-
Cisco Unified SCCP and SIP SRST System Administrator Guide (All Versions)
-
Cisco Unified SCCP and SIP SRST Feature Overview
-
Cisco Unified Survivable Remote Site Telephony Feature Roadmap
-
Setting Up the Network
-
Cisco Unified SIP SRST 4.1
-
Setting Up Cisco Unified IP Phones using SCCP
-
Setting Up Cisco Unified IP Phones using SIP
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Configuring Call Handling
-
Configuring Secure SRST for SCCP and SIP
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Integrating Voice Mail with Cisco Unified SRST
-
Setting Video Parameters
-
Monitoring and Maintaining Cisco Unified SRST
-
Appendix A: Configuring Cisco Unified SIP SRST Features Using Redirect Mode
-
Appendix B: Integrating Cisco Unified Communications Manager and Cisco Unified SRST to Use Cisco Unified SRST as a Multicast MOH Resource
-
Feedback
Table Of Contents
Cisco Unified Survivable Remote Site Telephony Feature Roadmap
Information About New Features in Cisco Unified SRST
New Features in Cisco Unified SRST Version 9.5
Afterhours Pattern Blocking Support for Regular Expressions
Display Support for Name of Called Voice Hunt Groups
Preventing Local-Call Forwarding to Final Agent in Voice Hunt Groups
Trunk-to-Trunk Transfer Blocking for Toll Fraud Prevention on Cisco Unified SIP IP Phones
New Features in Cisco Unified SRST Version 9.1
KEM Support for Cisco Unified 8961, 9951, and 9971 SIP IP Phones
Enhancement in Speed-Dial Support
New Features in Cisco Unified SRST Version 9.0
Support for Cisco Unified 6901 and 6911 SIP IP Phones
Support for Cisco Unified 6921, 6941, 6945, and 6961 SIP IP Phones
Support for Cisco Unified 8941 and 8945 SIP IP Phones
Voice and Fax Support on Cisco ATA-187
New Features in Cisco Unified SRST Version 8.8
Support for Cisco Unified 6945, 8941, and 8945 SCCP IP Phones
New Features in Cisco Unified SRST Version 8.0
New Features in Cisco Unified SRST Version 7.0/4.3
New Features in Cisco Unified SRST Version 4.2(1)
New Features in Cisco Unified SRST Version 4.1
New Features in Cisco Unified SRST Version 4.0
Additional Cisco Unified IP Phone Support
Fax Passthrough using SCCP and ATAs Support
H.323 VoIP Call Preservation Enhancements for WAN Link Failures for SCCP Phones
New Features in Cisco Unified SRST Version 3.4
New Features in Cisco SRST Version 3.3
Cisco Unified IP Phone 7970G and Cisco Unified 7971G-GE Support
Enhancement to the show ephone Command
New Features in Cisco SRST Version 3.2
Enhancement to the alias Command
Enhancement to the cor Command
Enhancement to the pickup Command
Enhancement to the user-locale Command
Increased the Number of Cisco Unified IP Phones Supported on the Cisco 3845
No Timeout for Call Preservation
New Features in Cisco SRST Version 3.1
Cisco Unified IP Phone 7920 Support
Cisco Unified IP Phone 7936 Support
New Features in Cisco SRST Version 3.0
Additional Language Options for IP Phone Display
Consultative Call Transfer and Forward Using H.450.2 and H.450.3 for SCCP Phones
Customized System Message for Cisco Unified IP Phones
Music-on-Hold for Multicast from Flash Files
Enhancement to the show ephone Command
System Log Messages for Phone Registrations
Three-Party G.711 Ad Hoc Conferencing
Support for Cisco VG248 Analog Phone Gateway 1.2(1) and Higher Versions
New Features in Cisco SRST Version 2.1
Additional Language Options for IP Phone Display
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
New Features in Cisco SRST Version 2.02
Cisco Unified IP Phone Conference Station 7935 Support
Cisco Unity Voice Mail Integration Using In-Band DTMF Signaling Across the PSTN and BRI/PRI
Cisco Unified Survivable Remote Site Telephony Feature Roadmap
Revised: March 15, 2013
This chapter contains a list of Cisco Unified Survivable Remote Site Telephony (Cisco Unified SRST) features and the location of feature documentation.
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.
Contents
•
Information About New Features in Cisco Unified SRST
Documentation Organization
This document consists of the following chapters or appendixes as shown in Table 1.
Table 1 Cisco Unified SRST Configuration Sequence
Gives a brief description of Cisco Unified SRST and provides information on the supported platforms and Cisco Unified IP Phones. In addition, it describes any prerequisites or restrictions that should be addressed before Cisco Unified SIP SRST is configured.
Describes how to set up a Cisco Unified SRST system to communicate with your network.
Describes the features for Cisco Unified SIP SRST Version 4.1 and provides the associated configuration procedures.
Describes how to set up the basic Cisco Unified SRST phone configuration.
Describes features available in Version 3.0 that are also necessary for Version 3.4. Features include instructions on how to provide a backup to an external SIP proxy server by providing basic registrar 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.
Describes how to configure incoming and outgoing calls.
Describes the Secure SRST security functionality to the Cisco Unified SRST.
Integrating Voice Mail with Cisco Unified SRST
Describes how to set up voice mail.
Describes how to set up video parameters.
Provides a list of useful show commands for monitoring and maintaining Cisco Unified SRST.
Appendix A: Configuring Cisco Unified SIP SRST Features Using Redirect Mode
Describes features using redirect mode, which applies to version 3.0 only.
Describes how to configure Cisco Unified CM and Cisco Unified SRST to enable multicast music-on-hold (MOH).
Feature Roadmap
Table 2 provides a feature history summary of Cisco Unified SRST features.
Table 2 Features by Cisco Unified SRST Software Version
Version 9.5
15.3(2)T
•
•
•
•
Version 9.1
15.2(4)M
•
•
Version 9.0
15.2(2)T
•
•
•
Version 8.8
15.2(1)T
Support for Cisco Unified 6945, 8941, and 8945 SCCP IP Phones
Version 8.6
15.1(4)M
Support for Cisco Unified 8941 and 8945 SCCP IP Phones were introduced. For more information, see Configuring Cisco Unified 8941 and 8945 SCCP IP Phones.
Version 8.0
15.1(1)T
Beginning with Cisco IP Phone firmware 8.5(3) and Cisco IOS Release 15.1(1)T, Cisco 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. For more information, see the following sections:
•
•
•
Version 7.0/4.3
See Cisco Feature Navigator for compatibility.
•
Version 4.2(1)
See Cisco Feature Navigator for compatibility.
The following new features are included:
•
•
•
•
Version 4.1
12.4(15)T
•
•
Version 4.0
12.4(4)XC
•
•
•
Version 3.4
12.4(4)T
•
•
Version 3.3
•
Version 3.2
12.3(11)T
•
•
•
•
•
Version 3.1
12.3(7)T
•
Version 3.0
12.2(15)ZJ 12.3(4)T
•
•
•
•
•
•
•
•
Version 2.1
•
•
•
•
•
•
Version 2.02
•
•
•
•
•
Version 2.01
•
•
•
Version 2.0
•
•
•
•
•
•
•
•
•
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Version 1.0
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
Information About New Features in Cisco Unified SRST
This section contains the following topics:
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•
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•
•
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New Features in Cisco Unified SRST Version 9.5
Cisco Unified SRST 9.5 supports the following new features:
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•
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Afterhours Pattern Blocking Support for Regular Expressions
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:
•
•
Note
If calls to the following numbers are to be blocked after hours:
•
•
•
then the following configurations can be used:
•
•
•
•
Note
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.
Call Park Recall Enhancement
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.
Examples
The following example configures the Call Prark Recall:
Router# configure terminalRouter(config)# telephony-serviceRouter(config)# srst mode auto-provision allRouter(config-telephony)# call-park system ? recall Configure parameters for recallRouter(config-telephony)# call-park system recall ? force Force recall for busy call parkinitiatorRouter(config-telephony)# call-park system recall forceDisplay Support for Name of Called Voice Hunt Groups
A voice hunt group is associated with a pilot number. But because there is no association with the name of the voice hunt group when calls are forwarded from the voice hunt group to the final number, the forwarding number is sent without the name of the forwarding party. The final number can be in the form of a voice mail, a Basic Automatic Call Distribution (BACD) script, or another extension.
In Cisco Unified SRST 9.5, the display of the name of the called voice-hunt-group pilot is supported by configuring the following command in voice hunt-group or ephone-hunt configuration mode:
[no] name "primary pilot name" [secondary "secondary pilot name"]
The secondary name is optional and when the secondary pilot name is not explicitly configured, the primary pilot name is applicable to both pilot numbers.
For configuration information, see the "Associating a Name with a Called Voice Hunt Group" section
Examples
The following example configures the primary pilot name for both the primary and secondary pilot numbers:
name SALESThe following example configures different names for the primary and secondary pilot numbers:
name SALES secondary SALES-SECONDARY
Note
The following example associates a two-word name for the primary pilot number and a one-word name for the secondary pilot number:
name "CUSTOMER SERVICE" secondary CSThe following example associates a one-word name for the primary pilot number and a two-word name for the secondary pilot number:
name FINANCE secondary "INTERNAL ACCOUNTING"The following example associates two-word names for the primary and secondary pilot numbers:
name "INTERNAL CALLER" secondary "EXTERNAL CALLER"For configuration information, see the "Associating a Name with a Called Voice Hunt Group" section of Cisco Unified Communications Manager Administration Guide.
For configuration examples, see the "Example: Associating a Name with a Called Voice Hunt Group" section of Cisco Unified Communications Manager Administration Guide.
Restrictions
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•
Preventing Local-Call Forwarding to Final Agent in Voice Hunt Groups
Local or internal calls are calls originating from a Cisco Unified SIP or Cisco Unified SCCP IP phone in the same Cisco Unified CME system.
Before Cisco Unified CME 9.5, the no forward local-calls command was configured in ephone-hunt group to prevent a local call from being forwarded to the next agent.
In Cisco Unified CME 9.5, local calls are prevented from being forwarded to the final destination using the no forward local-calls to-final command in parallel or sequential voice hunt-group configuration mode.
When the no forward local-calls to-final command is configured in sequential voice hunt-group configuration mode, local calls to the hunt-group pilot number are sent sequentially only to the list of members of the group using the rotary-hunt technique. In case all the group members of the voice hunt group are busy, the caller hears a busy tone. If any of the group members are available but do not answer, the caller hears a ringback tone and is eventually disconnected after the specified timeout. The call is not forwarded to the final number.
When the no forward local-calls to-final command is configured in parallel voice hunt-group configuration mode, local calls to the hunt-group pilot number are sent simultaneously to the list of members of the group using the blast technique. In case all the group members of the voice hunt group are busy, the caller hears a busy tone. If any of the group members are available but do not answer, the caller hears a ringback tone and is eventually disconnected after the specified timeout.The call is not forwarded to the final number. or configuration examples, see the "Preventing Local-Call Forwarding to Final Agent in Voice Hunt Groups" section of" section of Cisco Unified Communications Manager Administration Guide.
Trunk-to-Trunk Transfer Blocking for Toll Fraud Prevention on Cisco Unified SIP IP Phones
In Cisco Unified Survivable Remote Site Telephony (SRST) 4.0, trunk-to-trunk transfer blocking for toll bypass fraud prevention is supported on Cisco Unified Skinny Client Control Protocol (SCCP) IP phones.
Table 3 lists the transfer-blocking commands and the appropriate configuration modes for Cisco Unified CME and Cisco Unified SRST.
Note
transfer-pattern
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
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:
•
•
Backward Compatibility
To maintain backward compatibility, all call transfers from Cisco Unified SIP IP phones to any number (local or over 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, call transfers over trunk continue to be blocked when no transfer patterns are configured.
Dial Plans
Whatever dial plan is used for external calls, the same numbers should be configured as specific numbers using the transfer-pattern command.
If a dial plan requires "9" to be dialed before an external call is made, then "9" should be a prefix of the transfer-pattern number. For example, if 12345678 is an external number that requires "9" to be dialed before the external call can be made, then the transfer-pattern number should be 912345678.
transfer max-length
The transfer max-length command is used to indicate the maximum length of the number being dialed for a call transfer. When only a specific number of digits are to be allowed during a call transfer, a value between 3 and 16 is configured.When the number dialed exceeds the maximum length configured, then the call transfer is blocked.
For example, if the maximum length is configured as 5, then only call transfers from Cisco Unified SIP IP phones up to a five-digit directory number are allowed. All call transfers to directory numbers with more than five digits are blocked.
transfer-pattern blocked
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 non-local 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.
Table 4 compares the behaviors of Cisco Unified SCCP and SIP IP phones for specific configurations.
conference transfer-pattern
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 any of 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 CME:Example" section of Cisco Unified Communications Manager Administration Guide.
New Features in Cisco Unified SRST Version 9.1
Cisco Unified SRST 9.1 supports the following new features:
•
•
Note
KEM Support for Cisco Unified 8961, 9951, and 9971 SIP IP Phones
Cisco Unified IP Key Expansion Modules (KEMs) are supported on Cisco Unified 8961, 9951, and 9971 SIP IP phones in Cisco Unified SIP SRST 9.1.
You attach KEMs to supported phones to increase line key and feature key appearances, speed dials, or programmable buttons on your phones.
Table 5 lists the number of keys supported on Cisco Unified 8961, 9951, and 9971 SIP IP phones without KEMs.
Table 5 Number of Configurable Keys on Supported Cisco Unified SIP IP Phones Without KEMs
Fixed feature keys
5
5
6
Line keys
5
5
6
Programmable soft keys
5
5
6
With KEMs, the programmable buttons can be configured as phone line buttons, speed-dial buttons, or phone feature buttons.
Table 6 compares the number of feature keys that can be configured on supported Cisco Unified SIP IP phones with and without KEMs.
Table 6 Number of Configurable Feature Keys
Busy-Lamp-Field speed dial
1 to 11
1 to 113
Directory number
1 to 12
1 to 114
Speed dial
1 to 11
1 to 113
Table 7 lists the maximum number of KEMs supported on Cisco Unified 8961, 9951, and 9971 SIP IP phones.
Restrictions
•
•
•
•
•
For more information on how the blf-speed-dial, number, and speed-dial commands, in voice register pool configuration mode, have been modified, see Cisco Unified Communications Manager Express Command Reference.
For information on installing KEMs on Cisco Unified IP Phone, see the "Installing a Key Expansion Module on the Cisco Unified IP Phone" section of Cisco Unified IP Phone 8961, 9951, and 9971 Administration Guide for Cisco Unified Communications Manager 7.1 (3) (SIP).
Enhancement in Speed-Dial Support
In Cisco Unified SRST 9.1, the "," or comma (pause indicator) is ignored to avoid a break in speed-dial support.
Because the pause speed-dial feature (supported in Cisco Unified Communications Manager or Cisco Unified CM) is not supported in Cisco Unified SRST, Cisco Unified CM and phones (Cisco Unified SCCP IP phones and Cisco Unified SIP IP phones) registered in Cisco Unified SRST maintain backward compatibility in Cisco Unified SRST mode. When phones failover to the Cisco Unified SRST router during WAN outages and Cisco Unified CM failure, the phones only send the speed-dial numbers when the pause speed-dial buttons are pressed. The comma pause indicator is ignored and the preconfigured FAC, PIN, and DTMF are not sent.
For information on configuring speed-dial in Cisco Unified Communications Manager, see the "Device setup" chapter of Cisco Unified Communications Manager Administration Guide.
Voice Hunt Group Support
Cisco Unified SIP SRST 9.1 supports voice hunt groups. Voice hunt groups allow a call placed to a single (pilot) number to contact multiple destinations.
There are three different types of voice hunt groups. Each type uses a different strategy to determine the first number that rings for successive calls to the pilot number until a number answers.
•
•
•
While ephone hunt groups only support Cisco Unified SCCP IP phones, a voice hunt group supports Cisco Unified SCCP IP phones, Cisco Unified SIP IP phones, or a mixture of Cisco Unified SCCP IP phones and Cisco Unified SIP IP phones.
With the voice hunt group feature preconfigured in the Cisco Unified SIP SRST router, voice hunt groups continue to be supported after phones fallback from Cisco Unified CM to the Cisco Unified SIP SRST router.
Restrictions
•
•
New Features in Cisco Unified SRST Version 9.0
Cisco Unified SRST 9.0 supports the following new Cisco Unified SIP IP phones:
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•
•
Cisco Unified SRST 9.0 supports the following new features:
•
Support for Cisco Unified 6901 and 6911 SIP IP Phones
Table 8 lists all the features supported on the Cisco Unified 6901 and 6911 SIP IP Phones in Cisco Unified SRST 9.0.
Table 8 Features Supported on the Cisco Unified 6901 and 6911 SIP IP Phones in
Cisco Unified SRST 9.0After Hour
Not Supported
Not Supported
Barge
Not Supported
Not Supported
Busy-Lamp-Field Monitoring
Not Supported
Not Supported
Button Layout
Not Supported
Not Supported
Call Forward All Softkey
Not Supported
Not Supported
Call Park
Not Supported
Not Supported
Call Transfer
Supported
Supported
cBarge
Not Supported
Not Supported
Directory Service
Not Supported
Not Supported
Extension Mobility
Not Supported
Not Supported
Group Pickup
Not Supported
Not Supported
Hold
Supported
Supported
Intercom
Not Supported
Not Supported
KEM
Not Supported
Not Supported
Meet-Me Conference
Not Supported
Not Supported
Mobility
Not Supported
Not Supported
Multicast MoH
Not Supported
Not Supported
Multicast Paging
Not Supported
Not Supported
MyPhoneApp
Not Supported
Not Supported
Pickup
Not Supported
Not Supported
Privacy
Not Supported
Not Supported
Programmable Line Key
Not Supported
Not Supported
Redial
Supported
Supported
Resume
Supported
Supported
Shared Lines
Not Supported
Not Supported
Software Ad-Hoc Conference
Supported1
Supported1
Speakerphone
Supported
Supported
Speed Dial
Not Supported
Not Supported
Video
Not Supported
Not Supported
1 This feature is controlled by the phone.
Support for Cisco Unified 6921, 6941, 6945, and 6961 SIP IP Phones
Table 9 lists all the features supported on the Cisco Unified 6921, 6941 6945, and 6961 SIP IP Phones in Cisco Unified SRST 9.0.
Table 9 Features Supported on the Cisco Unified 6921, 6941, 6945, and 6961 SIP IP Phones in Cisco Unified SRST 9.0
After Hour
Not Supported
Not Supported
Not Supported
Not Supported
Barge
Not Supported
Not Supported
Not Supported
Not Supported
Busy-Lamp-Field Monitoring
Not Supported
Not Supported
Not Supported
Not Supported
Button Layout
Not Supported
Not Supported
Not Supported
Not Supported
Call Forward All Softkey
Supported1
Supported1
Supported1
Supported1
Call Park
Not Supported
Not Supported
Not Supported
Not Supported
Call Transfer
Supported
Supported
Supported
Supported
cBarge
Not Supported
Not Supported
Not Supported
Not Supported
Directory Service
Supported1
Supported1
Supported1
Supported1
Extension Mobility
Not Supported
Not Supported
Not Supported
Not Supported
Group Pickup
Not Supported
Not Supported
Not Supported
Not Supported
Hold
Supported
Supported
Supported
Supported
Intercom
Not Supported
Not Supported
Not Supported
Not Supported
KEM
Not Supported
Not Supported
Not Supported
Not Supported
Meet-Me Conference
Not Supported
Not Supported
Not Supported
Not Supported
Mobility
Not Supported
Not Supported
Not Supported
Not Supported
Multicast MoH
Not Supported
Not Supported
Not Supported
Not Supported
Multicast Paging
Not Supported
Not Supported
Not Supported
Not Supported
MyPhoneApp
Not Supported
Not Supported
Not Supported
Not Supported
Pickup
Not Supported
Not Supported
Not Supported
Not Supported
Privacy
Not Supported
Not Supported
Not Supported
Not Supported
Programmable Line Key
Not Supported
Not Supported
Not Supported
Not Supported
Redial
Supported
Supported
Supported
Supported
Resume
Supported
Supported
Supported
Supported
Shared Lines
Not Supported
Not Supported
Not Supported
Not Supported
Software Ad-Hoc Conference
Supported1
Supported1
Supported1
Supported1
Speakerphone
Supported
Supported
Supported
Supported
Speed Dial
Not Supported
Not Supported
Not Supported
Not Supported
Video
Not Supported
Not Supported
Not Supported
Not Supported
1 This feature is controlled by the phone.
Support for Cisco Unified 8941 and 8945 SIP IP Phones
Table 10 lists all the features supported on the Cisco Unified 8941 and 8945 SIP IP Phones in Cisco Unified SRST 9.0.
Table 10 Features Supported on the Cisco Unified 8941 and 8945 SIP IP Phones in
Cisco Unified SRST 9.0After Hour
Not Supported
Not Supported
Barge
Not Supported
Not Supported
Busy-Lamp-Field Monitoring
Not Supported
Not Supported
Button Layout
Not Supported
Not Supported
Call Forward All Softkey
Not Supported
Not Supported
Call Park
Not Supported
Not Supported
Call Transfer
Supported
Supported
cBarge
Not Supported
Not Supported
Directory Service
Not Supported
Not Supported
Extension Mobility
Not Supported
Not Supported
Group Pickup
Not Supported
Not Supported
Hold
Supported
Supported
Intercom
Not Supported
Not Supported
KEM
Not Supported
Not Supported
Meet-Me Conference
Not Supported
Not Supported
Mobility
Not Supported
Not Supported
Multicast MoH
Not Supported
Not Supported
Multicast Paging
Not Supported
Not Supported
MyPhoneApp
Not Supported
Not Supported
Pickup
Not Supported
Not Supported
Privacy
Not Supported
Not Supported
Programmable Line Key
Not Supported
Not Supported
Redial
Supported
Supported
Resume
Supported
Supported
Shared Lines
Not Supported
Not Supported
Software Ad-Hoc Conference
Supported1
Supported1
Speakerphone
Supported
Supported
Speed Dial
Not Supported
Not Supported
Video
Supported
Supported
1 This feature is controlled by the phone.
Multiple Calls Per Line
Cisco Unified SRST 9.0 provides support for the Multiple Calls Per Line (MCPL) feature on Cisco Unified 6921, 6941, 6945, and 6961 SIP IP phones and Cisco Unified 8941 and 8945 SCCP and SIP IP phones.
Before Cisco Unified SRST 9.0, the maximum number of calls supported for every directory number (DN) on Cisco Unified 8941 and 8945 SCCP IP phones was restricted to two.
With Cisco Unified SRST 9.0, the MCPL feature overcomes the limitation on the maximum number of calls per line.
In Cisco Unified SRST 9.0, the MCPL feature is not supported on Cisco Unified 6921, 6941, 6945, and 6961 SCCP IP phones. The maximum number of calls allowed on these phones is two and the maximum number of calls allowed on octo-line directory numbers on these phones before activating Call Forward Busy or a busy tone is one.
Cisco Unified 8941 and 8945 SCCP IP Phones
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 and timeouts 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 the "Configuring the Maximum Number of Calls" section.
Cisco Unified 6921, 6941, 6945, 6961, 8941, and 8945 SIP IP Phones
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
•
•
–
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Voice and Fax Support on Cisco ATA-187
Cisco ATA-187 is a SIP-based analog telephone adaptor that turns traditional telephone 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 Unfiied 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 more information on Cisco ATA-187, see Cisco ATA 187 Analog Telephone Adaptor Administration Guide for SIP.
New Features in Cisco Unified SRST Version 8.8
Cisco Unified SRST 8.8 supports the following new Cisco Unified SCCP IP phones:
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Support for Cisco Unified 6945, 8941, and 8945 SCCP IP Phones
Table 12 lists the features supported on Cisco Unified 6945, 8941, and 8945 SCCP IP Phones in Cisco Unified SRST.
Table 12 Features Supported on the Cisco Unified 6945, 8941, and 8945 SCCP IP Phones in Cisco Unified SRST
After Hours
Supported
Supported
Supported
Button Layout
Not Supported
Not Supported
Not Supported
Call Forward
Supported
Supported
Supported
Call Park
Not Supported
Not Supported
Not Supported
Call Transfer
Supported
Supported
Supported
Call Transfer Recall
Not Supported
Not Supported
Not Supported
cBarge
Not Supported
Not Supported
Not Supported
Conferencing1
Supported
Supported
Supported
Directory Services
Supported
Supported
Supported
Enhanced Busy-Lamp-Field Monitoring
Not Supported
Not Supported
Not Supported
Extension Mobility
Not Supported
Not Supported
Not Supported
Forced Authorization Code
Not Supported
Not Supported
Not Supported
Hold
Supported
Supported
Supported
Intercom
Not Supported
Not Supported
Not Supported
Live Record
Not Supported
Not Supported
Not Supported
Multicast MOH
Supported
Supported
Supported
Multicast Paging
Not Supported
Not Supported
Not Supported
My Phone Apps
Not Supported
Not Supported
Not Supported
Night Service
Not Supported
Not Supported
Not Supported
Privacy
Not Supported
Not Supported
Not Supported
Programmable Line Keys
Not Supported
Not Supported
Not Supported
Resume
Supported
Supported
Supported
Secure Real-time Transport Protocol
Supported
Supported
Supported
Shared Lines
Supported
Supported
Supported
Single Number Reach
Not Supported
Not Supported
Not Supported
Speakerphone
Supported
Supported
Supported
Speed Dial
Supported
Supported
Supported
Transfer to Voicemail
Not Supported
Not Supported
Not Supported
Video Telephony
Not Supported
Not Supported
Not Supported
Whisper Intercom
Not Supported
Not Supported
Not Supported
1 Only software conferencing is supported.
For information on the Cisco Unified 6945 SCCP IP Phone, see Cisco Unified IP Phone 6945 User Guide for Cisco Unified Communications Manager Express Version 8.8 (SCCP).
For information on the Cisco Unified 8941 and 8945 SCCP IP Phones, see Cisco Unified IP Phone 8941 and 8945 User Guide for Cisco Unified Communications Manager Express Version 8.8 (SCCP).
New Features in Cisco Unified SRST Version 8.0
Beginning with Cisco IP Phone firmware 8.5(3) and Cisco IOS Release 15.1(1)T, Cisco 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.
New Features in Cisco Unified SRST Version 7.0/4.3
Cisco Unified SRST 7.0/4.3 supports the following new features:
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New Features in Cisco Unified SRST Version 4.2(1)
Cisco Unified SRST Version 4.2(1) introduces the following new features:
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New Features in Cisco Unified SRST Version 4.1
Cisco Unified SRST Version 4.1 introduces the following new feature:
New Features in Cisco Unified SRST Version 4.0
Cisco Unified SRST Version 4.0 has introduced the following new features:
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Additional Cisco Unified IP Phone Support
The following IP phones are supported with Cisco Unified SRST systems:
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In addition, the Cisco Unified IP Phone 7914 Expansion Module can attach to the Cisco 7941G-GE and Cisco 7961G-GE. The Cisco 7914 Expansion Module adds additional features, such as adding 14 line appearances or speed-dial numbers to your phone. You can attach one or two expansion modules to your IP phone. When you use two expansion modules, you have 28 additional line appearances or speed-dial numbers, or a total of 34 line appearances or speed-dial numbers. For more information, see 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 Support
Cisco IP Communicator is a software-based application that delivers enhanced telephony support on personal computers. This SCCP-based application allows computers to function as IP phones, providing high-quality voice calls on the road, in the office, or from wherever users may have access to the corporate network. Cisco IP Communicator appears on a user's computer monitor as a graphical, display-based IP phone with a color screen, a key pad, feature buttons, and soft keys.
Fax Passthrough using SCCP and ATAs Support
Fax passthrough mode is now supported using Cisco VG 224 voice gateways, Analog Telephone Adaptors (ATA), and SCCP. ATAs ship with SIP firmware, so SCCP firmware must be loaded before this feature can be used.
Note
H.323 VoIP Call Preservation Enhancements for WAN Link Failures for SCCP Phones
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.Video Support
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 the "Setting Video Parameters" section.
New Features in Cisco Unified SRST Version 3.4
Cisco SRST V3.4 introduced the new features described in the following section:
Cisco SIP SRST 3.4
Cisco SIP SRST Version 3.4 describes SRST functionality for Session Initiation Protocol (SIP) networks. Cisco SIP SRST Version 3.4 provides backup to an external SIP proxy server by providing basic registrar and back-to-back user agent (B2BUA) services. These services are used by a SIP IP phone in the event of a WAN connection outage when the SIP phone is unable to communicate with its primary SIP proxy.
Cisco SIP SRST Version 3.4 can support SIP phones with standard RFC 3261 feature support locally and across SIP WAN networks. With Cisco SIP SRST Version 3.4, SIP phones can place calls across SIP networks in the same way as Skinny Client Control Protocol (SCCP) phones. For full information about SIP SRST, Version 3.4, see Cisco SIP SRST Version 3.4 System Administrator Guide.
New Features in Cisco SRST Version 3.3
Cisco SRST V3.3 introduced the new features described in the following sections:
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Secure SRST
Secure Cisco IP phones that are located at remote sites and that are attached to gateway routers can communicate securely with Cisco Unified 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 the "Configuring Secure SRST for SCCP and SIP" section.
Cisco Unified IP Phone 7970G and Cisco Unified 7971G-GE Support
The Cisco Unified IP Phones 7970G and 7971G-GE are full-featured telephones that provide voice communication over an IP network. They function much like a traditional analog telephones, allowing you to place and receive phone calls and to access features such as mute, hold, transfer, speed dial, call forward, and more. In addition, because the phones are connected to your data network, they offer enhanced IP telephony features, including access to network information and services, and customizeable features and services. The phones also support security features that include file authentication, device authentication, signaling encryption, and media encryption.
The Cisco Unified IP Phones 7970G and 7971G-GE also provide a color touchscreen, support for up to eight line or speed-dial numbers, context-sensitive online help for buttons and feature, and a variety of other sophisticated functions. No configurations specific to SRST are necessary.
For more information, see the Cisco Unified IP Phone 7900 Series documentation index.
Note
Enhancement to the show ephone Command
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).
New Features in Cisco SRST Version 3.2
Cisco SRST V3.2 introduced the new features described in the following sections:
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Enhancement to the alias Command
The alias command is enhanced as follows:
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For more information, see the alias command in Cisco Unified SRST and Cisco Unified SIP SRST Command Reference (All Versions).
Enhancement to the cor Command
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).
Enhancement to the pickup Command
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).
Enhancement to the user-locale Command
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).Increased the Number of Cisco Unified IP Phones Supported on the Cisco 3845
The Cisco 3845 now supports 720 phones and up to 960 ephone-dns or virtual voice ports.
MOH Live-Feed Support
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 adapter to provide a battery feed. Music from a live feed is obtained from a fixed source and is continuously fed into the MOH playout buffer instead of being read from a flash file. Live-feed MOH can also be multicast to Cisco IP phones. See the "Appendix B: Integrating Cisco Unified Communications Manager and Cisco Unified SRST to Use Cisco Unified SRST as a Multicast MOH Resource" section for configuration instructions.
No Timeout for Call Preservation
To preserve existing H.323 calls on the branch in the event of an outage, disable the H.225 keepalive timer by entering the no h225 timeout keepalive command. This feature is supported in Cisco IOS Releases 12.3(7)T1 and higher versions. See the "Cisco Unified SCCP and SIP SRST Feature Overview" section for more information.
H.323 is not supported with SIP phones.
RFC 2833 DTMF Relay Support
Cisco Skinny Client Control Protocol (SCCP) phones, such as those used with Cisco SRST systems, provide only out-of-band DTMF digit indications. To enable SCCP phones to send digit information to remote SIP-based IVR and voice-mail applications, Cisco SRST 3.2 and later versions provide conversion from the out-of-band SCCP digit indication to the SIP standard for DTMF relay, which is RFC 2833. You select this method in the SIP VoIP dial peer using the dtmf-relay rtp-nte command. See the "How to Configure DTMF Relay for SIP Applications and Voice Mail" section for configuration instructions.
To use voice mail 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.
Translation Profile Support
Cisco SRST 3.2 and later versions support translation profiles. Translation profiles allow you to group translation rules together and to associate translation rules with the following:
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See the "Enabling Translation Profiles" section 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).New Features in Cisco SRST Version 3.1
Cisco SRST V3.1 introduced the new features described in the following sections:
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Note
Cisco Unified IP Phone 7920 Support
The Cisco Unified Wireless IP Phone 7920 is an easy-to-use IEEE 802.11b wireless IP phone that provides comprehensive voice communications in conjunction with Cisco Unified CM and Cisco Aironet 1200, 1100, 350, and 340 Series of Wi-Fi (IEEE 802.11b) access points. As a key part of the Cisco AVVID Wireless Solution, the Cisco Unified Wireless IP Phone 7920 delivers seamless intelligent services, such as security, mobility, quality of service (QoS), and management, across an end-to-end Cisco network.
No configuration is necessary.
Cisco Unified IP Phone 7936 Support
The Cisco Unified IP Conference Station 7936 is an IP-based, hands-free conference room station that uses VoIP technology. The IP Conference Station replaces a traditional analog conferencing unit by providing business conferencing features—such as call hold, call resume, call transfer, call release, redial, mute, and conference—over an IP network.
No configuration is necessary.
New Features in Cisco SRST Version 3.0
Cisco SRST V3.0 introduced the new features described in the following sections:
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Additional Language Options for IP Phone Display
Displays for the Cisco Unified IP Phone 7940G and Cisco Unified IP Phone 7960G can be configured with additional ISO-3166 codes for German, Danish, Spanish, French, Italian, Japanese, Dutch, Norwegian, Portuguese, Russian, Swedish, United States.
Note
Consultative Call Transfer and Forward Using H.450.2 and H.450.3 for SCCP Phones
Cisco SRST V1.0, Cisco SRST V2.0, and Cisco SRST V2.1 allow blind call transfers and blind call forwarding. Blind calls do not give transferring and forwarding parties the ability to announce or consult with destination parties. These three versions of Cisco SRST use a Cisco SRST proprietary mechanism to perform blind transfers. Cisco SRST V3.0 adds the ability to perform call transfers with consultation using the ITU-T H.450.2 (H.450.2) standard and call forwarding using the ITU-T H.450.3 (H.450.3) standard for H.323 calls.
Cisco SRST V3.0 provides support for IP phones to initiate call transfer and forwarding with H.450.2 and H.450.3 by using the default session application. The built-in H.450.2 and H.450.3 support that is provided by the default session application applies to call transfers and call forwarding initiated by IP phones, regardless of the PSTN interface type.
Note
For more information about the default session application, see the Default Session Application Enhancements document.
For configuration information, see the "Enabling Consultative Call Transfer and Forward Using H.450.2 and H.450.3 with Cisco SRST 3.0" section.
Customized System Message for Cisco Unified IP Phones
The display message that appears on Cisco Unified IP Phone 7905G, Cisco Unified IP Phone 7940G, Cisco Unified IP Phone 7960G, and Cisco Unified IP Phone 7910 units when they are in fallback mode can be customized. The new system message command allows you to edit these display messages on a per-router basis. The custom system message feature supports English only.
For further information, see the "Configuring Customized System Messages for Cisco Unified IP Phones" section.
Dual-Line Mode
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 SRST fallback. The max-dn command affects all IP phones on a Cisco SRST router.
For configuration information, see the "Configuring Dual-Line Phones" section.
E1 R2 Signaling Support
Cisco SRST V3.0 supports E1 R2 signaling. R2 signaling is an international signaling standard that is common to channelized E1 networks; however, there is no single signaling standard for R2. The ITU-T Q.400-Q.490 recommendation defines R2, but a number of countries and geographic regions implement R2 in entirely different ways. Cisco Systems addresses this challenge by supporting many localized implementations of R2 signaling in its Cisco IOS software.
The Cisco Systems E1 R2 signaling default is ITU, which supports the following countries: Denmark, Finland, Germany, Russia (ITU variant), Hong Kong (ITU variant), and South Africa (ITU variant). The expression "ITU variant" means there are multiple R2 signaling types in the specified country, but Cisco supports the ITU variant.
Cisco Systems also supports specific local variants of E1 R2 signaling in the following regions, countries, and corporations:
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European Date Formats
The date format on Cisco IP phone displays can be configured with the following two additional formats:
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For configuration information, see the "Configuring IP Phone Clock, Date, and Time Formats" section.
Huntstop for Dual-Line Mode
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.
Music-on-Hold for Multicast from Flash Files
Cisco SRST can be configured 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.
Ringing Timeout Default
A ringing timeout default can be configured for extensions on which no-answer call forwarding has not been enabled. Expiration of the timeout causes incoming calls to return a disconnect code to the caller. This mechanism provides protection against hung calls for inbound calls received over interfaces such as Foreign Exchange Office (FXO) that do not have forward-disconnect supervision. For more information, see the "Configuring the Ringing Timeout Default" section.
Secondary Dial Tone
A secondary dial tone is available for Cisco Unified IP Phones running Cisco SRST. The secondary dial tone is generated when a user dials a predefined PSTN access prefix. An example would be the different dial tone heard when a designated number is pressed to reach an outside line.
The secondary dial tone is created through the secondary dialtone command. For more information, see the "Configuring a Secondary Dial Tone" section.
Enhancement to the show ephone Command
The show ephone command is enhanced to display the following:
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For more information, see the show ephone command in Cisco Unified SRST and Cisco Unified SIP SRST Command Reference (All Versions).
System Log Messages for Phone Registrations
Diagnostic messages are added to the system log whenever a phone registers or unregisters from Cisco Unified SRST.
Three-Party G.711 Ad Hoc Conferencing
Cisco SRST supports three-party ad hoc conferencing using the G.711 coding technique. For conferencing to be available, an IP phone must have a minimum of two lines connected to one or more buttons.
For more information, see the "Enabling Three-Party G.711 Ad Hoc Conferencing" section.
Support for Cisco VG248 Analog Phone Gateway 1.2(1) and Higher Versions
The Cisco VG248 Analog Phone Gateway is a mixed-environment solution, enabled by Cisco AVVID (Architecture for Voice, Video and Integrated Data), that allows organizations to support their legacy analog devices while taking advantage of the new opportunities afforded through the use of IP telephony. The Cisco VG248 is a high-density gateway for using analog phones, fax machines, modems, voice-mail systems, and speakerphones within an enterprise voice system based on Cisco Unified CM.
During Cisco Unified CM fallback, Cisco 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.New Features in Cisco SRST Version 2.1
Cisco SRST V2.1 introduced the new features described in the following sections:
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Note
Additional Language Options for IP Phone Display
Displays for the Cisco Unified IP Phone 7940G and Cisco Unified IP Phone 7960G can be configured with ISO-3166 codes for the following countries:
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Note
For configuration information, see the "Configuring IP Phone Language Display" section.
Cisco SRST Aggregation
For systems running Cisco Unified CM 3.3(2) and later versions, the restriction of running Cisco SRST on a default gateway was removed. Multiple SRST routers can be used to support additional phones. Note that dial peers and dial plans need to be carefully planned and configured for call transfer and forwarding to work properly.
Cisco ATA 186 and ATA 188 Support
The Cisco ATA analog telephone adaptors are handset-to-Ethernet adaptors that allow regular analog telephones to operate on IP-based telephony networks. Cisco ATAs support two voice ports, each with an independent telephone number. The Cisco ATA 188 also has an RJ-45 10/100BASE-T data port. Cisco SRST supports Cisco ATA 186 and Cisco ATA 188 using Skinny Client Control Protocol (SCCP) for voice calls only.
Cisco Unified IP Phone 7902G Support
The Cisco Unified IP Phone 7902G is an entry-level IP phone that addresses the voice communications needs of a lobby, laboratory, manufacturing floor, hallway, or other area where only basic calling capability is required.
The Cisco Unified IP Phone 7902G is a single-line IP phone with fixed feature keys that provide one-touch access to the redial, transfer, conference, and voice-mail access features. Consistent with other Cisco IP phones, the Cisco Unified IP Phone 7902G supports inline power, which allows the phone to receive power over the LAN. This capability gives the network administrator centralized power control and thus greater network availability.
Cisco Unified IP Phone 7905G Support
The Cisco Unified IP Phone 7905G is a basic IP phone that provides a core set of business features. It provides single-line access and four interactive soft keys that guide a user through call features and functions via the pixel-based liquid crystal display (LCD). The graphic capability of the display presents calling information, intuitive access to features, and language localization in future firmware releases. The Cisco Unified IP Phone 7905G supports inline power, which allows the phone to receive power over the LAN.
No configuration is necessary.
Cisco Unified IP Phone 7912G Support
The Cisco Unified IP Phone 7912G provides core business features and addresses the communication needs of a cubicle worker who conducts low to medium telephone traffic. Four dynamic soft keys provide access to call features and functions. The graphic display shows calling information and allows access to features.
The Cisco Unified IP Phone 7912G supports an integrated Ethernet switch, providing LAN connectivity to a local PC. In addition, the Cisco Unified IP Phone 7912G supports inline power, which allows the phone to receive power over the LAN. This capability gives the network administrator centralized power control and thus greater network availability. The combination of inline power and Ethernet switch support reduces cabling needs to a single wire to the desktop.
Cisco Unified IP Phone Expansion Module 7914 Support
The Cisco Unified IP Phone 7914 Expansion Module attaches to your Cisco Unified IP Phone 7960G, adding 14 line appearances or speed-dial numbers to your phone. You can attach one or two expansion modules to your IP phone. When you use two expansion modules, you have 28 additional line appearances or speed-dial numbers or a total of 34 line appearances or speed-dial numbers.
Enhancement to the dialplan-pattern Command
A new keyword was added to the dialplan-pattern command. The extension-pattern keyword sets an extension number's leading digit pattern when it is different from the E.164 telephone number's leading digits defined in the pattern variable. This enhancement allows manipulation of IP phone abbreviated extension number prefix digits. See the dialplan-pattern command in Cisco Unified SRST and Cisco Unified SIP SRST Command Reference (All Versions).
New Features in Cisco SRST Version 2.02
Cisco SRST Version 2.02 introduced the new features described in the following sections:
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Cisco Unified IP Phone Conference Station 7935 Support
The Cisco IP Conference Station 7935 is an IP-based, full-duplex hands-free conference station for use on desktops and offices and in small-to-medium-sized conference rooms. This device attaches a Cisco Catalyst 10/100 Ethernet switch port with a simple RJ-45 connection and dynamically configures itself to the IP network via the DHCP. Other than connecting the Cisco 7935 to an Ethernet switch port, no further administration is necessary. The Cisco 7935 dynamically registers to
Cisco Unified 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.
Increase in Directory Numbers
Table 13 shows the increases in directory numbers.
Cisco Unity Voice Mail Integration Using In-Band DTMF Signaling Across the PSTN and BRI/PRI
Cisco Unity Voice Mail and other voice-mail systems can be integrated with Cisco SRST. Voice-mail integration introduces six new commands:
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Where to Go Next
Proceed to the "Setting Up the Network" section.
