Cisco Unified SRST 14.4 Release introduces support for the following new features:

• Secure SIP SRST and Secure SCCP SRST supports TLS version 1.3 ciphers.

• SHA2 ciphers support with TLS version 1.3 for secure SCCP SRST.

• Note	Only SCCP Analog Voice Gateways support TLS version 1.3. The SCCP IP phone endpoints do not support TLS version 1.3.

  For configuration information, see Configure Secure SRST for SCCP and SIP.

Cisco Unified SRST 14.3 Release introduces support for the following new features:

• Webex Survivability Gateway—From Cisco IOS XE 17.9.3 and Cisco IOS XE Dublin 17.11.1a onward, configure a Webex Survivability Gateway to provide an on-site calling fallback service for Webex Calling endpoints. This feature also supports the colocation of a Webex Calling survivability configuration and a Unified SRST configuration on the same router.

  For general information, see Webex Survivability Gateway Mode.

  To configure a Survivability Gateway, see Site Survivability for Webex Calling.

• SFTP CDR Transfer for File Accounting — Allows transfer of SRST CDRs using SFTP. See Configuring File Accounting.

Cisco Unified SRST 14.2 Release introduces support for the following new features:

• Restrict Secure SIP SRST and Secure SCCP SRST to only using TLS 1.2 SHA2 Cipher Suites—SHA2-Cipher-Only Mode for Unified Secure SRST

• SIP OAuth Support for Secure SRST—SIP OAuth Client Registration for Unified Secure SRST

Unified SRST 14.1 Release introduces support for the following new features:

• Voice: Class of Restriction YANG Configuration Model—Programmability Guide for Cisco IOS XE Unified Communications VoIP Products

• Smart Licensing Using Policy—Cisco Smart Licensing for Unified SRST

• Smart Licensing Using Policy—Cisco Smart Licensing for Unified E-SRST

Unified SRST 12.7 Release introduces support for the following new feature:

• Support for maximum number of devices in Cisco 4451 and 4461 Integrated Services Routers was increased from 1500 to 2000.

Cisco Unified SRST 12.6 Release introduces support for the following new features:

• Simple Network Management Protocol (SNMP) Support for Cisco Unified SRST

• Toll Fraud Prevention for SIP Line Side on Cisco Unified SRST

• Cisco Unified SRST, Unified E-SRST, and Unified Secure SRST Password Policy

Cisco Unified SRST 12.3 Release introduces support for Secure SCCP SRST on Cisco 4000 Series Integrated Services Router.

Cisco Unified SRST 12.2 Release introduces support for Unified E-SRST with Support for Voice Hunt Group.

Cisco Unified SRST 12.1 introduces support for the following new features:

• Licensing

• Secure SIP SRST Support on Cisco 4000 Series Integrated Services Router

• Cisco Unified SRST and Cisco Unified Border Element co-location

Cisco Unified SRST 12.0 introduces support for IPv6 protocols on SIP IP Phones. For more information on IPv6 Support introduced for Cisco Unified SRST, see IPv6 Support for Cisco Unified SRST SIP IP Phones.

Cisco Unified SRST 11.0 supports the following new Cisco IP phones and adapters:

• Support for Cisco IP Phone 7811

• Support for Cisco IP Phones 8811, 8831, 841, 8851, 8851NR, 8861

• Support for Cisco ATA-190

For information on the phones supported in Cisco Unified SRST 11.0, see Phone Feature Support Guide for Cisco Unified Communications Manager Express, Cisco Unified SRST, Unified E-SRST, and Unified Secure SRST.

Cisco Unified SRST 10.5 supports the following features:

• Where to Go Next, Setting Up the Network

For more information on the Cisco Unified SRST 10.5 supported feature, see the SCCP: Configure Unified E-SRST.

Cisco Unified SRST 10.5 supports the following new Cisco Unified SIP IP phones:

• Support for Cisco Unified DX650 SIP IP Phones

• Support for Cisco Unified 78xx SIP IP Phones

Cisco Unified SRST 10.0 supports the following new features:

• Cisco Jabber for Windows

• SIP: Configure Unified E-SRST

To obtain an account on Cisco.com, go to www.cisco.com and click Register at the top of the screen.

In Cisco Unified SRST 9.5, support for afterhours pattern blocking is extended to regular expression patterns for dial plans on Cisco Unified SIP and Cisco Unified SCCP IP phones. With this support, users can add a combination of fixed dial plans and regular expression-based dial plans.

When a call is initiated after hours, the dialed number is matched against a combination of dial plans. If a match is found, the call is blocked.

To enable regular expression patterns to be included when configuring afterhours pattern blocking, the after-hours block pattern command is modified to include regular expressions as a value for the pattern argument in the following command syntax:

after-hours block pattern pattern-tag pattern

This command is available in the following configuration modes:

• telephony-service—For both SCCP and SIP Phones.

• ephone-template—For SCCP phones only.

Note	The maximum length of a regular expression pattern is 32 for both Cisco Unified SIP and Cisco Unified SCCP IP phones.

If calls to the following numbers are to be blocked after hours:

• numbers beginning with ‘0’ and ‘00’

• numbers beginning with 1800, followed by four digits

• numbers 9876512340 to 9876512345

then the following configurations can be used:

• after-hours block pattern 1 0*

• after-hours block pattern 2 00*

• after-hours block pattern 3 1800….

• after-hours block pattern 4 987651234[0-5]

Note	There is no change in the number of afterhours patterns that can be added. The maximum number is still 100.

For more information on configuration examples, see the “Configuring Afterhours Block Patterns of Regular Expressions: Example” section of Cisco Unified Communications Manager Administration Guide.

For a summary of the basic Cisco IOS regular expression characters and their functions, see the Cisco Regular Expression Pattern Matching Characters section of Terminal Services Configuration Guide.

Before Cisco Unified CME 9.5, a parked call could not be recalled by or transferred to the phone that put the call in park or the original phone that transferred the call when the destination phone was offhook or ringing.

In Cisco Unified CME 9.5, the recall force keyword is added to the call-park system command in telephony-service configuration mode to allow a user to force the recall or transfer of a parked call to the phone that put the call in park or the phone with the reserved-for number as its primary DN when the destination phone is available to answer the call.

In Cisco Unified CME 10.5, a new ring tone is introduced for park recall to assist the phone user to distinctly identify the type of call.

This feature is supported on all phone families for SCCP endpoints and on 89XX and 99XX phone families for SIP endpoints. No configurations are required to activate this feature.

Router# configure terminal
Router(config)# telephony-service
Router(config)# srst mode auto-provision all
Router(config-telephony)# call-park system ? recall Configure parameters for recall
Router(config-telephony)# call-park system recall ? force Force recall for busy call park
initiator
Router(config-telephony)# call-park system recall force

In Cisco Unified CME 8.5 and later versions, the park monitor feature allows you to park a call and monitor the status of the parked call until the parked call is retrieved or abandoned. When a Cisco Unified SIP IP Phone 8961, 9951, or 9971 parks a call using the park soft key, the park monitoring feature monitors the status of the parked call. The park monitoring call bubble is not cleared until the parked call gets retrieved or is abandoned by the parkee. This parked call can be retrieved using the same call bubble on the parker’s phone to monitor the status of the parked call.

Once a call is parked, Cisco Unified CME sends a SIP NOTIFY message to the parker phone indicating the “parked” event along with the park slot number so that the parker phone can display the park slot number as long as the call remains parked.

When a parked call is retrieved, Cisco Unified CME sends another SIP NOTIFY message to the parker phone indicating the “retrieved” event so that the phone can clear the call bubble. When a parked call is disconnected by the parkee, Cisco Unified CME sends a SIP NOTIFY message to the parker phone indicating the “abandoned” event and the parker phone clears the call bubble upon cancellation of the parked call.

When a parked call is recalled or transferred, Cisco Unified CME sends a SIP NOTIFY message to the parker phone indicating the “forwarded” event so that parker phone can clear the call bubble during park, recall, and transfer. You can also retrieve a parked call from the parker phone by directly selecting the call bubble or pressing the resume soft key on the phone.

The transfer-pattern command for Cisco Unified SIP IP phones functions like the transfer-pattern command for Cisco Unified SCCP IP phones by allowing all, not just local, transfers to take place.

The transfer-pattern command specifies the directory numbers for Call Transfer. The command can be configured up to 32 times using the following command syntax: transfer-pattern transfer-pattern [ blind ].

Note	The blind keyword in the transfer-pattern command applies only to Cisco Unified SCCP IP phones and does not apply to Cisco Unified SIP IP phones.

With the transfer-pattern command configured, only Call Transfers to numbers that match the configured transfer pattern are allowed to take place. With the transfer pattern configured, all or a subset of transfer numbers can be dialed and the transfer to a remote party can be initiated.

The following are examples of configurable transfer patterns:

• .T—This configuration allows Call Transfers to any destinations with one or more digits, like 123, 877656, or 76548765.

• 919........—This configuration only allows Call Transfers to remote numbers beginning with “919” and followed by eight digits, like 91912345678. However, Call Transfers to 9191234 or 919123456789 are not allowed.

To maintain backward compatibility, all Call Transfers from Cisco Unified SIP IP phones to any number (local or over the trunk) are allowed when no transfer patterns are configured through the transfer-pattern , transfer-pattern blocked , or transfer max-length commands.

For Cisco Unified SCCP IP phones, if you do not configure transfer patterns, Call Transfers over the trunk are blocked.

The transfer max-length command is used to indicate the maximum length of the number being dialed for Call Transfer. When only a specific number of digits are allowed during a Call Transfer, value from 3 through 16 is configured. When the number dialed exceeds the maximum length, then the Call Transfer is blocked.

For example, if you configure 5 as the maximum length, Call Transfers from Cisco Unified SIP IP phones allows up to a five-digit directory number. All Call Transfers to directory numbers with more than five digits are blocked.

Note	If only transfer max length is configured and conference max-length is not configured, then transfer max length takes effect for transfers and conferences.

When the transfer-pattern blocked command is configured for a specific phone, no Call Transfers are allowed from that phone over the trunk.

This feature forces unconditional blocking of all Call Transfers from the specific phone to any other nonlocal numbers (external calls from one trunk to another trunk). No Call Transfers from this specific phone are possible even when a transfer pattern matches the dialed digits for transfer.

The following table compares the behaviors of Cisco Unified SCCP and SIP IP phones for specific configurations.

The conference-pattern blocked command is used to prevent extensions on a voice register Pool from initiating conferences.

The following table summarizes the behavior of the conference-pattern blocked command in relation to no conference-pattern blocked ,conference max-length , no conference max-length , and transfer max-length commands.

When the transfer-pattern blocked command is configured for a specific phone, no Call Transfers are allowed from that phone over the trunk.

This feature forces unconditional blocking of all Call Transfers from the specific phone to any other nonlocal numbers (external calls from one trunk to another trunk). No Call Transfers from this specific phone are possible even when a transfer pattern matches the dialed digits for transfer.

The following table compares the behaviors of Cisco Unified SCCP and SIP IP phones for specific configurations.

When both the transfer-pattern and conference transfer-pattern commands are configured and dialed digits match the configured transfer pattern, conference calls are allowed. However, when the dialed digits do not match the configured transfer pattern, the conference call is blocked.

For information on provisioning Cisco Unified IP phones for secure access to web content using HTTPS, see the HTTPS Provisioning for Cisco Unified IP Phones section of Cisco Unified Communications Manager Express System Administrator Guide.

For configuration examples, see the Configuring HTTPS Support for Cisco Unified Communications Manager Express: Example section of Cisco Unified Communications Manager Administration Guide.

Cisco Unified SRST 9.1 supports the following new features:

• KEM Support for Cisco Unified SIP IP Phones

• Enhancement in Speed-Dial Support

• Voice Hunt Group Support

Note

If you have older routers, such as the VG26nn and VG37nn platforms and Cisco Integrated Services Router (ISR) Generation 1 platforms (Cisco ISR 1861, 2800, and 3800 Series), you must upgrade to Cisco ISR 881, 886VA, 887VA, 888, 888E, 1861E, 2900, 3900, and 3900E Series platforms to utilize these new features.

For information on feature support for the Cisco Unified 6901 and 6911SIP IP Phones in Cisco Unified SRST, see Phone Feature Support Guide for Cisco Unified Communications Manager Express, Cisco Unified SRST, Unified E-SRST, and Unified Secure SRST.

For information on feature support for the Cisco Unified 6921, 6941, 6945, and 6961 SIP IP Phones in Cisco Unified SRST, see Phone Feature Support Guide for Cisco Unified Communications Manager Express, Cisco Unified SRST, Unified E-SRST, and Unified Secure SRST.

For information on feature support for the Cisco Unified 8941 and 8945 SIP IP Phones in Cisco Unified SRST, see Phone Feature Support Guide for Cisco Unified Communications Manager Express, Cisco Unified SRST, Unified E-SRST, and Unified Secure SRST.

Cisco Unified SRST 9.0 supports the Multiple Calls Per Line (MCPL) feature on Cisco Unified 6921, 6941, 6945, and 6961 SIP IP phones. In addition, it supports Cisco Unified 8941, 8945 SCCP, and SIP IP phones.

Before Cisco Unified SRST 9.0, supports only two calls for every directory number (DN) on Cisco Unified 8941 and 8945 SCCP IP phones.

With Cisco Unified SRST 9.0, the MCPL feature overcomes the limitation on the maximum number of calls per line.

Cisco Unified SRST 9.0 does not support the MCPL feature on Cisco Unified 6921, 6941, 6945, and 6961 SCCP IP phones. Allows only two calls on these phones whereas allows only one call on octo-line directory numbers on these phones before activating Call Forward Busy or busy tone.

Before Cisco Unified SRST 9.0, the values for the max-dn and timeouts busy commands were hardcoded for Cisco Unified 8941 and 8945 SCCP IP phones.

In Cisco Unified SRST 9.0, you can configure the max-dn andtimeouts busy commands in call-manager-fallback configuration mode. Use the max-dn command to set the maximum number of DNs that can be supported by the router and enable dual-line mode, octo-line mode, or both modes. Use the timeouts busy command to set the timeout value for Call Transfers to busy destinations.

For configuration information, see Configuring the Maximum Number of Calls.

In Cisco Unified SRST 9.0, the maximum number of calls for Cisco Unified 6921, 6941, 6945, 6961, 8941, and 8945 SIP IP phones is controlled by the phones.

Prerequisites

• Cisco Unified SRST 9.0 and later versions.

• Correct firmware is installed:

  • 9.2(1) or a later version for Cisco Unified 6921, 6941, 6945 and 6961 SIP IP phones.

  • 9.2(2) or a later version for Cisco Unified 8941 and 8945 SIP IP phones.

Cisco ATA-187 is a SIP-based analog phone adapter that turns traditional phone devices into IP devices. Cisco ATA-187 can connect with a regular analog FXS phone or fax machine on one end, while the other end is an IP side that uses SIP for signaling and registers as a Cisco Unified SIP IP phone.

Cisco ATA-187 functions as a Cisco Unified SIP IP phone that supports T.38 fax relay and fax pass-through, enabling the real-time transmission of fax over IP networks. The fax rate is from 7.2 to 14.4 kbps.

For information on feature support for the Cisco ATA-187 in Cisco Unified SRST, see Phone Feature Support Guide for Cisco Unified Communications Manager Express, Cisco Unified SRST, Unified E-SRST, and Unified Secure SRST.

For more information on Cisco ATA-187, see Cisco ATA 187 Analog Telephone Adaptor Administration Guide for SIP.

Cisco Unified SRST 8.8 supports the following new Cisco Unified SCCP IP phones:

• Cisco Unified 6945 SCCP IP Phones

• Cisco Unified 8941 SCCP IP Phones

• Cisco Unified 8945 SCCP IP Phones

Beginning with Cisco IP Phone firmware 8.5(3) and Cisco IOS Release 15.1(1)T, Cisco Unified SRST supports SIP signaling over UDP, TCP, and TLS connections, providing both RTP and SRTP media connections based on the security settings of the IP phone.

Cisco Unified SRST 7.0/4.3 supports the following new features:

• Configuring Eight Calls per Button (Octo-Line)

• Configuring Consultative Transfer

Cisco Unified SRST Version 4.2(1) introduces the new feature enhancements for Enhanced 911 Services.

Cisco Unified SRST Version 4.1 introduces the following new feature:

• Enhanced 911 Services

The following IP phones are supported with Cisco Unified SRST systems:

• Cisco Unified IP Phone 7911G

• Cisco Unified IP Phone 7941G and Cisco Unified IP Phone 7941G-GE

• Cisco Unified IP Phone 7960G

• Cisco Unified IP Phone 7961G and Cisco Unified IP Phone 7961G-GE

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 Cisco IP Phone 7914 Expansion Module Quick Start Guide.

No additional SRST configuration is required for these phones.

The show ephone command is 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 Cisco Unified SRST and Cisco Unified SIP SRST Command Reference (All Versions).

To determine compatible firmware, platforms, memory, and additional voice products that are associated with Cisco Unified SRST 4.0, see Cisco Unified SRST 4.3 Supported Firmware, Platforms, Memory, and Voice Products.

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 pass-through 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.

Note

For ATAs that are registered to a Cisco Unified SRST system to participate in FAX calls, they must have their ConnectMode parameter set to use the “standard payload type 0/8” as the RTP payload type in FAX pass-through mode. For ATAs used with Cisco Unified SRST 4.0 and higher versions, this is done by setting bit 2 of the ConnectMode parameter to 1 on the ATA. For more information, see the “Parameters and Defaults” chapter in Cisco ATA 186 and Cisco ATA 188 Analog Telephone Adaptor Administrator's Guide for SCCP.

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 Communications Manager, 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. H.323 VoIP call preservation enhancements does not support SIP Phones.

For configuration information see the “Configuring H.323 Gateways” chapter in Cisco IOS H.323 Configuration Guide.

This feature allows you to set video parameters for the Cisco Unified SRST to maintain close feature parity with Cisco Unified CM. 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 CM. 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.

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 call control (IP-PBX) 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 Cisco SIP SRST Version 3.4 System Administrator Guide.

Secure Cisco IP phones that are located at remote sites and that are attached to gateway routers can communicate securely with Cisco Unified Communications Manager using the WAN. But if the WAN link or Cisco Unified Communications Manager 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 Communications Manager goes down. When the WAN link or Cisco Unified Communications Manager is restored, Cisco Unified Communications Manager 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 Configure Secure SRST for SCCP and SIP.

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 customizable 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 Cisco Unified IP Phone 7900 Series documentation index.

Note	The Cisco Unified IP Phone 7914 Expansion Module can attach to your Cisco Unified IP Phones 7970G and 7971G-GE. See the Cisco Unified IP Phone Expansion Module 7914 Support section for more information.

The show ephone command is 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 Cisco Unified SRST and Cisco Unified SIP SRST Command Reference (All Versions).

The alias command is enhanced as follows:

• The cfw keyword was added, providing call forward no-answer/busy capabilities.

• The maximum number of alias commands used for creating calls to telephone numbers that are unavailable during Cisco Unified Communications Manager fallback was increased to 50.

• The alternate-number argument can be used in multiple alias commands.

For more information, see the alias command in Cisco Unified SRST and Cisco Unified SIP SRST Command Reference (All Versions).

The maximum number of cor lists has increased to 20.

For more information, see the cor command in Cisco Unified SRST and Cisco Unified SIP SRST Command Reference (All Versions).

The pickup command was 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 Cisco Unified SRST and Cisco Unified SIP SRST Command Reference (All Versions).

The user-locale command is enhanced to display the Japanese Katakana country code. Japanese Katakana is available in Cisco Unified Communications Manager V4.0 or later versions.

For more information, see the user-locale command in the Cisco Unified SRST and Cisco Unified SIP SRST Command Reference (All Versions).

The Cisco 3845 now supports 720 phones and up to 960 ephone-dns or virtual voice ports.

Cisco Unified SRST is 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 adaptor 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 the Integrating Cisco Unified Communications Manager and Cisco Unified SRST to Use Cisco Unified SRST as a Multicast MOH Resource section for configuration instructions.

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 versions. See the Cisco Unified SRST Feature Overview section for more information.

H.323 is not supported with SIP phones.

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 the How to Configure DTMF Relay for SIP Applications and Voicemail section for configuration instructions.

To use voicemail on a SIP network that connects to a Cisco Unity Express 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.

Cisco SRST 3.2 and later versions support translation profiles. Translation profiles allow you to group translation rules together and to associate translation rules with the following:

• Called numbers

• Calling numbers

• Redirected called numbers

See the Enabling Translation Profiles section for more configuration information. For more information on the translation-profile command, see Cisco Unified SRST and Cisco Unified SIP SRST Command Reference (All Versions).

Cisco Unified SRST V3.1 introduced the new features described in the following sections:

• Cisco Unified IP Phone 7920 Support

• Cisco Unified IP Phone 7936 Support

Note	For information about Cisco Unified IP phones, see the Cisco Unified IP Phone 7900 Series documentation.

Displays for the Cisco Unified IP Phone 7940G and Cisco Unified IP Phone 7960G can be configured with extra ISO-3166 codes for German, Danish, Spanish, French, Italian, Japanese, Dutch, Norwegian, Portuguese, Russian, Swedish, United States.

Note	This feature is available only for Cisco Unified SRST running under Cisco Unified Communications Manager V3.2.

Cisco Unified SRST V1.0, Cisco Unified SRST V2.0, and Cisco Unified 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 Unified SRST use a Cisco Unified SRST proprietary mechanism to perform blind transfers. Cisco Unified 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 Unified 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 the PSTN interface type.

Note

All voice gateway routers in the VoIP network must support H.450. For H.450 support, routers with Cisco Unified SRST must run either Cisco Unified SRST V3.0 and higher versions or Cisco IOS Release 12.2(15)ZJ and later releases. Routers without Cisco Unified SRST must run either Cisco Unified SRST V2.1 and higher versions or Cisco IOS Release 12.2(11)YT and later releases. SIP phones do not support this feature.

For more information about the default session application, see the Default Session Application Enhancements Guide.

For configuration information, see the Enabling Consultative Call Transfer and Forward Using H.450.2 and H.450.3 with Cisco Unified SRST 3.0 section.

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.

A new keyword that was 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 Unified SRST fallback. The max-dn command affects all IP phones on a Cisco Unified SRST router.

For configuration information, see the Configuring Dual-Line Phones section.

Cisco Unified 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 several countries and geographic regions implement R2 in entirely different ways. Cisco addresses this challenge by supporting many localized implementations of R2 signaling in its Cisco IOS Software.

The Cisco 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 that there are multiple R2 signaling types in the specified country, but Cisco supports the ITU variant.

Cisco also supports specific local variants of E1 R2 signaling in the following regions, countries, and corporations:

• Argentina

• Australia

• Bolivia

• Brazil

• Bulgaria

• China

• Colombia

• Costa Rica

• East Europe (includes Croatia, Russia, and Slovak Republic)

• Ecuador (ITU)

• Ecuador (LME)

• Greece

• Guatemala

• Hong Kong (uses the China variant)

• Indonesia

• Israel

• Korea

• Laos

• Malaysia

• Malta

• New Zealand

• Paraguay

• Peru

• Philippines

• Saudi Arabia

• Singapore

• South Africa (Panaftel variant)

• Telmex Corporation (Mexico)

• Telnor Corporation (Mexico)

• Thailand

• Uruguay

• Venezuela

• Vietnam

The date format on a Cisco IP phone display can be configured with the following two extra formats:

• yy-mm-dd (year-month-day)

• yy-dd-mm (year-day-month)

For configuration information, see the Configuring IP Phone Clock, Date, and Time Formats section.

A new keyword was 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.

You can configure Cisco Unified SRST to support continuous multicast output of MOH from a flash MOH file in flash memory.

For more information, see the Defining XML API Schema section.

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.

Secondary dial tone is available for Cisco Unified IP Phones running Cisco Unified SRST. The secondary dial tone is generated when you dial a predefined PSTN access prefix. For example, you would hear different dial tone when a designated number is pressed to reach an outside line.

The secondary dial tone is created through the secondary dial tone command. For more information, see the Configuring a Secondary Dial Tone section.

The show ephone command is enhanced to display the following:

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

• Status of all phones with the call-forwarding all (CFA) feature enabled on at least one of their DNs (new keyword: cfa )

For more information, see the show ephone command in Cisco Unified SRST and Cisco Unified SIP SRST Command Reference (All Versions).

Diagnostic messages are added to the system log whenever a phone registers or unregisters from Cisco Unified SRST.

Cisco Unified SRST supports three-party instant meeting conferencing using the G.711 coding technique. For conferencing to be available, connect two lines to one or more buttons of an IP phone.

For more information, see the Enabling Three-Party G.711 Ad Hoc Conferencing section.

The Cisco VG248 Analog Phone Gateway is a mixed-environment solution, enabled by Cisco Unified Communications system. It allows organizations to support their legacy analog devices while taking advantage of the new opportunities afforded by using IP telephony. The Cisco VG248 is a high-density gateway for using analog phones, fax machines, modems, voicemail systems, and speakerphones within an enterprise voice system based on Cisco Unified Communications Manager.

During Cisco Unified Communications Manager fallback, Cisco Unified SRST considers the Cisco VG248 to be a group of Cisco Unified IP Phones. Cisco Unified 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 versions is also available in Cisco Unified SRST Version 2.1.

For more information, see Cisco VG248 Analog Phone Gateway Data Sheet and Cisco VG248 Analog Phone Gateway Version 1.2(1) Release Notes.

Cisco SRST V2.1 introduced the new features described in the following sections:

• Additional Language Options for IP Phone Display

• Cisco SRST Aggregation

• Cisco ATA 186 and ATA 188 Support

• Cisco Unified IP Phone 7902G Support

• Cisco Unified IP Phone 7905G Support

• Cisco Unified IP Phone 7912G Support

• Cisco Unified IP Phone Expansion Module 7914 Support

• Enhancement to the dialplan-pattern Command

Note	For information about Cisco Unified IP phones, see the Cisco Unified IP Phone 7900 Series documentation.

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 CM for connection services and receives the appropriate endpoint phone number and any software enhancements or personalized settings, which are preloaded within Cisco Unified CM.

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 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.

The following table shows the increases in directory numbers.

Cisco Unity voicemail and other voicemail systems can be integrated with Cisco Unified SRST. Voicemail integration introduces six new commands:

• Pattern direct

• Pattern ext-to-ext busy

• Pattern ext-to-ext no-answer

• Pattern trunk-to-ext busy

• Pattern trunk-to-ext no-answer

• Vm-integration