco IP phones on the Cisco 3660 multiservice routers.
12.1(5)YD
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Information About New Features in Cisco Unified SRST V4.0
Cisco Unified SRST Version 4.0 has introduced the following new features:
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Additional Cisco Unified IP Phone Support
The following IP phones are supported with Cisco Unified SRST systems:
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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 the Cisco IP Phone 7914 Expansion Module Quick Start Guide.
No additional SRST configuration is required for these phones. They are supported in the appropriate Cisco IOS commands.
The show ephone command has been enhanced to display the configuration and status of the new Cisco IP Phones added to SRST Version 4.0. For more information, see the show ephone command in the Cisco Unified SRST and Cisco Unified SIP SRST Command Reference (All Versions).
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.
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H.323 VoIP Call Preservation Enhancements for WAN Link Failures
H.323 VoIP call preservation enhancements for WAN link failures sustains connectivity for H.323 topologies where signaling is handled by an entity, such as Cisco Unified CallManager, 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.
For configuration information see the "Configuring H.323 Gateways" chapter in the Cisco IOS H.323 Configuration Guide, Release 12.4T at http://www.cisco.com/univercd/cc/td/doc/product/software/ios123/123cgcr/vvfax_c/callc_c/h323_c/323confg/4gwconf.htm.
Video Support
This feature allows you to set video parameters for the Cisco Unified SRST to maintain close feature parity with Cisco Unified CallManager. When the Cisco Unified SRST is enabled, Cisco Unified IP Phones do not have to be reconfigured for video capabilities because all ephones retain the same configuration used with Cisco Unified CallManager. However, you must enter call-manager-fallback configuration mode to set video parameters for Cisco Unified SRST. The feature set for video is the same as that for Cisco Unified SRST audio calls.
For more information, see "Setting Video Parameters" chapter of this guide.
Information About New Features in Cisco SRST V3.4
Cisco SRST V3.4 introduced the new features described in the following section:
SIP SRST, Version 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 the Cisco SIP SRST Version 3.4 System Administrator Guide.
Information About New Features in Cisco SRST V3.3
Cisco SRST V3.3 introduced the new features described in the following sections:
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Secure SRST
Secure Cisco IP phones that are located at remote sites and that are attached to gateway routers can communicate securely with Cisco Unified CallManager using the WAN. But if the WAN link or Cisco Unified CallManager goes down, all communication through the remote phones becomes nonsecure. To overcome this situation, gateway routers can now function in secure SRST mode, which activates when the WAN link or Cisco Unified CallManager goes down. When the WAN link or Cisco Unified CallManager is restored, Cisco Unified CallManager 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 chapter "Setting Up Secure Survivable Remote Site Telephony" section on page 105.
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 following URL:
http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/ipp7970/index.htm
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Enhancement to the show ephone Command
The show ephone command has been enhanced to display the configuration and status of the Cisco Unified IP Phone 7970G and Cisco Unified IP Phone 7971G-GE. For more information, see the show ephone command in the Cisco Unified Survivable Remote Site Telephony (SRST) Command Reference (All Versions).
Information About New Features in Cisco SRST V3.2
Cisco SRST V3.2 introduced the new features described in the following sections:
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Enhancement to the alias Command
The alias command has been enhanced as follows:
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For more information, see the alias command in the Cisco Unified Survivable Remote Site Telephony (SRST) Command Reference (All Versions).
Enhancement to the cor Command
The maximum number of cor lists has been increased to 20.
For more information, see the cor command in the Cisco Unified Survivable Remote Site Telephony (SRST) Command Reference (All Versions).
Enhancement to the pickup Command
The pickup command has been 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 the Cisco Unified Survivable Remote Site Telephony (SRST) Command Reference (All Versions).
Enhancement to the user-locale Command
Theuser-locale command has been enhanced to display the Japanese Katakana country code. Japanese Katakana is available under Cisco Unified CallManager V4.0 or later.
For more information, see the user-locale command in the Cisco Unified Survivable Remote Site Telephony (SRST) Command Reference (All Versions).
Increased the Number of Cisco Unified IP Phones Supported on the Cisco 3845
The Cisco 3845 now supports 720 phones and up to 960 ephone-dns or virtual voice ports. 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:
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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).
Information About New Features in Cisco SRST V3.1
Cisco SRST V3.1 introduced the new features described in the following sections:
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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 CallManager and Cisco Aironet 1200, 1100, 350, and 340 Series of Wi-Fi (IEEE 802.11b) access points. As a key part of the Cisco AVVID Wireless Solution, the Cisco Unified Wireless IP Phone 7920 delivers seamless intelligent services, such as security, mobility, quality of service (QoS), and management, across an end-to-end Cisco network.
No configuration is necessary.
For more information, see the following URL:
http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/wip7920/
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.
For more information, see the following URL:
http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/ipp7936/
Information About New Features in Cisco SRST V3.0
Cisco SRST V3.0 introduced the new features described in the following sections:
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Additional Language Options for IP Phone Display
Displays for the Cisco Unified Unified IP Phone 7940G and Cisco Unified 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 .
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Consultative Call Transfer and Forward Using H.450.2 and H.450.3
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 H.450.3 (H.450.3) 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 PSTN interface type.
For consultative transfer to be available, the Cisco SRST router must be configured with the dual-line mode. See the "Configuring Dual-Line Phones" section on page 58.
Note
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 82.
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 55.
Dual-Line Mode
A new keyword that has been 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 command affects all IP phones on a Cisco SRST router.
For configuration information, see the "Configuring Dual-Line Phones" section on page 58.
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.
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:
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European Date Formats
The date format on Cisco IP phone displays can be configured with the following two additional formats:
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For configuration information, see the "Configuring IP Phone Clock, Date, and Time Formats" section on page 52.
Huntstop for Dual-Line Mode
A new keyword has been 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 78.
Music on Hold for Multicast from Flash Files
Cisco SRST can be configured to support continuous multicast output of music on hold (MOH) from a flash MOH file in flash memory.
For more information, see the "Configuring MOH from Flash Files" section on page 102.
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 80.
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 57.
Enhancement to the show ephone Command
Theshow ephone command has been enhanced to display the following:
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For more information, see the show ephone command in the Cisco Unified 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 100.
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 Unified CallManager.
During Cisco Unified CallManager fallback, Cisco SRST considers the Cisco VG248 to be a group of Cisco Unified IP Phones. Cisco 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 is also available in Cisco SRST Version 2.1.
For more information, see the Cisco VG248 Analog Phone Gateway Data Sheet and the Cisco VG248 Analog Phone Gateway Version 1.2(1) Release Notes.
Information About Features That Were New in Cisco SRST V2.1
Cisco SRST V2.1 introduced the new features described in the following sections:
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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:
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For configuration information, see the "Configuring IP Phone Language Display" section on page 53.
Cisco SRST Aggregation
For systems running Cisco Unified CallManager 3.3(2) and later, 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 in order 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.
For further information, go to Cisco.com and click Products & Solutions > Voice & IP Communications > 7900 Series IP Phones > Product Literature > Data Sheets or go to http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/ipp7902/index.htm.
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.
For more information, see the following URL:
http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/ipp7905g/index.htm
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.
For further information, go to Cisco.com and click Products & Solutions > Voice & IP Communications > 7900 Series IP Phones > Product Literature > Data Sheets.
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.
No configuration is necessary.
For more information, see Cisco IP Phone 7914 Expansion Module.
Enhancement to the dialplan-pattern Command
A new keyword has been 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 phone abbreviated extension number prefix digits. See the dialplan-pattern command in the
Cisco Unified Survivable Remote Site Telephony (SRST) Command Reference (All Versions).Information About Features That Were New in Cisco SRST V2.02
Cisco SRST Version 2.02 introduced the new features described in the following sections:
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Cisco Unified IP Phone Conference Station 7935 Support
The Cisco IP Conference Station 7935 is an IP-based, full-duplex hands-free conference station for use on desktops and offices and in small-to-medium-sized conference rooms. This device attaches a Cisco Catalyst 10/100 Ethernet switch port with a simple RJ-45 connection and dynamically configures itself to the IP network via the DHCP. Other than connecting the Cisco 7935 to an Ethernet switch port, no further administration is necessary. The Cisco 7935 dynamically registers to Cisco Unified CallManager for connection services and receives the appropriate endpoint phone number and any software enhancements or personalized settings, which are preloaded within Cisco Unified CallManager.
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 UnifiedCisco 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
Directory numbers were increased for the platforms shown in Table 3.
Unity Voice Mail Integration Using In-Band DTMF Signaling Across the PSTN and BRI/PRI
Unity Voice Mail and other voice-mail systems can be integrated with Cisco SRST. Voice-mail integration introduces six new commands:
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For further information, see the Cisco Unified Survivable Remote Site Telephony (SRST) Command Reference (All Versions) and the "Integrating Voice Mail with Cisco Unified SRST" section on page 147.
Posted: Mon Jun 19 11:22:33 PDT 2006
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