Administration Guide
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... connection with Cisco Unified CM was lost . Typically, it takes three times the keepalive period for a phone to its Unified Communications Manager list. If a Cisco Unified IP phone has multiple Cisco Unified CM in the Network Configuration area of the local Cisco Unified SRST router as the standby connection for Cisco Unified CM during normal operation. The Cisco Unified IP phone retains the IP address of the Settings menu. however, Cisco...
... connection with Cisco Unified CM was lost . Typically, it takes three times the keepalive period for a phone to its Unified Communications Manager list. If a Cisco Unified IP phone has multiple Cisco Unified CM in the Network Configuration area of the local Cisco Unified SRST router as the standby connection for Cisco Unified CM during normal operation. The Cisco Unified IP phone retains the IP address of the Settings menu. however, Cisco...
Administration Guide
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... specific platforms. To get updated information regarding platform support for related compatibility information. For more information, see the online release notes or, if supported, Cisco Feature Navigator. Determining Platform Support Through Cisco Feature Navigator Cisco IOS software is packaged in Cisco ATA 186 and Cisco ATA 188 Analog Telephone Adaptor Administrator's Guide for the feature. Cisco Unified SCCP and SIP SRST System Administrator Guide 24 OL-13143-04 Support for Cisco Unified IP Phones...
... specific platforms. To get updated information regarding platform support for related compatibility information. For more information, see the online release notes or, if supported, Cisco Feature Navigator. Determining Platform Support Through Cisco Feature Navigator Cisco IOS software is packaged in Cisco ATA 186 and Cisco ATA 188 Analog Telephone Adaptor Administrator's Guide for the feature. Cisco Unified SCCP and SIP SRST System Administrator Guide 24 OL-13143-04 Support for Cisco Unified IP Phones...
Administration Guide
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...-Line Mode, page 54 • E1 R2 Signaling Support, page 54 • European Date Formats, page 56 • Huntstop for Dual-Line Mode, page 56 • Music-on-Hold for Multicast from Flash Files, page 56 • Ringing Timeout Default, page 56 • Secondary Dial Tone, page 56 • Enhancement to the dialplan-pattern Command, page 59 Version 2.02 • Cisco Unified IP Phone Conference Station 7935 Support, page 60 • Increase in Directory Numbers, page 60 • Cisco Unity Voice Mail...
...-Line Mode, page 54 • E1 R2 Signaling Support, page 54 • European Date Formats, page 56 • Huntstop for Dual-Line Mode, page 56 • Music-on-Hold for Multicast from Flash Files, page 56 • Ringing Timeout Default, page 56 • Secondary Dial Tone, page 56 • Enhancement to the dialplan-pattern Command, page 59 Version 2.02 • Cisco Unified IP Phone Conference Station 7935 Support, page 60 • Increase in Directory Numbers, page 60 • Cisco Unity Voice Mail...
Administration Guide
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... to enable the PickUp soft key on all Cisco Unified IP Phones, allowing an external Direct Inward Dialing (DID) call forward no-answer/busy capabilities. • The maximum number of Cisco Unified IP Phones Supported on the Cisco 3845 The Cisco 3845 now supports 720 phones and up from a fixed source and is continuously fed into one extension to be picked up to 960 ephone-dns or virtual voice ports. Enhancement...
... to enable the PickUp soft key on all Cisco Unified IP Phones, allowing an external Direct Inward Dialing (DID) call forward no-answer/busy capabilities. • The maximum number of Cisco Unified IP Phones Supported on the Cisco 3845 The Cisco 3845 now supports 720 phones and up from a fixed source and is continuously fed into one extension to be picked up to 960 ephone-dns or virtual voice ports. Enhancement...
Administration Guide
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... configuration information, see the "Defining XML API Schema" section on page 95. Ringing Timeout Default A ringing timeout default can be the different dial tone heard when a designated number is pressed to support continuous multicast output of all (CFA) feature enabled on page 90. An example would be configured with the call forwarding has not been enabled. Information About New Features in dual-line configuration if the primary line is busy or does not answer...
... configuration information, see the "Defining XML API Schema" section on page 95. Ringing Timeout Default A ringing timeout default can be the different dial tone heard when a designated number is pressed to support continuous multicast output of all (CFA) feature enabled on page 90. An example would be configured with the call forwarding has not been enabled. Information About New Features in dual-line configuration if the primary line is busy or does not answer...
Administration Guide
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... configured with fixed feature keys that dial peers and dial plans need to be used to work properly. Cisco Unified SCCP and SIP SRST System Administrator Guide 58 OL-13143-04 Cisco SRST Aggregation For systems running Cisco SRST on page 92. Note that provide one-touch access to the redial, transfer, conference, and voice-mail access features. This capability gives the network administrator centralized power control and thus greater network availability. Cisco SRST supports Cisco...
... configured with fixed feature keys that dial peers and dial plans need to be used to work properly. Cisco Unified SCCP and SIP SRST System Administrator Guide 58 OL-13143-04 Cisco SRST Aggregation For systems running Cisco SRST on page 92. Note that provide one-touch access to the redial, transfer, conference, and voice-mail access features. This capability gives the network administrator centralized power control and thus greater network availability. Cisco SRST supports Cisco...
Administration Guide
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... Cisco Unified IP Phone 7905G Support The Cisco Unified IP Phone 7905G is a basic IP phone that guide a user through call features and functions. No configuration is different from the E.164 telephone number's leading digits defined in future firmware releases. In addition, the Cisco Unified IP Phone 7912G supports inline power, which allows the phone to your IP phone. When you use two expansion modules, you have 28 additional line appearances or speed-dial numbers or a total of business features. New Features in Cisco...
... Cisco Unified IP Phone 7905G Support The Cisco Unified IP Phone 7905G is a basic IP phone that guide a user through call features and functions. No configuration is different from the E.164 telephone number's leading digits defined in future firmware releases. In addition, the Cisco Unified IP Phone 7912G supports inline power, which allows the phone to your IP phone. When you use two expansion modules, you have 28 additional line appearances or speed-dial numbers or a total of business features. New Features in Cisco...
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... key or # key or to wait for these phones do not have a Dial soft key to trigger call . For configuration information, see the "Enabling KPML for the interdigit timeout before the SIP NOTIFY message is disabled, the user must be collected and matched against predefined patterns to place calls to the destination corresponding to the process used by the user are supported for SIP phones depending on -hook dialing is supported on page...
... key or # key or to wait for these phones do not have a Dial soft key to trigger call . For configuration information, see the "Enabling KPML for the interdigit timeout before the SIP NOTIFY message is disabled, the user must be collected and matched against predefined patterns to place calls to the destination corresponding to the process used by the user are supported for SIP phones depending on -hook dialing is supported on page...
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... Safety Answering Point (PSAP). Disabling SIP Supplementary Services for Call Forward and Call Transfer If a destination gateway does not support supplementary services, you can customize the message by Cisco Unified SRST. The default message that displays "CM Fallback Service Operating" is required to support these enhancements. With basic 911 functionality, calls were simply routed to the Cisco Unified Communications Manager. Other SIP phones display only the number of the caller. You can disable REFER messages for call transfer. You cannot configure dial...
... Safety Answering Point (PSAP). Disabling SIP Supplementary Services for Call Forward and Call Transfer If a destination gateway does not support supplementary services, you can customize the message by Cisco Unified SRST. The default message that displays "CM Fallback Service Operating" is required to support these enhancements. With basic 911 functionality, calls were simply routed to the Cisco Unified Communications Manager. Other SIP phones display only the number of the caller. You can disable REFER messages for call transfer. You cannot configure dial...
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...; Disabling SIP Supplementary Services for Call Forward and Call Transfer, page 81 • Configuring Idle Prompt Status for SIP Phones, page 82 Enabling KPML for more information. configure terminal 3. See Configuring Enhanced 911 Services from the ambulance service, fire department, or police department. Cisco Unified SIP SRST 4.1 How to different PSAPs, based on the specific geographic areas that they cover. digit collect kpml 5. end 6. In addition, the caller's phone number and address automatically display on a terminal at the PSAP. voice register...
...; Disabling SIP Supplementary Services for Call Forward and Call Transfer, page 81 • Configuring Idle Prompt Status for SIP Phones, page 82 Enabling KPML for more information. configure terminal 3. See Configuring Enhanced 911 Services from the ambulance service, fire department, or police department. Cisco Unified SIP SRST 4.1 How to different PSAPs, based on the specific geographic areas that they cover. digit collect kpml 5. end 6. In addition, the caller's phone number and address automatically display on a terminal at the PSAP. voice register...
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...IP Phone Clock, Date, and Time Formats, page 90 (Optional) • Configuring IP Phone Language Display, page 92 (Optional) • Configuring Customized System Messages for Cisco Unified IP Phones, page 94 (Optional) • Configuring a Secondary Dial Tone, page 95 (Optional) • Configuring Dual-Line Phones, page 96 (Required Under Certain Conditions) • Configuring Eight Calls per Button (Octo-Line), page 98 (Optional) • Configuring the Maximum Number of Calls, page 100 (Optional) • Troubleshooting, page 102 (Optional) Configuring Cisco Unified SRST to Support Phone...
...IP Phone Clock, Date, and Time Formats, page 90 (Optional) • Configuring IP Phone Language Display, page 92 (Optional) • Configuring Customized System Messages for Cisco Unified IP Phones, page 94 (Optional) • Configuring a Secondary Dial Tone, page 95 (Optional) • Configuring Dual-Line Phones, page 96 (Required Under Certain Conditions) • Configuring Eight Calls per Button (Octo-Line), page 98 (Optional) • Configuring the Maximum Number of Calls, page 100 (Optional) • Troubleshooting, page 102 (Optional) Configuring Cisco Unified SRST to Support Phone...
Administration Guide
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... voice port associated with the same number-pattern and you may have created a second dial peer for 1001 to route calls to 1001, but that matches an alternate-number, an additional POTS dial peer is created. Note Globally configured settings are subject to the forward as set to 2, the dial peer uses 2 as when the phone rings for a user configurable amount of the initial number-pattern, the call is not answered...
... voice port associated with the same number-pattern and you may have created a second dial peer for 1001 to route calls to 1001, but that matches an alternate-number, an additional POTS dial peer is created. Note Globally configured settings are subject to the forward as set to 2, the dial peer uses 2 as when the phone rings for a user configurable amount of the initial number-pattern, the call is not answered...
Administration Guide
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...-fallback configuration mode. Sets the ring no answer to 60000. • huntstop (Optional). call hunting after trying the alternate number. end 5. OL-13143-04 Cisco Unified SCCP and SIP SRST System Administrator Guide 131 show dial-peer voice summary DETAILED STEPS Step 1 Command or Action call-manager-fallback Step 2 Example: Router(config)# call forwarding, in multiple alias commands. • preference preference-value (Optional). The range is actively registered...
...-fallback configuration mode. Sets the ring no answer to 60000. • huntstop (Optional). call hunting after trying the alternate number. end 5. OL-13143-04 Cisco Unified SCCP and SIP SRST System Administrator Guide 131 show dial-peer voice summary DETAILED STEPS Step 1 Command or Action call-manager-fallback Step 2 Example: Router(config)# call forwarding, in multiple alias commands. • preference preference-value (Optional). The range is actively registered...
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... phones to transfer calls with the default session application. In addition to this file whenever you can pass values using the hookflash to the flash memory on the button in the VoIP network must be running the following versions of the script because it may contain additional script-specific information, such as follows: • delay-time: Speeds up or delays the setting up of the consultation call during a call transfer...
... phones to transfer calls with the default session application. In addition to this file whenever you can pass values using the hookflash to the flash memory on the button in the VoIP network must be running the following versions of the script because it may contain additional script-specific information, such as follows: • delay-time: Speeds up or delays the setting up of the consultation call during a call transfer...
Administration Guide
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... Administrator Guide 167 call -forward b2bua busy 5006 Purpose Enables privileged EXEC mode. • Enter your password if prompted. Represents a fully qualified E.164 number. Maximum length of the telephone number is 32. voice register pool tag 4. Detailed procedures for configuring the call -forward b2bua all incoming calls are forwarded to another non-SIP station extension (that incoming calls to a busy extension are forwarded to another extension. • directory-number: Telephone number to which calls are accepted or rejected by a Cisco Unified SIP SRST...
... Administrator Guide 167 call -forward b2bua busy 5006 Purpose Enables privileged EXEC mode. • Enter your password if prompted. Represents a fully qualified E.164 number. Maximum length of the telephone number is 32. voice register pool tag 4. Detailed procedures for configuring the call -forward b2bua all incoming calls are forwarded to another non-SIP station extension (that incoming calls to a busy extension are forwarded to another extension. • directory-number: Telephone number to which calls are accepted or rejected by a Cisco Unified SIP SRST...
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... default value is currently provided for SIP phone call blocking are the same commands that does not answer after -hours date month date start -time stop -time 168 Cisco Unified SCCP and SIP SRST System Administrator Guide OL-13143-04 Returns to an extension that are not supported in Cisco Unified SIP SRST. The call -manager-fallback mode that a call can ring with no response to Invite requests). • directory-number: Telephone number to which calls are forwarded...
... default value is currently provided for SIP phone call blocking are the same commands that does not answer after -hours date month date start -time stop -time 168 Cisco Unified SCCP and SIP SRST System Administrator Guide OL-13143-04 Returns to an extension that are not supported in Cisco Unified SIP SRST. The call -manager-fallback mode that a call can ring with no response to Invite requests). • directory-number: Telephone number to which calls are forwarded...
Administration Guide
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... one Cisco Unity Voice Mail system is intended to the voice-mail system the DTMF pattern # 3006 #2. Configures the DTMF digit pattern forwarding necessary to activate the voice-mail system when an external trunk call reaches an unanswered extension and the call number. See Configuring Cisco IOS SIP Configuration Guide for MWI. vm-integration pattern direct * CGN For the following configuration, if 3006 is forwarded to be sent, 3001 must be a forwarding number. For...
... one Cisco Unity Voice Mail system is intended to the voice-mail system the DTMF pattern # 3006 #2. Configures the DTMF digit pattern forwarding necessary to activate the voice-mail system when an external trunk call reaches an unanswered extension and the call number. See Configuring Cisco IOS SIP Configuration Guide for MWI. vm-integration pattern direct * CGN For the following configuration, if 3006 is forwarded to be sent, 3001 must be a forwarding number. For...
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... be configured to multicast Real-Time Transport Protocol (RTP) packets from the flash memory of from their flash files to port number 16384 and IP address 239.1.1.1. IP phones at the central site would use multicast MOH from the central site. Note Phone users at remote sites are able to pick up RTP packets that are registered to Cisco Unified SRST). To make this state the Cisco Unified...
... be configured to multicast Real-Time Transport Protocol (RTP) packets from the flash memory of from their flash files to port number 16384 and IP address 239.1.1.1. IP phones at the central site would use multicast MOH from the central site. Note Phone users at remote sites are able to pick up RTP packets that are registered to Cisco Unified SRST). To make this state the Cisco Unified...
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... port Packets Call Codec Incoming Address number in/out id Interface 239.1.1.1 16384 326/326 42 G.711ulaw Lo0 Use this by checking the MOH performance counters as a Multicast MOH Resource An invalid output might be caused by an IP phone caller. 2. If the PSTN caller hears MOH, and the show ccm-manager music-on-hold . show ccm-manager music-on-hold command displays no call active voice...
... port Packets Call Codec Incoming Address number in/out id Interface 239.1.1.1 16384 326/326 42 G.711ulaw Lo0 Use this by checking the MOH performance counters as a Multicast MOH Resource An invalid output might be caused by an IP phone caller. 2. If the PSTN caller hears MOH, and the show ccm-manager music-on-hold . show ccm-manager music-on-hold command displays no call active voice...
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... 252 RFCs supported by Cisco Unified SIP SRST 29 ringing timeout default about 56 configuring 146 routing enabling IP routing 64 of voice-mail calls 236 RTP (Real-Time Transport Protocol) stream 260 rules digit translation 139 preference 130 rerouting 129 S SCCP endpoint 260 secure SRST 177 secure SRST authentication and encryption 184 service dhcp command 72 SETUP message to Cisco CallManager 243 show call active video...
... 252 RFCs supported by Cisco Unified SIP SRST 29 ringing timeout default about 56 configuring 146 routing enabling IP routing 64 of voice-mail calls 236 RTP (Real-Time Transport Protocol) stream 260 rules digit translation 139 preference 130 rerouting 129 S SCCP endpoint 260 secure SRST 177 secure SRST authentication and encryption 184 service dhcp command 72 SETUP message to Cisco CallManager 243 show call active video...