Table Of Contents
New Voice and Telephony Features in Cisco IOS Releases 12.3T and 12.4
Finding Support Information for Platforms and Cisco IOS Software Images
New Voice and Telephony Features in Cisco IOS Releases 12.3T and 12.4 in Alphabetical Order
New Voice and Telephony Features in Cisco IOS Releases 12.3T and 12.4
This document lists new Cisco IOS voice and telephony features in Cisco IOS Releases 12.3T and 12.4, and the location in the Cisco IOS Voice Configuration Library where each feature is documented. This information is presented in two tables:
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New Voice Features in Cisco IOS Releases 12.3T and 12.4 in Alphabetical Order
Unless otherwise specified, all features supported in Cisco IOS Release 12.3T are also supported in Cisco IOS Release 12.4.
To determine what software releases and platform support each feature uses, see
Cisco Feature Navigator.
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Finding Support Information for Platforms and Cisco IOS Software Images
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.
New Voice and Telephony Features in Cisco IOS Releases 12.3T and 12.4 in Alphabetical Order
Table 1 lists in alphabetical order new voice and telephony features in Cisco IOS Releases 12.3T and 12.4.
Table 1 New Voice Features in Cisco IOS Releases 12.3T and 12.4 in Alphabetical Order
Accounting Server Connectivity Failure and Recovery Detection
12.3(4)T
Provides the scriptable option to reject new calls entering the VoIP network and tear down all existing calls on detecting connectivity failure to the method list that is associated with RADIUS-based accounting servers.
AIM-CUE
12.3(7)T
Provides support for Cisco Unity Express voice mail and auto attendant for either Cisco CallManager or Cisco CallManager Express IP Communications networks. The AIM-CUE is supported on the Cisco 2600XM series, Cisco 2691, and Cisco 3700 series voice gateway routers on an AIM form factor.
Call Failure Recovery (Rotary) on IPIPGW
12.3(11)T
This enhancement eliminates the need for identical codec capabilities for all dial peers in the rotary group, and allows the IP-to-IP gateway to restart the codec negotiation process with the originating endpoint based on the codec capabilities of the next dial peer in the rotary group.
"Configuring a Cisco Multiservice IP-to-IP Gateway" chapter in the Cisco Multiservice IP-to-IP Gateway Application Guide
Call Pickup ringing extension
12.3(11)T
The SRST pickup command has been introduced to enable the PickUp soft key on all Cisco IP phones, allowing an external direct inward dialing (DID) call coming into one extension to be picked up from another extension during SRST.
The pickup command in the Cisco Survivable Remote Site Telephony Version 3.2 Command Reference.
Call Routing Enhancements to the H.323 Gatekeeper and GKTMP (GK API)
12.3(7)T
Improves routing flexibility in customer networks where an external route server is used to select potential endpoints for call completion.
(1) Nonblocking GKTMP (GK API): Timing changes associated with recovery processing when socket errors occur.
(2) Separate DNIS for alternate endpoints: It is now possible to associate a unique DNIS with each alternate endpoint.
(3) Support for "z" tag in RESPONSE xRQ:
Enhances the responses a route server can provide to the H.323 gatekeeper to allow greater flexibility for combinations of gateway endpoints and gatekeepers.
Call Statistics on Voice-Enabled Gateways
12.3(4)T
Enables the collection of voice call statistics based on user-configured time ranges.
The statistics that can be collected are from the following functional areas:
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"Voice Performance Statistics on Cisco Gateways" chapter in the Cisco IOS Voice Troubleshooting and Monitoring Guide
Cisco CallManager Express 3.0
12.3(4)T
Enables Cisco routers to deliver key system or hybrid PBX functionality for enterprise branch offices or small businesses. Cisco CME is ideal for customers who have data connectivity requirements and also have a need for a telephony solution in the same office. Whether offered through a service provider's managed services offering or purchased directly by a corporation, Cisco CME offers most of the core telephony features required in the small office, in addition to many advanced features not available with traditional telephony solutions. Being able to deliver IP telephony and data routing using a single converged solution allows customers to optimize their operations and maintenance costs, resulting in a very cost-effective solution that meets office needs.
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Cisco CallManager Express 3.1
12.3(7)T
Introduces enhancements to allow interoperability with a mix of platforms in a WAN across an H.323 network. The mix of platforms can include earlier versions of Cisco CME, Cisco CallManager, Cisco BTS Softswitch (BTS), and Cisco PSTN Gateway (PGW), and also other Cisco IOS voice gateways. The enhancements include support of H.450.12 standards for H.450 capabilities exchange with remote H.323 endpoints, automatic detection of Cisco CallManager endpoints, H.323-to H.323 hairpin call routing, and H.323-to-H.323 routing to H.450 tandem gateways.
Other enhancements introduced in this feature include Call Park, CFwdAll Soft Key Restriction Control, and enhancements to automatic line selection and ephone hunt groups. Language display localization and directory search are supported on Cisco IP Phone 7905 and Cisco IP Phone 7912. Call progress tone localization is supported on Cisco IP Phone 7902, Cisco IP Phone 7905, and Cisco IP Phone 7912.
The Cisco Wireless IP Phone 7920 and Cisco IP Phone Conference Station 7936 are fully supported in the Cisco IOS CLI and Cisco CME GUI.
Cisco CallManager Express 3.2
12.3(11)T
Cisco CallManager Express 3.2 adds a number of key telephony features including support for 240 phones, transcoding, and RCF 2833 DTMF support.
Cisco VG224 24-Port Analog Phone Gateway
12.3(7)T
Enables a hybrid of using VoIP Technology (AVVID-based architectures with Cisco CallManager as call control) with TDM analog endpoints (analog phones, fax machines, analog modems). Supported on Cisco CallManager Release 3.2 or later.
COR list increase from 10 to 20
12.3(11)T
The maximum number of COR lists has been increased to 20.
The cor command in the Cisco Survivable Remote Site Telephony Version 3.2 Command Reference.
Customizable Tone Download to Cisco IOS MGCP Gateways from Cisco Call Manager
12.3(4)T
Implements the downloading of region-specific tones and the associated frequencies, amplitudes, and cadences using XML-based configuration files during gateway registration. The feature also introduces the ability to play out tones that contain up to four frequencies.
"Configuring Tone Download to MGCP Gateway" chapter in the Cisco CallManager and Cisco IOS Interoperability Guide
Enhanced Conferencing and Transcoding for Voice Gateway Routers
12.3(8)T
Provides conferencing, transcoding, and MTP services for Cisco voice gateways in a Cisco CallManager network. Uses DSP resources on the NM-HDV2 and NM-HD high-density digital voice/fax network modules.
"Configuring Enhanced Conferencing and Transcoding for Voice Gateway Routers" chapter in the Cisco CallManager and Cisco IOS Interoperability Guide
Enhanced ITU-T G.168 Echo Cancellation
12.3(4)T
Extends the existing ITU-T G.168 Echo Cancellation functionality by adding support for medium complexity CODECs.
"Configuring Echo Cancellation" chapter in the "Cisco IOS Voic Port Configuration Guide
Enhanced ITU-T G.168 Echo Cancellation
12.3(7)T
Includes platforms using the TI C5510 DSP.
"Configuring Echo Cancellation" chapter in the "Cisco IOS Voic Port Configuration Guide
ETSI Call Transfer
12.3(8)T
Provides support for European Telecommunications Standards Institute (ETSI) explicit call transfer functionality on Cisco IOS gateways.
"Configuring Telephony Call-Redirect Features" chapter in the Cisco IOS TcL IVR and VoiceXML Application Guide
External Music on Hold Source
12.3(11)T
Cisco SRST has been enhanced with the 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. Music from a live feed is 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.
Integrating Cisco CallManager and Cisco SRST to Use Cisco SRST As a Multicast MOH Resource
Gatekeeper Prefix Selection for Hair-Pinned Calls
12.3(11)T
Allows H.323 gatekeeper to terminate/hairpin calls from a TDM/PSTN endpoint back through the same originating gateway based on priority/zone prefix values.
High-Density Analog (FXO/FXS/DID) and Digital (BRI) Extension Module for Voice/Fax (EVM-HD)
12.3(11)T
The High-Density Analog (FXO/FXS/DID) and Digital (BRI) Extension Module for Voice/Fax (EVM-HD) feature delivers a higher density integrated analog/digital voice interface. The EVM-HD-8FXS/DID baseboard provides eight FXS and DID ports. This network module accesses digital signal processor (DSP) modules on the motherboard, instead of using onboard DSPs. You can increase the port density by plugging in up to two optional expansion modules in any combination:
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PVDM2 DSP modules are used in combination with the EVM-HD-8FXS/DID baseboard and its expansion modules. PVDM2 modules are available separately and installed in the DSP module slots located inside the router chassis.
HTTP Client API for Tcl IVR
12.3(14)T
Enables Tcl IVR applications to retrieve data from or post data to an external HTTP server. Also introduces a new command-line-interface structure for configuring voice applications and support for additional Tcl 8.3.4 commands.
Inactive Call Detection
12.3(4)T
Detects inactive (silent) H.323 or SIP call-legs on Cisco IOS software-based gateways, and reports this situation to the TcL IVR 2.0 application (which can disconnect the call). Enables the Cisco IOS software to not automatically disconnect detected inactive calls. Inactivity is defined as no RTP/RTCP packets for a configurable length of time.
"Troubleshooting Voice Applications" chapter of the Cisco IOS Voice Troubleshooting and Monitoring Guide
Increase alias command from 10 to 50
12.3(11)T
The SRST alias command has been enhanced in the following areas:
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The alias command in the Cisco Survivable Remote Site Telephony Version 3.2 Command Reference
Increase phones supported from 240 to 720 on Access Router
12.3(11)T
The Cisco 3845 now supports 720 phones and up to 960 ephone-dns or virtual ports.
Interoperability Enhancements to the Cisco Multiservice IP-IP Gateway
12.3(7)T
Introduces the following enhancements:
(1) IP-to-IP gateway is now interoperable with the Cisco ATA-188 and Microsoft NetMeeting.
(2) Tcl IVR 2.0: Full support for programmable/scripted applications with VoIP endpoints; prepaid applications are one example.
(3) IP-to-IP gateway image consolidation: Combines TDM-to-IP voice gateway and IP- to-IP gateway feature sets in a single Cisco IOS image. The combined functions can run concurrently on the same hardware platform. This reduces the cost of deployment for the IP-to-IP gateway in enterprise networks.
IP Communications High-Density Digital Voice/Fax Network Module
12.3(7)T
Supports high-density digital voice and low-density analog voice connectivity along with data and integrated access connectivity. The network modules offer built-in T1/E1 VIC/VWIC ) slot for FXS, FXO, E&M, Centralized Automatic Message Accounting (CAMA), direct inward dialing (DID), BRI, or E1 and T1 cards, up to a maximum of four T1/E1 ports. Supports up to 32 HDLC channels with an aggregate capacity of 2.048 Mbps.
"Configuring Digital Voice Ports" chapter in the Cisco IOS Voice Port Configuration Guide
IP Communications Voice/Fax Network Module
12.3(4)T
Provides the ability to directly connect the PSTN and legacy telephony equipment to Cisco 2600XM series, Cisco 2691, Cisco 3600 series, and Cisco 3700 series multiservice routers, enabling applications such as IP telephony, toll bypass, and full gateway integration. The VWICs supported by the new network modules include 2- and 4-port FXS; 2- and 4-port FXO; 2-port DID, E&M, and BRI (S/T); and 1- and 2-port T1/E1.
ISDN Calling Name Display
12.3(4)T
Provides calling name display to SIP customers on calls that originate on ISDN networks.
"Configuring SIP ISDN Support Features" chapter of the Cisco IOS SIP Configuration Guide
Japanese Katakana Localization
12.3(11)T
Japanese Katakana is now supported when using the SRST user-locale command. The new JP keyword is available to Cisco SRST systems running under Cisco CallManager V4.0.
The user-locale command in the Cisco Survivable Remote Site Telephony Version 3.2 Command Reference.
Lossless Compression R1, ATM Cell Switching, and BITS Clocking
12.3(7)T
Introduces a new compression technique in DSP firmware and add enhancements to Cisco IOS software that include cell switching on ATM segmentation and reassembly (SAR), and the use of an external BITS clocking source. These features enable Cisco multiservice routers to be used to transparently groom and compress traffic in a wireless service provider network and enable a service provider to optimize the bandwidth used to backhaul the traffic from a cell site to the mobile central office for more efficient use of existing T1 and E1 lines.
Lossless Compression R1, ATM Cell Switching and BITS Clocking
Malicious Caller Identification Invocation Support for Enterprise Networks
12.3(14)T
Extends support for MCID service in the PSTN to the Cisco 2801.
Malicious Caller Identification Invocation Support for Enterprise Networks
MCID for Cisco IOS Voice Gateways
12.3(11)T
Supports the Malicious Call Identification (MCID) supplementary service that enables Cisco CallManager 4.0 to identify the source of malicious calls.
"Configuring MCID for Cisco IOS Voice Gateways" chapter in the Cisco CallManager and Cisco IOS Interoperability Guide.
Media and Signaling Authentication and Encryption Feature for Cisco IOS MGCP Gateways
12.3(14)T
Delivers media and signaling authentication and encryption on the Cisco 2600XM series, Cisco 2691, Cisco 3660 series, Cisco 3700 series, and Cisco VG224. This feature enables secure gateway-to-gateway and IP-phone-to-gateway calls, and interoperates with Cisco 7970 IP phones.
Media and Signaling Authentication and Encryption Feature for Cisco IOS MGCP Gateways
MGCP Configuration Control for Setting Fax Rate
12.3(8)T
Establishes the maximum fax transmission rate for MGCP T.38 sessions. MGCP fax rate is set to the highest possible transmission speed allowed by the voice codec.
"Configuring T.38 Fax Relay" section in the Cisco Fax Services over IP Application Guide
MGCP Gateway Support for Cisco CallManager Network Specific Facilities
12.3(4)T
Provides an enhancement to Cisco CallManager functionality enabling users to configure the NSF ISDN information element of the route pattern. This feature is compatible with Cisco CallManager Version 3.3(2) and later.
"Configuring MGCP PRI Backhaul and T1 CAS Support for Cisco CallManager" chapter in the Cisco CallManager and Cisco IOS Interoperability Guide
MGCP Line Control Signaling Package support
12.3(8)T
Supports the transport of line supervision signals in the media stream using RFC 2833 event packets in PacketCable GR303-switched IP systems, using the modified mgcp package-capability package command. When the lcs-package keyword is used, the named telephony events (NTEs) associated with the LCS package are enabled automatically.
"Basic MGCP Configuration" chapter of the Cisco IOS MGCP and Related Protocols Configuration Guide
MGCP-Controlled Backhaul of BRI Signaling in Conjunction with Cisco CallManager
12.3(2)T
Provides MGCP service to remote-office media gateways that connect by means of ISDN BRI trunks to a centralized Cisco CallManager media-gateway controller for the purpose of call processing. D-channel signal information is backhauled to the call manager through a Transmission Control Protocol (TCP) session.
"Configuring MGCP-Controlled Backhaul of BRI Signaling in Conjunction with Cisco CallManager" chapter in the Cisco CallManager and Cisco IOS Interoperability Guide.
MLPP for Analog and BRI Endpoints on Cisco IOS Voice Gateways
12.3(14)T
Provides the capability for Cisco IOS voice gateways to present analog and basic rate interface (BRI) phones to be controlled by Cisco CallManager as though they were Cisco IP phones, enabling the following:
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MLPP for Analog and BRI Endpoints on Cisco IOS Voice Gateways
MLPP for Cisco IOS Voice Gateways
12.3(11)T
Supports Multilevel Precedence and Preemption (MLPP) service, allowing authorized users to preempt lower priority voice calls using Cisco CallManager version 4.0.
"Configuring MGCP Gateway Support for Cisco CallManager" chapter in the Cisco CallManager and Cisco IOS Interoperability Guide.
NextPort Voice Tuning and Background Noise Statistics with NextPort Dual-Filter G.168 Echo Cancellation
12.3(11)T
This feature allows you to dynamically configure voice services on the NextPort-based platforms: Cisco AS5350, Cisco AS5400, Cisco AS5400HPX, and Cisco AS5850. This feature also provides improved voice quality and statistics reporting and adds dual-filter G.168 echo canceller capability in NextPort SPE firmware (SPEware) version 10.2.2 and later with Cisco IOS Release 12.3(11)T.
"NextPort Voice Tuning and Background Noise Statistics with NextPort Dual-Filter G.168 Echo Cancellation" chapter in the Cisco IOS Voice Port Configuration
Option to disable H225 TCP Timer from phone to gateway to maintain calls in progress during WAN outage
12.3(11)T
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 for SRST on Cisco IOS Releases 12.3(7)T1 and higher.
"Overview of Cisco IOS SRST" chapter in the Cisco IOS SRST Version 3.2 System Administrator Guide
Out-of-Band to In-Band DTMF Relay for Cisco IOS Voice Gateways
12.3(11)T
RFC 2833 capability enabling DTMF relay communication between SIP devices and nonSIP endpoints using Cisco CallManager version 4.0.
"Configuring Enhanced Conferencing and Transcoding for Voice Gateway Routers" chapter in the Cisco CallManager and Cisco IOS Interoperability Guide.
Overlap Signaling Processing on H.323 Terminating Gateways
12.3(11)T
In an overlap signaling scenario, The called number in the SETUP message does not contain enough digits to match the incoming dial peer for the dial peer to select the right application. With this change, the H.323 layer determines if a partial match is detected, appends the called number with the needed digits. The new called number is checked to see if it matches any of the incoming dial-peers. If either full match or no match is returned, the call proceeds with SETUP procedure.
PCR Support for the Cisco Signaling Link Terminal
12.3(2)T
Adds support for Message Transfer Part Layer 2 (MTP2) PCR on the Cisco signaling link terminal (SLT). Preventive Cyclic Retransmission (PCR) adds basic MTP2 functionality that is used when Signaling System 7 (SS7) signaling links are transmitted over satellite connections between the Cisco SLT and a signal transfer point (STP).
Persistent TDM Switched Circuits
12.3(2)T
Enables the Cisco AS5850 universal gateway to connect one or more DS0s from one or more E1s or T1s to another set of E1s or T1s.
PLAR (Private Line Automatic Ring-down) for Trading Turrets
12.3(4)T
Re-establishes an H.323 call automatically if one end hangs up or goes on hold and then goes off-hook again. This provides PLAR functionality used in the financial industry for trading Turret operation of PLAR.
QSIG Supplementary Features for Cisco IOS Voice Gateways
12.3(11)T
Supports Q Signaling (QSIG) over PRI backhaul interfaces on MGCP gateways to Cisco CallManager version 4.0.
"Configuring MGCP PRI Backhaul and T1 CAS Support for Cisco CallManager" chapter in the Cisco CallManager and Cisco IOS Interoperability Guide.
RCF 2833 DTMF Support from SCCP Devices to Cisco Unity Express
12.3(11)T
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 tmf-relay rtp-nte command.
Cisco IOS Survivable Remote Site Telephony Version 3.2 System Administrator Guide
Second-Generation 1- and 2-Port T1/E1 Multiflex Trunk Voice/WAN Interface Cards
12.3(14)T
Enables T1/E1 multiflex voice/WAN interface cards to support enhanced voice and data applications in Cisco multiservice routers. Provides the following: flexible T1 and E1 support; drop-and-insert multiplexing capability on all versions; support for a dedicated hardware echo-cancellation module; and, on 2-port cards, capability for each port to be clocked from an independent clock source.
Second-Generation 1- and 2-Port T1/E1 Multiflex Trunk Voice/WAN Interface Cards
"Configuring Hardware Echo Cancellation" chapter in the Voice Port Configuration Guide.
Secure SRST
12.3(14)T
Secure Cisco IP phones that are located at remote sites and that are attached to gateway routers can communicate securely using the WAN with Cisco CallManager. But if the WAN link or Cisco 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 CallManager goes down. When the WAN link or Cisco CallManager is restored, Cisco 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.
"Setting Up Secure SRST" chapter in the Cisco IOS Survivable Remote Site Telephony Version 3.3 System Administrator Guide.
Signal ISDN B-Channel ID to Enable Application Control of Voice Gateway Trunks
12.3(7)T
Enables call management applications to identify specific ISDN bearer (B) channels used during a voice gateway call for billing purposes. With the identification of the B channel, SIP and H.323 gateways can enable port-specific features such as voice recording and call transfer.
"Configuring SIP ISDN Support Features chapter of the Cisco IOS SIP Configuration Guide and "Configuring H.323 Gateway" chapter of the Cisco IOS H.323 Configuration Guide
SIP Audible Message-Waiting Indicator for FXS Phones
12.3(8)T
Enables an FXS port on a voice gateway to receive audible MWI in a SIP-enabled network. The FXS port on a voice gateway is an RJ-11 connector that allows connections to basic telephone service equipment.
"Configuring SIP MWI Support" chapter of the Cisco IOS SIP Configuration Guide
SIP Debug Output Filtering Support
12.3(4)T
Provides the ability to filter relevant SIP debugging traces of desired calls.
SIP Gateway Support Enhancements to the bind Command
12.3(4)T
Allows configuration of the source IP address of signaling packets or both signaling and media packets using the bind command.
"Configuring SIP Gateway Support for the bind Command" chapter of the Cisco IOS SIP Configuration Guide
SIP Header Support and SUBSCRIBE and NOTIFY for External Triggers
12.3(4)T
Provides a mechanism for applications to send and receive SIP headers and to send SUBSCRIBE messages and receive NOTIFY events.
"Configuring Additional SIP Application Support" chapter of the Cisco IOS SIP Configuration Guide
SIP NOTIFY-Based Out-of-Band DTMF Relay Support
12.3(4)T
Supports SCCP devices through SIP originating and terminating gateway use of Cisco proprietary NOTIFY-based out-of-band DTMF relay, which can also be used by analog phones attached to analog voice ports (FXS) on a router.
"SIP NOTIFY-Based Out-of-Band DTMF Relay Support" chapter of the Cisco IOS SIP Configuration Guide
SIP Redirect Processing Enhancement
12.3(4)T
Allows flexibility in the handling of incoming redirect or 3xx class of responses. Redirect responses can now be enabled or disabled through the command-line interface, providing critical functionality for service providers who deploy Cisco gateways.
"Basic SIP Configuration" chapter of the Cisco IOS SIP Configuration Guide
SIP Register Support
12.3(4)T
Allows SIP gateways to register E.164 numbers to a SIP proxy or registrar on behalf of analog telephone voice ports (FXS), IP phone virtual voice ports (EFXS), and local SCCP phones.
"Basic SIP Configuration" chapter of the Cisco IOS SIP Configuration Guide
SIP RFC 3261 Enhancements
12.3(4)T
Provides enhanced SIP functionality on Cisco IOS gateways, as defined by RFCs 3261 and 3311, and support for the following:
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"Basic SIP Configuration" chapter of the Cisco IOS SIP Configuration Guide
SIP Survivable Remote Site Telephony (SRST) Version 3.0
12.3(4)T
Describes Survivable Remote Site Telephony (SRST) functionality for Session Initiation Protocol (SIP) networks. SIP SRST provides backup to an external SIP proxy server by providing basic registrar and redirect services. These services are used by a SIP IP phone in the event of a WAN connection outage where the SIP phone is unable to communicate with its primary SIP proxy. The SIP SRST device also provides PSTN gateway access for placing and receiving PSTN calls.
SIP: Cisco IOS Gateway HTTP Digest Authentication and Registration
12.3(8)T
Implements authentication using the digest access on the client side of a common SIP stack. The Cisco IOS Session Initiation Protocol (SIP) gateway responds to authentication challenges from an authenticating server, proxy server, or user agent server (UAS). This feature also maintains parity between the Cisco gateways, proxy servers, and SIP phones that already support authentication.
"Configuring SIP AAA Features" chapter in the Cisco IOS SIP Configuration Guide
SIP: Cisco IOS Gateway Reason Header and Buffered Calling Name Completion
12.3(8)T
Implements support for the followingm functions: Reason Header and Buffered Calling Name Completion. The Cisco IOS SIP gateway reason header support provided on Cisco IOS gateways is defined by RFC 3326.
"Configuring SIP Message, Timer, and Response Features" chapter of the Cisco IOS SIP Configuration Guide
SRST: Survivable Remote Site Telephony Version 3.0
12.3(4)T
Provides Cisco CallManager with fallback support for Cisco IP phones attached to a Cisco router on your local network. Cisco SRST enables routers to provide call-handling support for Cisco IP phones when they lose connection to remote primary, secondary, or tertiary Cisco CallManager installations or when the WAN connection is down.
Cisco CallManager supports Cisco IP phones at remote sites attached to Cisco multiservice routers across the WAN. Prior to Cisco SRST, when the WAN connection between a router and the Cisco CallManager failed or connectivity with Cisco CallManager was lost for some reason, Cisco IP phones on the network became unusable for the duration of the failure. Cisco SRST overcomes this problem and ensures that the Cisco IP phones offer continuous (although minimal) service by providing call-handling support for Cisco IP phones directly from the Cisco 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 IP phones that are registered with the router. When the WAN link or connection to the primary Cisco CallManager is restored, call handling reverts back to the primary Cisco CallManager.
Support Translation Profiles (CME and SRST)
12.3(11)T
Cisco SRST 3.2 supports translation profiles. Translation profiles allow you to group translation rules together and to associate translation rules with the following:
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"Setting Up Call Handling" chapter in the Cisco IOS Survivable Remote Site Telephony Version 3.2 System Administrator Guide.
Also, the translation-profile command in the Cisco IOS Survivable Remote Site Telephony Version 3.2 System Administrator Guide.
Survivable Remote Site Telephony 3.1
12.3(7)T
Provides Cisco CallManager with fallback support for Cisco IP phones attached to a Cisco router on a local network. The Cisco Wireless IP Phone 7920 and Cisco IP Phone Conference Station 7936 are fully supported.
T.37 Fax Status Notification Enhancement in an MTA Environment
12.3(7)T
Provides the ability to delegate control of fax operations directly to a mail transfer agent (MTA) by configuring a T.37 fax off-ramp gateway to deliver all fax delivery errors to the fax mail originator in Simple Mail Transfer Protocol (SMTP) delivery status notification (DSN) messages with descriptive error codes.
"Configuring T.37 Store-and-Forward Fax" chapter in the Cisco Fax Services over IP Application Guide
T.38 Fax Relay on the Cisco Catalyst 6000 and Cisco 7600 Communication Media Module
12.3(14)T
Provides support for T.38 fax relay on the Cisco Catalyst 6000 and Cisco 7600 CMM.
"Configuring T.38 Fax Relay" chapter in the Cisco Fax Services over IP Application Guide.
T.38 Fax Statistics
12.3(14)T
Enables access servers with NextPort digital signal processors to gather detailed statistics about T.38 fax-relay calls. Statistics can be compiled into detailed call records for diagnostic and billing purposes.
Tcl IVR 2.0 Session Interaction
12.3(4)T
Allows different instances of Tcl IVR applications (sessions) to communicate with other sessions on the same gateway and for applications to dynamically bridge call legs between different sessions. This enables different callers on the same gateway to be notified of each others' presence and to interact. You can also start a session without an active call leg so that a session can act as an application service for other sessions. This feature is useful for implementing a call-monitoring "server" application that is responsible for monitoring incoming calls and dynamically connecting selected callers.
Configuring Tcl IVR 2.0 Session Interaction chapter of the Cisco IOS Tcl IVR and VoiceXML Applications Guide
V.110 Support for MGCP-Dial
12.3(2)T
Provides support for V.110 encapsulation for Dial-MGCP Applications.
"Configuring NAS Package for MGCP" chapter of the Cisco IOS MGCP and Related Protocols Configuration Guide
Videoconferencing for the Cisco Multiservice IP-to-IP Gateway
12.3(4)T
Provides enhanced Quality of Service through RSVP synchronization with H.323 signaling protocol and DSCP packet marking.
"Configuring Multiservice IP-to-IP Videoconferencing" chapter in the Cisco Multiservice IP-to-IP Gateway
Voice Application HTTP Client Cookie Support
12.3(8)T
Implements HTTP cookie support for Cisco IOS VoiceXML applications.
Configuring Basic Functionality for Tcl IVR and VoiceXML Applications" chapter in the Cisco IOS Tcl IVR and VoiceXML Application Guide.pdf
Voice Application Monitoring and Troubleshooting Enhancements
12.3(8)T
Enables detailed monitoring of voice application instances and call legs using event logs and statistics. Records for terminated application instances and call legs are saved in history to assist in fault isolation. This comprehensive management information helps you diagnose problems in the network and identify the causes.
Monitoring and Troubleshooting Voice Applications chapter in the Cisco IOS Tcl IVR and VoiceXML Application Guide.pdf
Voice DSP Crash Dump Analysis
12.3(4)T
Allows Cisco IOS voice platforms using TI DSPs the ability to capture the contents of the DSP memory into a dump file if there is a DSP crash.
"Troubleshooting Digital Voice Interfaces to the IP Newtwork" chapter in th Cisco IOS Voice Troubleshooting and Monitoring Guide
VoiceXML Voice Store and Forward
12.3(7)T
Adds VXML capability on Cisco 2691 router and Cisco 3725 and Cisco 3745 routers.
VoIP Alternate Path Fallback SNMP Trap
12.3(14)T
Enhances support for the PSTN Fallback feature by providing the capability to generate SNMP traps when the fallback subsystem redirects or rejects an H.323 VoIP call because a network condition fails to meet a configured threshold.
VoIP Debug Filtering
12.3(4)T
Allows you to filter and trace voice call debug messages based on selected filtering criteria, reducing the volume of output for more efficient troubleshooting.
"Filtering Troubleshooting Output" chapter of the Cisco IOS Voice Troubleshooting and Monitoring Guide
VoIP Internal Error Codes
12.3(4)T
Generates internal error codes (IECs) for gateway-detected errors that cause the gateway to release or refuse a call. IECs enhance troubleshooting for VoIP networks by helping to determine the source and reason for call termination.
"Cisco VoIP Intewrnal Error Codes" chapter in the Cisco IOS Voice Troubleshooting and Monitoring Guide
New Voice and Telephony Features in Cisco IOS Releases 12.3T and 12.4 Listed by First Supported Release
Table 2 lists new voice and telephony features in Cisco IOS Release 12.3T and 12.4 by the maintenance release in which each feature was added. The most recent release is listed first.
Table 2 New Voice Features in Cisco IOS Releases 12.3T and 12.4 Listed by First Supported Release
12.3(14)T
HTTP Client API for Tcl IVR
Enables Tcl IVR applications to retrieve data from or post data to an external HTTP server. Also introduces a new command-line-interface structure for configuring voice applications and support for additional Tcl 8.3.4 commands.
12.3(14)T
Malicious Caller Identification Invocation Support for Enterprise Networks
Extends support for MCID service in the PSTN to the Cisco 2801.
Malicious Caller Identification Invocation Support for Enterprise Networks
12.3(14)T
MCID for Cisco IOS Voice Gateways
Extends support for the Malicious Call Identification (MCID) supplementary service to the Cisco 2801. This feature enables Cisco CallManager to identify the source of malicious calls.
"Configuring MCID for Cisco IOS Voice Gateways" chapter in the Cisco CallManager and Cisco IOS Interoperability Guide.
12.3(14)T
Media and Signaling Authentication and Encryption Feature for Cisco IOS MGCP Gateways
Delivers media and signaling authentication and encryption on the Cisco 2600XM series, Cisco 2691, Cisco 3660 series, Cisco 3700 series, and Cisco VG224. This feature enables secure gateway-to-gateway and IP-phone-to-gateway calls, and interoperates with Cisco 7970 IP phones.
Media and Signaling Authentication and Encrypti on Feature for Cisco IOS MGCP Gateways
12.3(14)T
MLPP for Analog and BRI Endpoints on Cisco IOS Voice Gateways
Provides the capability for Cisco IOS voice gateways to present analog and basic rate interface (BRI) phones to be controlled by Cisco CallManager as though they were Cisco IP phones, enabling the following:
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MLPP for Analog and BRI Endpoints on Cisco IOS Voice Gateways
12.3(14)T
Second-Generation 1- and 2-Port T1/E1 Multiflex Trunk Voice/WAN Interface Cards
Enables T1/E1 multiflex voice/WAN interface cards to support enhanced voice and data applications in Cisco multiservice routers. Provides the following: flexible T1 and E1 support; drop-and-insert multiplexing capability on all versions; support for a dedicated hardware echo-cancellation module; and, on 2-port cards, capability for each port to be clocked from an independent clock source.
Second-Generation 1- and 2-Port T1/E1 Multiflex Trunk Voice/WAN Interface Cards
"Configuring Hardware Echo Cancellation" chapter in the Voice Port Configuration Guide.
12.3(14)T
Secure SRST
Secure Cisco IP phones that are located at remote sites and that are attached to gateway routers can communicate securely using the WAN with Cisco CallManager. But if the WAN link or Cisco 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 CallManager goes down. When the WAN link or Cisco CallManager is restored, Cisco 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.
"Setting Up Secure SRST" chapter in the Cisco IOS Survivable Remote Site Telephony Version 3.3 System Administrator Guide.
12.3(14)T
T.38 Fax Relay on the Cisco Catalyst 6000 and Cisco 7600 Communication Media Module
Provides support for T.38 fax relay on the Cisco Catalyst 6000 and Cisco 7600 CMM.
"Configuring T.38 Fax Relay" chapter in the Cisco Fax Services over IP Application Guide.
12.3(14)T
T.38 Fax Statistics
Enables access servers with NextPort digital signal processors to gather detailed statistics about T.38 fax-relay calls. Statistics can be compiled into detailed call records for diagnostic and billing purposes.
12.3(14)T
VoIP Alternate Path Fallback SNMP Trap
Enhances support for the PSTN Fallback feature by providing the capability to generate SNMP traps when the fallback subsystem redirects or rejects an H.323 VoIP call because a network condition fails to meet a configured threshold.
12.3(11)T
Cisco CallManager Express 3.2
Cisco CallManager Express 3.2 adds a number of key telephony features including support for 240 phones, transcoding, and RFC 2833 DTMF support.
12.3(11)T
Call Failure Recovery (Rotary) on IPIPGW
This enhancement eliminates the need for identical codec capabilities for all dial peers in the rotary group, and allows the IP-to-IP gateway to restart the codec negotiation process with the originating endpoint based on the codec capabilities of the next dial peer in the rotary group.
12.3(11)T
Call Pickup ringing extension
The SRST pickup command has been introduced to enable the PickUp soft key on all Cisco IP phones, allowing an external direct inward dialing (DID) call coming into one extension to be picked up from another extension during SRST.
The pickup command in the Cisco Survivable Remote Site Telephony Version 3.2 Command Reference.
12.3(11)T
COR list increase from 10 to 20
The maximum number of COR lists has been increased to 20.
The cor command in the Cisco Survivable Remote Site Telephony Version 3.2 Command Reference.
12.3(11)T
External Music on Hold Source
Cisco SRST has been enhanced with the 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. Music from a live feed is 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.
Integrating Cisco CallManager and Cisco SRST to Use Cisco SRST As a Multicast MOH Resource
12.3(11)T
Gatekeeper Prefix Selection for Hair-Pinned Calls
Allows the H.323 Gatekeeper to terminate/hairpin calls from a TDM/PSTN endpoint back through the same originating gateway based on priority/zone prefix values.
12.3(11)T
High-Density Analog (FXO/FXS/DID) and Digital (BRI) Extension Module for Voice/Fax (EVM-HD)
The High-Density Analog (FXO/FXS/DID) and Digital (BRI) Extension Module for Voice/Fax (EVM-HD) feature delivers a higher density integrated analog/digital voice interface. The EVM-HD-8FXS/DID baseboard provides eight FXS and DID ports. This network module accesses digital signal processor (DSP) modules on the motherboard, instead of using onboard DSPs. You can increase the port density by plugging in up to two optional expansion modules in any combination:
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PVDM2 DSP modules are used in combination with the EVM-HD-8FXS/DID baseboard and its expansion modules. PVDM2 modules are available separately and installed in the DSP module slots located inside the router chassis.
12.3(11)T
Increase alias command from 10 to 50
The SRST alias command has been enhanced in the following areas:
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The alternate-number argument can be used in multiple alias commands.
The alias command in the Cisco Survivable Remote Site Telephony Version 3.2 Command Reference
12.3(11)T
Increase phones supported from 240 to 720 on Access Router
The Cisco 3845 now supports 720 phones and up to 960 ephone-dns or virtual ports.
12.3(11)T
Japanese Katakana Localization
Japanese Katakana is now supported when using the SRST user-locale command. The new JP keyword is available to Cisco SRST systems running under Cisco CallManager V4.0.
The user-locale command in the Cisco Survivable Remote Site Telephony Version 3.2 Command Reference.
12.3(11)T
MCID for Cisco IOS Voice Gateways
Supports the Malicious Call Identification (MCID) supplementary service that enables Cisco CallManager 4.0 to identify the source of malicious calls.
"Configuring MCID for Cisco IOS Voice Gateways" chapter in the Cisco CallManager and Cisco IOS Interoperability Guide.
12.3(11)T
MLPP for Cisco IOS Voice Gateways
Supports Multilevel Precedence and Preemption (MLPP) service, allowing authorized users to preempt lower priority voice calls using Cisco CallManager 4.0.
"Configuring MGCP Gateway Support for Cisco CallManager" chapter in the Cisco CallManager and Cisco IOS Interoperability Guide.
12.3(11)T
NextPort Voice Tuning and Background Noise Statistics with NextPort Dual-Filter G.168 Echo Cancellation
This feature allows you to dynamically configure voice services on the NextPort-based platforms: Cisco AS5350, Cisco AS5400, Cisco AS5400HPX, and Cisco AS5850. This feature also provides improved voice quality and statistics reporting and adds dual-filter G.168 echo canceller capability in NextPort SPE firmware (SPEware) version 10.2.2 and later with Cisco IOS Release 12.3(11)T.
"NextPort Voice Tuning and Background Noise Statistics with NextPort Dual-Filter G.168 Echo Cancellation" chapter in the Cisco IOS Voice Port Configuration
12.3(11)T
Option to disable H225 TCP Timer from phone to gateway to maintain calls in progress during WAN outage
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 for SRST on Cisco IOS Releases 12.3(7)T1 and higher.
"Overview of Cisco IOS SRST" chapter in the Cisco IOS SRST Version 3.2 System Administrator Guide
12.3(11)T
Out-of-Band to In-Band DTMF Relay for Cisco IOS Voice Gateways
RFC 2833 capability enabling DTMF relay communication between SIP devices and nonSIP endpoints using Cisco CallManager 4.0
"Configuring Enhanced Conferencing and Transcoding for Voice Gateway Routers" chapter in the Cisco CallManager and Cisco IOS Interoperability Guide.
12.3(11)T
Overlap Signaling Processing on H.323 Terminating Gateways
In an overlap signaling scenario, The called number in the SETUP message does not contain enough digits to match the incoming dial peer for the dial peer to select the right application. With this change, the H.323 layer determines if a partial match is detected, appends the called number with the needed digits. The new called number is checked to see if it matches any of the incoming dial-peers. If either full match or no match is returned, the call will proceeds with SETUP procedure.
12.3(11)T
QSIG Supplementary Features for Cisco IOS Voice Gateways
Supports Q Signaling (QSIG) over PRI backhaul interfaces on MGCP gateways to Cisco CallManager version 4.0.
"Configuring MGCP PRI Backhaul and T1 CAS Support for Cisco CallManager" chapter in the Cisco CallManager and Cisco IOS Interoperability Guide.
12.3(11)T
RCF 2833 DTMF Support from SCCP Devices to Cisco Unity Express
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.
Cisco IOS Survivable Remote Site Telephony Version 3.2 System Administrator Guide
12.3(11)T
Support Translation Profiles (CME and SRST)
Cisco SRST 3.2 supports translation profiles. Translation profiles allow you to group translation rules together and to associate translation rules with the following:
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"Setting Up Call Handling" chapter in the Cisco IOS Survivable Remote Site Telephony Version 3.2 System Administrator Guide.
Also, the translation-profile command in the Cisco IOS Survivable Remote Site Telephony Version 3.2 System Administrator Guide
12.3(8)T
Enhanced Conferencing and Transcoding for Voice Gateway Routers
Provides conferencing, transcoding, and MTP services for Cisco voice gateways in a Cisco CallManager network. Uses DSP resources on the NM-HDV2 and NM-HD high-density digital voice/fax network modules.
"Configuring Enhanced Conferencing and Transcoding for Voice Gateway Routers" chapter in the Cisco CallManager and Cisco IOS Interoperability Guide
12.3(8)T
ETSI Call Transfer
Provides support for European Telecommunications Standards Institute (ETSI) explicit call transfer functionality on Cisco IOS gateways.
"Configuring Telephony Call-Redirect Features" chapter in the Cisco IOS TcL IVR and VoiceXML Application Guide.
12.3(8)T
MGCP Configuration Control for Setting Fax Rate
Establishes the maximum fax transmission rate for MGCP T.38 sessions. The MGCP fax rate is set to the highest possible transmission speed allowed by the voice codec by default.
"Configuring T.38 Fax Relay" section in the Cisco Fax Services over IP Application Guide
12.3(8)T
MGCP Line Control Signaling Package support
Supports the transport of line supervision signals in the media stream using RFC 2833 event packets in PacketCable GR303-switched IP systems, using the modified mgcp package-capability package command. When the lcs-package keyword is used, the named telephony events (NTEs) associated with the LCS package are enabled automatically.
"Basic MGCP Configuration" chapter of the Cisco IOS MGCP and Related Protocols Configuration Guide
12.3(8)T
SIP Audible Message-Waiting Indicator for FXS Phones
Enables an FXS port on a voice gateway to receive audible MWI in a SIP-enabled network. The FXS port on a voice gateway is an RJ-11 connector that allows connections to basic telephone service equipment.
"Configuring SIP MWI Support" chapter of the Cisco IOS SIP Configuration Guide
12.3(8)T
SIP: Cisco IOS Gateway HTTP Digest Authentication and Registration
Implements authentication using the digest access on the client side of a common SIP stack. The Cisco IOS Session Initiation Protocol (SIP) gateway responds to authentication challenges from an authenticating server, proxy server, or user agent server (UAS). This feature also maintains parity between the Cisco gateways, proxy servers, and SIP phones that already support authentication.
"Configuring SIP AAA Features" chapter in the Cisco IOS SIP Configuration Guide
12.3(8)T
SIP: Cisco IOS Gateway Reason Header and Buffered Calling Name Completion
Implements support for the following functions: Reason Header and Buffered Calling Name Completion. The Cisco IOS SIP gateway reason header support provided on Cisco IOS gateways is defined by RFC 3326.
"Configuring SIP Message, Timer, and Response Features" chapter of the Cisco IOS SIP Configuration Guide
12.3(8)T
Voice Application HTTP Client Cookie Support
Implements HTTP cookie support for Cisco IOS VoiceXML applications.
Configuring Basic Functionality for Tcl IVR and VoiceXML Applications" chapter in the Cisco IOS Tcl IVR and VoiceXML Application Guide.pdf
12.3(8)T
Voice Application Monitoring and Troubleshooting Enhancements
Enables detailed monitoring of voice application instances and call legs using event logs and statistics. Records for terminated application instances and call legs are saved in history to assist in fault isolation. This comprehensive management information helps you diagnose problems in the network and identify the causes.
"Monitoring and Troubleshooting Voice Applications" chapter in the Cisco IOS Tcl IVR and VoiceXML Application Guide.pdf
12.3(7)T
AIM-CUE
Provides support for Cisco Unity Express voice mail and auto attendant for either Cisco CallManager or Cisco CallManager Express IP Communications networks. The AIM-CUE is supported on the Cisco 2600XM series, Cisco 2691, and Cisco 3700 series voice gateway routers on an AIM form factor.
12.3(7)T
Call Routing Enhancements to the H.323 Gatekeeper and GKTMP (GK API)
Improves routing flexibility in customer networks where an external route server is used to select potential endpoints for call completion.
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Enhances the responses a route server can provide to the H.323 gatekeeper to allow greater flexibility for combinations of gateway endpoints and gatekeepers.
12.3(7)T
Cisco CallManager Express 3.1
Introduces enhancements to allow interoperability with a mix of platforms in a WAN across an H.323 network. The mix of platforms can include earlier versions of Cisco CME, Cisco CallManager, Cisco BTS Softswitch (BTS), and Cisco PSTN Gateway (PGW), and also other Cisco IOS voice gateways. The enhancements include support of H.450.12 standards for H.450 capabilities exchange with remote H.323 endpoints, automatic detection of Cisco CallManager endpoints, H.323-to H.323 hairpin call routing, and H.323-to-H.323 routing to H.450 tandem gateways.
Other enhancements introduced in this feature include Call Park, CFwdAll Soft Key Restriction Control, and enhancements to automatic line selection and ephone hunt groups. Language display localization and directory search are supported on Cisco IP Phone 7905 and Cisco IP Phone 7912. Call progress tone localization is supported on Cisco IP Phone 7902, Cisco IP Phone 7905, and Cisco IP Phone 7912.
The Cisco Wireless IP Phone 7920 and Cisco IP Phone Conference Station 7936 are fully supported in the Cisco IOS CLI and Cisco CME GUI.
12.3(7)T
Cisco VG224 24-Port Analog Phone Gateway
Enables a hybrid of using VoIP Technology (AVVID based architectures with Cisco CallManager as call control) with TDM analog endpoints (analog phones, fax machines, analog modems). Supported on Cisco CallManager Release 3.2 or a later release.
12.3(7)T
Enhanced ITU-T G.168 Echo Cancellation
Includes platforms using the TI C5510 DSP.
"Configuring Echo Cancellation" chapter in the "Cisco IOS Voic Port Configuration Guide
12.3(7)T
Interoperability Enhancements to the Cisco Multiservice IP-IP Gateway
Introduces the following enhancements:
(1) IP-to-IP gateway is now interoperable with the Cisco ATA-188 and Microsoft NetMeeting.
(2) Tcl IVR 2.0: Full support for programmable/scripted applications with VoIP endpoints; prepaid applications are one example.
(3) IP-to-IP gateway image consolidation: Combines TDM-to-IP voice gateway and IP-to-IP gateway feature sets in a single Cisco IOS image. The combined functions can run concurrently on the same hardware platform. This reduces the cost of deployment for the IP-to-IP gateway in enterprise networks.
12.3(7)T
IP Communications High-Density Digital Voice/Fax Network Module
Supports high-density digital voice and low-density analog voice connectivity along with data and integrated access connectivity. The network modules offer built-in T1/E1 VIC/VWIC ) slot for FXS, FXO, E&M, Centralized Automatic Message Accounting (CAMA), direct inward dialing (DID), BRI, or E1 and T1 cards, up to a maximum of four T1/E1 ports. Supports up to 32 HDLC channels with an aggregate capacity of 2.048 Mbps.
"Configuring Digital Voice Ports" chapter in the Cisco IOS Voice Port Configuration Guide
12.3(7)T
Lossless Compression R1, ATM Cell Switching and BITS Clocking
Introduces a new compression technique in DSP firmware and add enhancements to Cisco IOS software that include cell switching on ATM segmentation and reassembly (SAR), and the use of an external BITS clocking source. These features enable Cisco multiservice routers to be used to transparently groom and compress traffic in a wireless service provider network and enable a service provider to optimize the bandwidth used to backhaul the traffic from a cell site to the mobile central office for more efficient use of existing T1 and E1 lines.
Lossless Compression R1, ATM Cell Switching and BITS Clocking
12.3(7)T
Signal ISDN B-Channel ID to Enable Application Control of Voice Gateway Trunks
Enables call management applications to identify specific ISDN bearer (B) channels used during a voice gateway call for billing purposes. With the identification of the B channel, SIP and H.323 gateways can enable port-specific features such as voice recording and call transfer.
"Configuring SIP ISDN Support Features" chapter of the Cisco IOS SIP Configuration Guide and "Configuring H.323 Gateway" chapter of the Cisco IOS H.323 Configuration Guide
12.3(7)T
Survivable Remote Site Telephony 3.1
Provides Cisco CallManager with fallback support for Cisco IP phones attached to a Cisco router on a local network. The Cisco Wireless IP Phone 7920 and Cisco IP Phone Conference Station 7936 are fully supported.
12.3(7)T
T.37 Fax Status Notification Enhancement in an MTA Environment
Provides the ability to delegate control of fax operations directly to a mail transfer agent (MTA) by configuring a T.37 fax off-ramp gateway to deliver all fax delivery errors to the fax mail originator in Simple Mail Transfer Protocol (SMTP) delivery status notification (DSN) messages with descriptive error codes.
"Configuring T.37 Store-and-Forward Fax" chapter in the Cisco Fax Services over IP Application Guide
12.3(7)T
VoiceXML Voice Store and Forward
Adds VXML capability on Cisco 2691 router and Cisco 3725 and Cisco 3745 routers.
12.3(4)T
Accounting Server Connectivity Failure and Recovery Detection
Provides the scriptable option to reject new calls entering the VoIP network and tear down all existing calls on detecting connectivity failure to the method list that is associated with RADIUS-based accounting servers.
12.3(4)T
Call Statistics on Voice-Enabled Gateways
Enables the collection of voice call statistics based on user-configured time ranges.
The statistics that can be collected are from the following functional areas:
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"Voice Performance Statistics on Cisco Gateways" chapter in the Cisco IOS Voice Troubleshooting and Monitoring Guide
12.3(4)T
Cisco CallManager Express 3.0
Enables Cisco routers to deliver key system or hybrid PBX functionality for enterprise branch offices or small businesses. Cisco CME is ideal for customers who have data connectivity requirements and also have a need for a telephony solution in the same office. Whether offered through a service provider's managed services offering or purchased directly by a corporation, Cisco CME offers most of the core telephony features required in the small office, in addition to many advanced features not available with traditional telephony solutions. Being able to deliver IP telephony and data routing using a single converged solution allows customers to optimize their operations and maintenance costs, resulting in a very cost-effective solution that meets office needs.
Note
12.3(4)T
Customizable Tone Download to Cisco IOS MGCP Gateways from Cisco Call Manager
Implements the downloading of region-specific tones and the associated frequencies, amplitudes, and cadences using XML-based configuration files during gateway registration. The feature also introduces the ability to play out tones that contain up to four frequencies.
"Configuring Tone Download to MGCP Gateway" chapter in the Cisco CallManager and Cisco IOS Interoperability Guide
12.3(4)T
Enhanced ITU-T G.168 Echo Cancellation
Extends the existing ITU-T G.168 Echo Cancellation functionality by adding support for medium complexity CODECs.
"Configuring Echo Cancellation" chapter in the "Cisco IOS Voic Port Configuration Guide
12.3(4)T
Inactive Call Detection
Detects inactive (silent) H.323 or SIP call-legs on Cisco IOS software-based gateways, and reports this situation to the Tcl IVR 2.0 application (which can disconnect the call). Enables the Cisco IOS software to not automatically disconnect detected inactive calls. Inactivity is defined as no RTP/RTCP packets for a configurable length of time.
"Troubleshooting Voice Applications" chapter of the Cisco IOS Voice Troubleshooting and Monitoring Guide
12.3(4)T
IP Communications Voice/Fax Network Module
Provides the ability to directly connect the PSTN and legacy telephony equipment to Cisco 2600XM series, Cisco 2691, Cisco 3600 series, and Cisco 3700 series multiservice routers, enabling applications such as IP telephony, toll bypass, and full gateway integration. The VWICs supported by the new network modules include 2- and 4-port FXS; 2- and 4-port FXO; 2-port DID, E&M, and BRI (S/T); and 1- and 2-port T1/E1.
12.3(4)T
ISDN Calling Name Display
Provides calling name display to SIP customers on calls that originate on ISDN networks.
"Configuring SIP ISDN Support Features" chapter of the Cisco IOS SIP Configuration Guide
12.3(4)T
MGCP Gateway Support for Cisco CallManager Network Specific Facilities
Provides an enhancement to Cisco CallManager functionality enabling users to configure the NSF ISDN information element of the route pattern. This feature is compatible with Cisco CallManager Version 3.3(2) and later.
"Configuring MGCP PRI Backhaul and T1 CAS Support for Cisco CallManager" chapter in the Cisco CallManager and Cisco IOS Interoperability Guide
12.3(4)T
PLAR (Private Line Automatic Ring-down) for Trading Turrets
Re-establishes an H.323 call automatically if one end hangs up or goes on hold and then goes off-hook again. This provides PLAR functionality used in the financial industry for trading Turret operation of PLAR.
12.3(4)T
SIP Debug Output Filtering Support
Provides the ability to filter relevant SIP debugging traces of desired calls.
12.3(4)T
SIP Gateway Support Enhancements to the bind Command
Allows configuration of the source IP address of signaling packets or both signaling and media packets using the bind command.
"Configuring SIP Gateway Support for the bind Command" chapter of the Cisco IOS SIP Configuration Guide
12.3(4)T
SIP Header Support and SUBSCRIBE and NOTIFY for External Triggers
Provides a mechanism for applications to send and receive SIP headers and to send SUBSCRIBE messages and receive NOTIFY events.
"Configuring Additional SIP Application Support" chapter of the Cisco IOS SIP Configuration Guide
12.3(4)T
SIP NOTIFY-Based Out-of-Band DTMF Relay Support
Supports SCCP devices through SIP originating and terminating gateway use of Cisco proprietary NOTIFY-based out-of-band DTMF relay, which can also be used by analog phones attached to analog voice ports (FXS) on a router.
"SIP NOTIFY-Based Out-of-Band DTMF Relay Support" chapter of the Cisco IOS SIP Configuration Guide
12.3(4)T
SIP Redirect Processing Enhancement
Allows flexibility in the handling of incoming redirect or 3xx class of responses. Redirect responses can now be enabled or disabled through the command-line interface, providing critical functionality for service providers who deploy Cisco gateways.
"Basic SIP Configuration" chapter of the Cisco IOS SIP Configuration Guide
12.3(4)T
SIP Register Support
Allows SIP gateways to register E.164 numbers to a SIP proxy or registrar on behalf of analog telephone voice ports (FXS), IP phone virtual voice ports (EFXS), and local SCCP phones.
"Basic SIP Configuration" chapter of the Cisco IOS SIP Configuration Guide
12.3(4)T
SIP RFC 3261 Enhancements
Provides enhanced SIP functionality on Cisco IOS gateways, as defined by RFCs 3261 and 3311, and support for the following:
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"Basic SIP Configuration" chapter of the Cisco IOS SIP Configuration Guide
12.3(4)T
SIP Survivable Remote Site Telephony (SRST) Version 3.0
Describes Survivable Remote Site Telephony (SRST) functionality for Session Initiation Protocol (SIP) networks. SIP SRST provides backup to an external SIP proxy server by providing basic registrar and redirect services. These services are used by a SIP IP phone in the event of a WAN connection outage where the SIP phone is unable to communicate with its primary SIP proxy. The SIP SRST device also provides PSTN gateway access for placing and receiving PSTN calls.
12.3(4)T
SRST: Survivable Remote Site Telephony Version 3.0
Provides Cisco CallManager with fallback support for Cisco IP phones attached to a Cisco router on your local network. Cisco SRST enables routers to provide call-handling support for Cisco IP phones when they lose connection to remote primary, secondary, or tertiary Cisco CallManager installations or when the WAN connection is down.
Cisco CallManager supports Cisco IP phones at remote sites attached to Cisco multiservice routers across the WAN. Prior to Cisco SRST, when the WAN connection between a router and the Cisco CallManager failed or connectivity with Cisco CallManager was lost for some reason, Cisco IP phones on the network became unusable for the duration of the failure. Cisco SRST overcomes this problem and ensures that the Cisco IP phones offer continuous (although minimal) service by providing call-handling support for Cisco IP phones directly from the Cisco 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 IP phones that are registered with the router. When the WAN link or connection to the primary Cisco CallManager is restored, call handling reverts back to the primary Cisco CallManager.
12.3(4)T
Tcl IVR 2.0 Session Interaction
Allows different instances of Tcl IVR applications (sessions) to communicate with other sessions on the same gateway and for applications to dynamically bridge call legs between different sessions. This enables different callers on the same gateway to be notified of each others' presence and to interact. You can also start a session without an active call leg so that a session can act as an application service for other sessions. This feature is useful for implementing a call-monitoring "server" application that is responsible for monitoring incoming calls and dynamically connecting selected callers.
"Configuring Tcl IVR 2.0 Session Interaction" chapter of the Cisco IOS Tcl IVR and VoiceXML Applications Guide
12.3(4)T
Videoconferencing for the Cisco Multiservice IP-to-IP Gateway
Provides enhanced Quality of Service through RSVP synchronization with H.323 signaling protocol and DSCP packet marking.
"Configuring Multiservice IP-to-IP Videoconferencing" chapter in the Cisco Multiservice IP-to-IP Gateway
12.3(4)T
Voice DSP Crash Dump Analysis
Allows Cisco IOS voice platforms using TI DSPs the ability to capture the contents of the DSP memory into a dump file if there is a DSP crash.
"Troubleshooting Digital Voice Interfaces to the IP Newtwork" chapter in th Cisco IOS Voice Troubleshooting and Monitoring Guide
12.3(4)T
VoIP Debug Filtering
Allows you to filter and trace voice call debug messages based on selected filtering criteria, reducing the volume of output for more efficient troubleshooting.
"Filtering Troubleshooting Output" chapter of the Cisco IOS Voice Troubleshooting and Monitoring Guide
12.3(4)T
VoIP Internal Error Codes
Generates internal error codes (IECs) for gateway-detected errors that cause the gateway to release or refuse a call. IECs enhance troubleshooting for VoIP networks by helping to determine the source and reason for call termination.
"Cisco VoIP Intewrnal Error Codes" chapter in the Cisco IOS Voice Troubleshooting and Monitoring Guide
12.3(2)T
MGCP-Controlled Backhaul of BRI Signaling in Conjunction with Cisco CallManager
Provides MGCP service to remote-office media gateways that connect by means of ISDN BRI trunks to a centralized Cisco CallManager media-gateway controller for the purpose of call processing. D-channel signal information is backhauled to the call manager through a Transmission Control Protocol (TCP) session.
"Configuring MGCP-Controlled Backhaul of BRI Signaling in Conjunction with Cisco CallManager" chapter in the Cisco CallManager and Cisco IOS Interoperability Guide.
12.3(2)T
PCR Support for the Cisco Signaling Link Terminal
Adds support for Message Transfer Part Layer 2 (MTP2) PCR on the Cisco signaling link terminal (SLT). Preventive Cyclic Retransmission (PCR) adds basic MTP2 functionality that is used when Signaling System 7 (SS7) signaling links are transmitted over satellite connections between the Cisco SLT and a signal transfer point (STP).
12.3(2)T
Persistent TDM Switched Circuits
Enables the Cisco AS5850 universal gateway to connect one or more DS0s from one or more E1s or T1s to another set of E1s or T1s.
12.3(2)T
V.110 Support for MGCP-Dial
Provides support for V.110 encapsulation for Dial-MGCP Applications.
"Configuring NAS Package for MGCP" chapter of the Cisco IOS MGCP and Related Protocols Configuration Guide
Guest
