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Cisco Unified SCCP and SIP SRST System Administrator Guide (All Versions) April 23, 2012 Americas Headquarters Cisco Systems, Inc. 170 West Tasman Drive San Jose, CA 95134-1706 http://www.cisco.com Tel: 408 526-4000 800 553-NETS (6387) Fax: 408 527-0883...
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OR ITS SUPPLIERS HAVE BEEN ADVISED OF THE POSSIBILITY OF SUCH DAMAGES. Cisco and the Cisco logo are trademarks or registered trademarks of Cisco and/or its affiliates in the U.S. and other countries. To view a list of Cisco trademarks, go to this URL: www.cisco.com/go/trademarks.
Restrictions for Configuring Cisco Unified SIP SRST MGCP Gateways and SRST Support for Cisco Unified IP Phones and Platforms Finding Cisco IOS Software Releases That Support Cisco Unified SRST Cisco Unified IP Phone Support Platform and Memory Support Cisco Unified Communications Manager Compatibility...
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Disabling SIP Supplementary Services for Call Forward and Call Transfer Idle Prompt Status Enhanced 911 Services How to Configure Cisco Unified SIP SRST 4.1 Features Enabling KPML for SIP Phones Disabling SIP Supplementary Services for Call Forward and Call Transfer...
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Configuring Dual-Line Phones Configuring Eight Calls per Button (Octo-Line) Configuring the Maximum Number of Calls Troubleshooting How to Set Up Cisco IP Communicator for Cisco Unified SRST Verifying Cisco IP Communicator Troubleshooting Cisco IP Communicator Where to Go Next Setting Up Cisco Unified IP Phones using SIP...
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H.323 VoIP Call Preservation Enhancements for WAN Link Failures Toll Fraud Prevention Information About Configuring SIP SRST Features Using Back-to-Back User Agent Mode Cisco Unified SIP SRST and Cisco SIP Communications Manager Express Feature Crossover How to Configure Cisco Unified SCCP SRST Configuring Incoming Calls...
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Contents Information About Integrating Voice Mail with Cisco Unified SCCP SRST How to Integrate Voice Mail with Cisco Unified SCCP and SIP SRST Configuring Direct Access to Voice Mail Configuring Message Buttons Redirecting to Cisco Unified Communications Manager Gateway...
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Information About Cisco Unified SIP SRST Features Using Redirect Mode How to Configure Cisco Unified SIP SRST Features Using Redirect Mode Configuring Call Redirect Enhancements to Support Calls Between SIP IP Phones for Cisco Unified SIP SRST Configuring Sending 300 Multiple Choice Support...
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Contents Feature Information for Cisco Unified SRST as a Multicast MOH Resource Where to Go Next Index Cisco Unified SCCP and SIP SRST System Administrator Guide OL-13143-04...
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Contents Cisco Unified SCCP and SIP SRST System Administrator Guide OL-13143-04...
For the most up-to-date information about Cisco Unified IP Phone support, the maximum number of Cisco Unified IP Phones, the maximum number of directory numbers (DNs) or virtual voice ports, and memory requirements for Cisco Unified SRST and Cisco Unified SIP SRST, see Cisco Unified SRST 4.0...
Communications Manager list. Otherwise, the phone activates a standby connection to its secondary Cisco Unified CM. The time it takes for a Cisco Unified IP Phone to fallback to the SRST router can vary depending Note on the phone type. Phones such as the Cisco 7902, Cisco 7905, and Cisco 7912 can take approximately 2.5 minutes to fallback to SRST mode.
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Unified IP phones can bounce between Cisco Unified CM and Cisco Unified SRST. A Cisco Unified IP phone cannot re-establish a connection with the primary Cisco Unified CM at the central office if it is currently engaged in an active call....
Cisco IP phones On H.323 gateways for SCCP SRST, when the WAN link fails, active calls from Cisco Unified IP phones to the PSTN are not maintained by default. Call preservation may work with the no h225 timeout keepalive command.
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To update Cisco Unified SRST, follow the installation instructions described in this section. Installing Cisco Unified SRST V3.0 and Later Versions Install the Cisco IOS software release image containing the Cisco SRST or Cisco Unified SRST version that is compatible with your Cisco Unified Communications Manager version. See the “Cisco Unified...
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If You Have Cisco Communications Manager V3.3 or Later Versions If you have Cisco Communications Manager V3.3 or later versions, you must create an SRST reference and apply it to a device pool. An SRST reference is the IP address of the Cisco Unified SRST Router. Step 1 Create an SRST reference.
Cisco Unified SCCP SRST Restrictions for Configuring Cisco Unified SCCP SRST Table 1 provides a history of restrictions from Cisco SCCP SRST Version 1.0 to the present version of Cisco Unified SCCP SRST. Table 1 Restrictions from Cisco SCCP SRST from the Present Version to Version 1.0 Cisco Unified SRST...
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Cisco Unified SRST feature, this feature is not recommended as a solution for enterprise branch offices. Version 1.0 12.2(2)XB Does not support first generation Cisco Unified IP phones, such as Cisco IP Phone • 30 VIP and Cisco IP Phone 12 SP+. 12.2(2)XG •...
Cisco Unified SIP SRST functionality on a SIP PSTN gateway provides service reliability for SIP-based IP phones in the event of a WAN outage. Cisco Unified SIP SRST enables the SIP IP phones to continue to make and receive calls to and from the PSTN and also to make and receive calls to and from other SIP IP phones.
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TLS connection with NULL-SHA cipher for signaling. If such an Authenticated SIP phone fails over to the Cisco Unified SRST device, and if the Cisco Unified CM and SRST device are configured to support secure SIP SRST, it will register using TCP instead of TLS/TCP, thus disabling the Authenticated mode until the phone fails back to the Cisco Unified CM.
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• emergency call history table when remote IP Phones are in Cisco Unified SRST fallback mode. Therefore, if the PSAP calls back after the Cisco Unified IP Phones register back to Cisco Unified Communications Manager, Cisco Emergency Responder will not have any history of those calls. As a result, those calls will not get routed to the original 911 caller.
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Transcoding is not supported. • Phone Features For call waiting to work on the Cisco ATA and Cisco IP Phone 7912 and Cisco • Unified IP Phone 7905G with a 1.0(2) build, the incoming call leg should be configured with the G.711 codec.
MGCP gateway. These two commands allow SRST to assume control over the voice port and over call processing on the MGCP gateway. With Cisco IOS earlier than 12.3(14)T, the two commands are the ccm-manager fallback-mgcp and call application alternate commands. With Cisco IOS releases after 12.3(14)T, the ccm-manager fallback-mgcp and service commands must be...
Compatibility Information. 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 passthrough 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.
• Where to Go Next The next chapters of this book describe how to configure Cisco Unified SIP SRST. As shown in Table each chapter takes you through tasks in the order in which they need to be performed. The first task for configuring Cisco Unified SRST is to ensure that the basic software and hardware in your system are configured correctly for Cisco Unified SRST.
For information on obtaining documentation, obtaining support, providing documentation feedback, security guidelines, and also recommended aliases and general Cisco documents, see the monthly What’s New in Cisco Product Documentation, which also lists all new and revised Cisco technical documentation, at http://www.cisco.com/en/US/docs/general/whatsnew/whatsnew.html.
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Cisco Unified SCCP and SIP SRST Feature Overview Obtaining Documentation, Obtaining Support, and Security Guidelines Cisco Unified SCCP and SIP SRST System Administrator Guide OL-13143-04...
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.
Cisco Unified SIP SRST is configured. Setting Up the Network, page 63 Describes how to set up a Cisco Unified SRST system to communicate with your network. Cisco Unified SIP SRST 4.1, Describes the features for Cisco Unified SIP SRST Version 4.1 and provides the...
Voice and Fax Support on Cisco ATA-187, page 43 • Version 8.8 15.2(1)T Support for Cisco Unified 6945, 8941, and 8945 SCCP IP Phones, page 44 Version 8.6 15.1(4)M Support for Cisco Unified 8941 and 8945 SCCP IP Phones were introduced. For...
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• Enhancement to the pickup Command, page 50 • Enhancement to the user-locale Command, page 50 • Increased the Number of Cisco Unified IP Phones Supported on the Cisco • 3845, page 50 MOH Live-Feed Support, page 50 • No Timeout for Call Preservation, page 51 •...
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Cisco 7200 routers (NPE-225, NPE-300, and NPE400). Support was removed for the Cisco MC3810-V3 concentrator. • Version 2.01 Cisco Unified SRST was implemented on the Cisco 1760 routers, and support • for the Cisco 1750 was removed. Support was added for additional connected Cisco IP phones.
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Features by Cisco Unified SRST Software Version (continued) Cisco Unified SRST Cisco IOS Release Enhancements or Modifications Version 2.0 Cisco Unified SRST was implemented on the Cisco 2600XM and Cisco 2691 • routers. Cisco Unified SRST was integrated into Cisco IOS Release 12.2(8)T and •...
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• Cisco 3600 series multiservice routers and the Cisco IAD2420 series integrated access devices. Cisco IP phones able to establish a connection with an SRST router in the • event of a WAN link to Cisco Unified Communications Manager failure.
Voice and Fax Support on Cisco ATA-187, page 43 Support for Cisco Unified 6901 and 6911 SIP IP Phones Table 3 lists all the features supported on the Cisco Unified 6901 and 6911 SIP IP Phones in Cisco Unified SRST 9.0. ...
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Cisco Unified Survivable Remote Site Telephony Feature Roadmap Information About New Features in Cisco Unified SRST Table 3 Features Supported on the Cisco Unified 6901 and 6911 SIP IP Phones in Cisco Unified SRST 9.0 (continued) Features 6901 6911...
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Information About New Features in Cisco Unified SRST Support for Cisco Unified 6921, 6941, 6945, and 6961 SIP IP Phones Table 4 lists all the features supported on the Cisco Unified 6921, 6941 6945, and 6961 SIP IP Phones in Cisco Unified SRST 9.0. Table 4 Features Supported on the Cisco Unified 6921, 6941, 6945, and 6961 SIP IP Phones in Cisco Unified SRST 9.0...
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1. This feature is controlled by the phone. Support for Cisco Unified 8941 and 8945 SIP IP Phones Table 5 lists all the features supported on the Cisco Unified 8941 and 8945 SIP IP Phones in Cisco Unified SRST 9.0. ...
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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.
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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.
• Support for Cisco Unified 6945, 8941, and 8945 SCCP IP Phones Table 7 lists the features supported on Cisco Unified 6945, 8941, and 8945 SCCP IP Phones in Cisco Unified SRST. Table 7 Features Supported on the Cisco Unified 6945, 8941, and 8945 SCCP IP Phones in...
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(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.
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.
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Adaptors (ATA), and SCCP. ATAs ship with SIP firmware, so SCCP firmware must be loaded before this feature can be used. For ATAs that are registered to a Cisco Unified SRST system to participate in FAX calls, they must have Note their ConnectMode parameter set to use the “standard payload type 0/8”...
• 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.
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.
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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.
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“How to Configure DTMF Relay for SIP Applications and Voice Mail” section on page 252 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.
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...
Support for Cisco VG248 Analog Phone Gateway 1.2(1) and Higher Versions, page 57 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.
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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.
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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.
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Cisco Unified Survivable Remote Site Telephony Feature Roadmap Information About New Features in Cisco Unified SRST European Date Formats The date format on Cisco IP phone displays can be configured with the following two additional formats: yy-mm-dd (year-month-day) • yy-dd-mm (year-day-month) •...
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.
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Information About New Features in Cisco Unified SRST 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: France •...
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).
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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 Survivable Remote Site Telephony Feature Roadmap Where to Go Next Where to Go Next Proceed to the “Setting Up the Network” section on page Cisco Unified SCCP and SIP SRST System Administrator Guide OL-13143-04...
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Cisco Unified Survivable Remote Site Telephony Feature Roadmap Where to Go Next Cisco Unified SCCP and SIP SRST System Administrator Guide OL-13143-04...
Setting Up the Network Revised: February 3, 2011 This chapter describes how to configure your Cisco Unified Survivable Remote Site Telephony (SRST) router to run DHCP and to communicate with the IP phones during Cisco Unified Communications Manager fallback. Contents •...
Information About Setting Up the Network When the WAN link fails, the Cisco Unified IP Phones detect that they are no longer receiving keepalive packets from Cisco Unified CM. The Cisco Unified IP Phones then register with the router. The Cisco Unified SRST software is automatically activated and builds a local database of all Cisco Unified IP Phones attached to it (up to its configured maximum).
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MGCP-controlled PRI fails when normal MGCP operation is restored. Configuring Cisco Unified SRST on an MGCP Gateway Prior to Cisco IOS Release 12.3(14)T Perform this task to enable SRST on a MGCP Gateway if you are using a software release prior to Cisco IOS Release 12.3(14)T.
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Example: Router(config)# exit Configuring SRST on an MGCP Gateway Using Cisco IOS Release 12.3(14)T or Later Releases Perform this task to enable SRST on an MGCP Gateway if you are using Cisco IOS Release 12.3(14)T or later version. Restrictions Effective with Cisco IOS Release 12.3(14)T, the call application alternate command is replaced by the service command.
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If a specific application Example: name is not entered, the gateway uses the DEFAULT Router(config) application app-xfer application. Step 5 Enters global configuration mode. global Example: Router(config)# global Cisco Unified SCCP and SIP SRST System Administrator Guide OL-13143-04...
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Router(config)# exit Configuration Example of Enabling SRST on a MGCP Gateway using Cisco IOS Release 12.3(14)T The following is an example of configuring SRST on an MGCP Gateway if you are using Cisco IOS Release 12.3(14)T or later release: isdn switch-type primary-net5...
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How to Set Up the Network ! call rsvp-sync ! call application alternate DEFAULT !--- For Cisco IOS® Software Release 12.3(14)T or later, this command was replaced by the service command in global application configuration mode. application global service alternate Default...
Cisco Unified Communications Manager documentation. When a Cisco IP phone is connected to the Cisco Unified SRST system, it automatically queries for a DHCP server. The DHCP server responds by assigning an IP address to the Cisco IP phone and providing the IP address of the TFTP server through DHCP option 150.
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Example: Router(config-dhcp)# exit Defining a Separate DHCP IP Address Pool for Each Cisco Unified IP Phone This task creates a name for the DHCP server address pool and specifies IP addresses. This method requires you to make an entry for every Cisco Unified IP phone.
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For further details about DHCP configuration, see the Cisco IOS DHCP Server document. The Cisco IOS DHCP server feature is enabled on routers by default. If the DHCP server is not enabled on your Cisco Unified SRST router, use the following steps to enable it. SUMMARY STEPS...
If you plan to use the default time interval between messages, which is 30 seconds, you do not have to Note perform this task. SUMMARY STEPS call-manager-fallback keepalive seconds exit Cisco Unified SCCP and SIP SRST System Administrator Guide OL-13143-04...
Example: Router(config)# call-manager-fallback Step 2 Sets the time interval, in seconds, between keepalive keepalive seconds messages that are sent to the router by Cisco Unified IP Phones. Example: seconds: Range is 10 to 65535. Default is 30. • Router(config-cm-fallback)# keepalive 60 Step 3 Exits call-manager-fallback configuration mode.
• Idle Prompt Status • With Cisco IOS Release 12.4(15)T, the number of SIP phones supported on each platform is now Note equivalent to the number of SCCP phones supported. For example, 3845 now supports 720 phones regardless of whether these are SIP or SCCP.
Out-of-dialog REFER (OOD-R) enables remote applications to establish calls by sending a REFER message to Cisco Unified SRST without an initial INVITE. After the REFER is sent, the remainder of the call setup is independent of the application and the media stream does not flow through the application.
SIP phone sends an INVITE message to Cisco Unified SRST to initiate the call to the number matching the user's input. All of the digits entered by the user are presented as a block to Cisco Unified SRST for processing. Because digit collection is done by the phone, dial plans reduce signaling messages overhead compared to KPML digit collection.
SCCP and SIP endpoints. Idle Prompt Status A message displays on the status line of a SIP phone after the phone registers to Cisco Unified SRST to indicate that Cisco Unified SRST is providing fallback support for the Cisco Unified Communications Manager.
Enabling KPML for SIP Phones Perform the following steps to enable KPML digit collection on a SIP phone. Restrictions This feature is supported only on Cisco Unified IP Phone 7911G, 7941G, 7941GE, 7961G, 7961GE, • 7970G, and 7971GE. A dial plan assigned to a phone has priority over KPML.
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Router# show voice register dial-peers What to Do Next After changing the KPML configuration in Cisco Unified SRST, you do not need to create new configuration profiles and restart the phones. Enabling or disabling KPML is effective immediately in Cisco Unified SRST.
Perform the following steps to disable REFER messages for call transfers and redirect responses for call forwarding from being sent to the destination by Cisco Unified SRST. You can disable these supplementary features if the destination gateway does not support them.
You do not need to create new configuration files with the create profile command and restart the phones Note after changing the idle status message in Cisco Unified SRST. Modifying the status message takes effect immediately in Cisco Unified SRST.
Example: Router# show voice register global Where to Go Next The next step is configuring Cisco Unified IP phones using SCCP. For instructions, see the “Setting Up Cisco Unified IP Phones using SCCP” section on page For additional information, see the “Additional References”...
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Cisco Unified SIP SRST 4.1 Where to Go Next Cisco Unified SCCP and SIP SRST System Administrator Guide OL-13143-04...
Setting Up Cisco Unified IP Phones using SCCP Revised: April 23, 2012 This chapter describes how to set up the displays and features that callers will see and use on Cisco Unified IP Phones during Cisco Unified CM fallback. Note Ciso Unified IP Phones discussed in this chapter are just examples.
(Optional) Configuring Cisco Unified SRST to Support Phone Functions When the Cisco Unified SRST is enabled, Cisco Unified IP Phones do not have to be reconfigured while in Cisco Unified Communications Manager fallback mode because phones retain the same configuration that was used with Cisco Unified Communications Manager.
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Example: Router(config)# call-manager-fallback Step 2 Enables the router to receive messages from the Cisco IP ip source-address ip-address [port port] [any-match | strict-match] phones through the specified IP addresses and provides for strict IP address verification. The default port number is 2000.
Router(config-cm-fallback)# exit Configuring Cisco Unified 8941 and 8945 SCCP IP Phones To configure Cisco Unified 8941 and 8945 SCCP IP Phones in SRST mode, perform the following commands: This section is required only in SRST version 8.6 and is not required for version 8.6 and higher.
Enter the show call-manager-fallback all command to verify that the Cisco Unified SRST feature is Step 2 enabled. Use the Settings display on the Cisco IP phones in your network to verify that the default router IP Step 3 address on the phones matches the IP address of the Cisco Unified SRST router.
UTC time, and apply a timezone offset to produce the correct time display. Cisco IP Phone 7960 IP phones and other similar SCCP phones such as the Cisco IP Phone 7940, get their display clock information from the local time of the SRST router during SRST registration. If the Cisco Unified SRST router is configured to use the Network Time Protocol (NTP) to automatically sync the Cisco Unified SRST router time from an NTP time server, only UTC time is delivered to the router.
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• The default is set to mm-dd-yy. Step 4 Sets the time display format on all Cisco Unified IP Phones time-format {12 | 24} registered with the router. The default is set to a 12-hour clock. Example:...
Cisco Unified IP Phones default to the ISO-3166 country code of US (United States). The Cisco Unified IP Phone 7940 and Cisco Unified IP Phone 7960 can be configured for different languages (character sets and spelling conventions) using the user-locale command.
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Enters call-manager-fallback configuration mode. call-manager-fallback Example: Router(config)# call-manager-fallback Step 2 Selects a language by country for displays on the Cisco IP user-locale country-code Phone 7940 and Cisco IP Phone 7960. The following ISO-3166 codes are available to Cisco SRST Example:...
IP phones in fallback mode is “CM Fallback Service Operating.” The secondary keyword is for Cisco Unified IP Phones that do not support static text messages and have a limited display space. Secondary IP phones flash messages during fallback. The default display message for secondary IP phones in fallback mode is “CM Fallback Service.”...
Setting Up Cisco Unified IP Phones using SCCP How to Set Up Cisco Unified IP Phones Examples The following example sets “SRST V3.0” as the system display message for all Cisco Unified IP Phones on a router: call-manager-fallback system message primary SRST V3.0 system message secondary SRST V3.0...
Cisco SRST 3.0” section on page 148. Dual-line IP phones are supported during Cisco Unified CM fallback using the max-dn command. Dual-line IP phones have one voice port with two channels to handle two independent calls. This capability enables call waiting, call transfer, and conference functions on a phone-line button.
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Examples The following example sets the maximum number of DNs or virtual voice ports that can be supported by a router to 10 and activates the dual-line mode for all IP phones in Cisco Unified CM fallback mode: call-manager-fallback max-dn 10 dual-line...
Cisco Unified CM 6.0 • Cisco IOS Release 12.4(15)XZ • Restrictions Octo-line directory numbers are not supported by the Cisco Unified IP Phone 7902, 7920, or 7931, or by analog phones connected to Cisco ATA or Cisco VG224. SUMMARY STEPS enable configure terminal...
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The range is 1 to 8 and the default is 8. The command is supported for octo-line directory • numbers only. Step 6 Returns to privileged EXEC mode. Example: Router(config)# end Cisco Unified SCCP and SIP SRST System Administrator Guide OL-13143-04...
23 octo-line 8 huntstop channel 6 Configuring the Maximum Number of Calls To configure the maximum number of calls on a Cisco Unified SCCP IP phone in Cisco Unified SRST 9.0, perform the following steps. Prerequisites Cisco Unified SRST 9.0 and later versions.
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The default is 0 directory numbers and 1 channel per virtual port. • dual-line—(Optional) Sets all Cisco Unified IP phones connected to a Cisco Unified SRST router to one virtual voice port with two channels. octo-line—(Optional) Sets all Cisco Unified IP phones •...
Headset with microphone for your PC (Optional; you can use PC internal speakers and microphone) • Download the latest version of the Cisco IP Communicator software and install it on your PC. The Step 1 software is available for download at http://www.cisco.com/cisco/web/download/index.html.
Open the Network > User Preferences window. Enter the IP address of the Cisco Unified SRST TFTP server. Ensure that Cisco IP Communicator has at least once registered to Cisco Unified CM. For more details, Step 6 Install and Configure IP Communicator with CallManager.
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Setting Up Cisco Unified IP Phones using SCCP Where to Go Next Cisco Unified SCCP and SIP SRST System Administrator Guide OL-13143-04...
Setting Up Cisco Unified IP Phones using SIP Revised: February 3, 2011 Session Initiation Protocol (SIP) registrar functionality in Cisco IOS software is an essential part of Cisco Unified SIP Survivable Remote Site Telephony (SRST). According to RFC 3261, a SIP registrar is a server that accepts Register requests and is typically collocated with a proxy or redirect server.
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. The Cisco Unified SIP SRST device also provides PSTN gateway access for placing and receiving PSTN calls.
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Router(config)# voice service voip Step 4 Allows connections from SIP to SIP endpoints. allow-connections sip to sip Example: Router(config-voi-srv)# allow-connections sip to sip Step 5 Enters SIP configuration mode. Example: Router(config-voi-srv)# sip Cisco Unified SCCP and SIP SRST System Administrator Guide OL-13143-04...
The commands in the configuration below provide registration permission control and set up a basic voice register pool. The pool gives users control over which registrations are accepted by a Cisco Unified SIP SRST device and which can be rejected. Registrations that match this pool create VoIP SIP dial peers with the dial-peer attributes set to these configurations.
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Proxy dial peers are autogenerated dial peers that route all calls from the PSTN to • Cisco Unified SIP SRST. When a SIP phone registers to Cisco Unified SIP SRST and the proxy command is enabled, two dial peers are automatically created. The first dial peer routes to the proxy, and the second (or fallback) dial peer routes to the SIP phone.
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Use this command to control which registrations are • Example: accepted or rejected by a Cisco Unified SIP SRST Router(config)# voice register pool 12 device. Step 5 Explicitly identifies a locally available individual or set of...
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Autogenerates additional VoIP dial peers to reach the main proxy ip-address [preference value] [monitor probe {icmp-ping | rtr} [alternate-ip-address]] SIP proxy whenever a Cisco Unified SIP IP Phone registers with a Cisco Unified SIP SRST gateway. The keywords and arguments are defined as follows:...
Enters voice register pool configuration mode. voice register pool tag Use this command to control which registrations are • accepted or rejected by a Cisco Unified SIP SRST Example: device. Router(config)# voice register pool 12 Cisco Unified SCCP and SIP SRST System Administrator Guide...
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1 Step 5 Allows Cisco Unified SIP IP Phones to handle inbound alias tag pattern to target [preference value] PSTN calls to telephone numbers that are unavailable when the main proxy is not available. The keywords and...
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Step 8 Indicates the E.164 phone numbers that the registrar permits number tag number-pattern {preference value} [huntstop] to handle the Register message from the Cisco Unified SIP IP Phone. The keywords and arguments are defined as follows: Example: Router(config-register-pool)# number 1 50..
Verifying SIP Registrar Configuration To help you troubleshoot a SIP registrar and voice register pool, perform the following steps. SUMMARY STEPS debug voice register errors debug voice register events show sip-ua status registrar Cisco Unified SCCP and SIP SRST System Administrator Guide OL-13143-04...
To use the icmp-ping keyword with the proxy command to assist in troubleshooting proxy dial peers, perform the following steps. SUMMARY STEPS configure terminal voice register pool tag proxy ip-address [preference value] [monitor probe {icmp-ping | rtr} [alternate-ip-address]] show voice register dial-peers show dial-peer voice Cisco Unified SCCP and SIP SRST System Administrator Guide OL-13143-04...
Last Disconnect Cause is "", Last Disconnect Text is "", Last Setup Time = 0. Where to Go Next The next step is configuring incoming and outgoing calls for Cisco Unified SRST. For more information, see the “Configuring Call Handling” section on page 123.
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Setting Up Cisco Unified IP Phones using SIP Where to Go Next Cisco Unified SCCP and SIP SRST System Administrator Guide OL-13143-04...
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Setting Up Cisco Unified IP Phones using SIP Where to Go Next Cisco Unified SCCP and SIP SRST System Administrator Guide OL-13143-04...
Configuring Call Handling Revised: November 14, 2011 This chapter describes how to configure Cisco Unified Survivable Remote Site Telephony (Cisco Unified SRST) for incoming and outgoing calls for SCCP phones. This chapter also describes support for standardized RFC 3261 features for SIP phones. Features include call blocking and call forwarding.
Information About Configuring SCCP SRST Call Handling Cisco Unified SRST offers a smaller set of call handling capabilities than Cisco Unified CM, and much of the configuration for these feature involves enabling existing Cisco Unified CM or Cisco Unified IP Phone settings.
Cisco Unified SIP SRST also uses a back-to-back user agent (B2BUA), which is a separate call agent that has more features than Cisco SIP SRST 3.0, which used a redirect server that only accepted and forwarded calls. The main advantage of a B2BUA call agent is in call forwarding, because it forwards calls on behalf of the phone.
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Configuring Call Handling Information About Configuring SIP SRST Features Using Back-to-Back User Agent Mode Table 1 Version 3.4 New or Enhanced Commands for Cisco Unified SRST and Cisco Unified CME (Sorted by Configuration Mode) (continued) Configurable for Voice Applicable to ...
Configuring Call Handling How to Configure Cisco Unified SCCP SRST Table 1 Version 3.4 New or Enhanced Commands for Cisco Unified SRST and Cisco Unified CME (Sorted by Configuration Mode) (continued) Configurable for Voice Applicable to Dial Register...
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– Configuring Call Forwarding During a Busy Signal or No Answer Incoming calls that reach a busy signal or go unanswered during Cisco Unified CM fallback can be configured to be forwarded to one or more E.164 numbers. SUMMARY STEPS...
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Up to 50 sets of rerouting alias rules can be created for calls to telephone numbers that are unavailable during Cisco Unified Communications Manager fallback. Sets of alias rules are created using the alias command. An alias is activated when a telephone registers that has a phone number matching a configured alternate-number alias.
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The configured alternate-number must be a specific E.164 phone number or extension that belongs to an IP phone registered on the Cisco Unified SRST router. When an IP phone registers with a number that matches an alternate-number, an additional POTS dial peer is created. The destination pattern is set to the initial configured number-pattern, and the POTS dial peer voice port is set to match the voice port associated with the alternate-number.
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(Optional). Sets the ring • no-answer timeout duration for call forwarding, in seconds. Range is from 3 to 60000. huntstop (Optional). Stops call hunting after trying • the alternate number. Cisco Unified SCCP and SIP SRST System Administrator Guide OL-13143-04...
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The SRST pickup command is designed to operate in a manner compatible with Cisco Unified Communications Manager. Cisco Unified SCCP and SIP SRST System Administrator Guide OL-13143-04...
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The default phone load on Cisco Unified Communications Manager, Release 4.0(1) for the Cisco 7905 Note and Cisco 7912 IP phones does not enable the PickUp soft key during fallback. To enable the PickUp soft key on Cisco 7905 and Cisco 7912 IP phones, upgrade your default phone load to ...
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Configuring Consultative Transfer Before Cisco Unified SRST 4.3, the consultative transfer feature played a dial tone and collected dialed digits until the digits matched the pattern for consultative transfer, blind transfer, or PSTN transfer blocking. The after-hours blocking criteria was applied after the consultative transfer digit collection and pattern matching.
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Cisco IOS software to transfer the call, and then terminates the original call bubble. The PARK FAC code is handled in the same way as by a new call which requires that a ten-second timer is applied by the Cisco IOS software.
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Cisco Unified SRST 4.3 • • Cisco Unified CM 6.0 • Cisco IOS Release 12.4(15)XZ Restrictions for Cisco Unified SRST 4.3 The Cisco 3200 Series Mobile Access Router does not support SRST. • SUMMARY STEPS enable configure terminal call-manager-fallback transfer-digit-collect {new-call | orig-call}...
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You have the option not to register numbers to the gatekeeper so that those numbers can be used for other telephony services. SUMMARY STEPS call-manager-fallback dialplan-pattern tag pattern extension-length length [extension-pattern extension-pattern] [no-reg] exit Cisco Unified SCCP and SIP SRST System Administrator Guide OL-13143-04...
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4085550144 matches dial-plan pattern 1. It will use the extension-length keyword to extract the last three digits of the number 144 and present this as the caller ID for the incoming call. call-manager-fallback dialplan-pattern 1 40855501.. extension-length 3 no-reg Cisco Unified SCCP and SIP SRST System Administrator Guide OL-13143-04...
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Cisco IOS Voice Configuration Library. If you are running Cisco SRST 3.2 and later or Cisco Unified SRST 4.0 and later, use the configuration described in the “Enabling Translation Profiles” section on page 140 instead of using the translate command as described below.
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Enabling Translation Profiles Cisco SRST 3.2 and later and Cisco Unified SRST 4.0 and later support translation profiles. Translation profiles are the suggested way to allow you to group translation rules and provide instructions on how to apply the translation rules to the following: Called numbers •...
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For SRST, you apply your profiles in call-manager fallback mode. Cisco IP phones support one incoming and one outgoing translation profile when in SRST mode. For Cisco SRST 3.2 and later versions and Cisco Unified SRST 4.0 and later versions, use the voice Note...
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The reference number of the • translation rule from 1 to 2147483647. Step 6 Exits translation-profile configuration mode. exit Example: Router(cfg-translation-profile)# exit Step 7 Enters call-manager-fallback configuration mode. call-manager-fallback Example: Router(config)# call-manager-fallback Cisco Unified SCCP and SIP SRST System Administrator Guide OL-13143-04...
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The following example shows the configuration where a translation profile called name1 is created with two voice translation rules. Rule1 consists of associated calling numbers, and rule2 consists of redirected called numbers. The Cisco Unified IP Phones in SRST mode are configured with name1. voice translation-profile name1...
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Dial-peer hunting, the search through a group of dial peers for an available phone line, is disabled during Cisco Unified CM fallback by default. To enable dial-peer hunting, use the no huntstop command. For more information about dial-peer hunting, see Cisco IOS Voice Configuration Library.
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Exits call-manager-fallback configuration mode. exit Example: Router(config-cm-fallback)# exit Examples The following example disables dial-peer hunting during Cisco Unified CM fallback and hunting to the secondary channels in dual-line phone configurations: call-manager-fallback no huntstop channel Configuring Busy Timeout This task sets the timeout value for call transfers to busy destinations. The busy timeout value is the amount of time that can elapse after a transferred call reaches a busy signal before the call is disconnected.
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Foreign Exchange Office (FXO) that do not have forward-disconnect supervision. It is used only for extensions that do not have no-answer call forwarding enabled. SUMMARY STEPS call-manager-fallback timeouts ringing seconds exit Cisco Unified SCCP and SIP SRST System Administrator Guide OL-13143-04...
You must configure Cisco Unified SRST to allow Cisco Unified IP Phones to transfer telephone calls from outside the local IP network to another Cisco Unified IP Phone. By default, all Cisco Unified IP Phone directory numbers or virtual voice ports are allowed as transfer targets. A maximum of 32 transfer patterns can be entered.
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Router(config-cm-fallback)# exit Examples In the following example, the transfer-pattern command permits transfers from a non-IP phone number to any Cisco Unified IP Phone on the same IP network with a number in the range from 5550100 to 5550199: call-manager-fallback transfer-pattern 55501..
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{blind | full-blind | full-consult | local-consult} (call transfer only) transfer-pattern transfer-pattern (call transfer only) exit voice service voip h323 h450 h450-2 timeout {T1 | T2 | T3 | T4} milliseconds Cisco Unified SCCP and SIP SRST System Administrator Guide OL-13143-04...
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H.450.3 standard. A pattern of .T forwards all calling parties using the H.450.3 standard. Step 3 Not supported if the transfer-to destination is on the Cisco transfer-system {blind | full-blind | full-consult | local-consult} ATA, Cisco VG224, or an SCCP-controlled FXS port.
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4000 voip destination-pattern 4… session-target ipv4:10.1.1.1 call-manager-fallback transfer-pattern 4… transfer-system full-consult The following example enables call forwarding using the H.450.3 standard: dial-peer voice 100 pots destination-pattern 9.T port 1/0/0 Cisco Unified SCCP and SIP SRST System Administrator Guide OL-13143-04...
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TFTP server that is available to the Cisco Unified SRST router or copied to the flash memory on the Cisco Unified SRST router. To apply this script globally to all dial peers, use the call application global command in global configuration mode.
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After the initial call transfer operation is completed and the transferee and transfer-to parties are now the only parties in the call, the transfer-to party may further transfer the call. Call transfer with consultation is not supported for Cisco ATA-186, Cisco ATA-188, and Cisco IP •...
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Flash memory (flash) or a directory on a server (TFTP, HTTP, or RTSP) are all valid. Prompts are required for call transfer from analog FXS phones. No prompts are needed for call transfer from IP phones. Cisco Unified SCCP and SIP SRST System Administrator Guide OL-13143-04...
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1 second: call application voice transfer_app flash:app-h450-transfer.tcl call application voice transfer_app language 1 en call application voice transfer_app set-location en 0 flash:/prompts call application voice transfer_app delay-time 1 Cisco Unified SCCP and SIP SRST System Administrator Guide OL-13143-04...
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POTS voice dial peers that are active during Cisco Unified CM fallback only. These temporary dial peers, which can be matched to voice ports (BRI, E&M, FXO, and PRI), allow Cisco Unified IP Phones access to trunk lines during Cisco Unified CM mode. When Cisco Unified SRST is active, all PSTN interfaces of the same type are treated as equivalent, and any port may be selected to place the outgoing PSTN call.
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Configuring interdigit timeout values involves specifying how long, in seconds, all Cisco Unified IP Phones attached to a Cisco Unified SRST router are to wait after an initial digit or a subsequent digit is dialed. The timeouts interdigit timer is enabled when a caller enters a digit and is restarted each time the caller enters subsequent digits until the destination address is identified.
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Router(config-cm-fallback)# exit Examples The following example sets the interdigit timeout value to 5 seconds for all Cisco Unified IP Phones. In this example, 5 seconds are the elapsed time after which an incompletely dialed number times out. For example, a caller who dials nine digits (408555010) instead of the required ten digits (4085550100) will hear a busy tone after the second timeout elapses.
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How to Configure Cisco Unified SCCP SRST a Cisco Unified SRST router and try to make a call from that phone, the call will be considered an incoming call to the router and voice port. If you make a call to the FXS phone, the call will be considered outgoing.
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Cisco Unified IP Phone dial peers and directory numbers created during fallback: call-manager-fallback cor outgoing LockforPhoneC 1 5010 - 5020 The following example shows how to set the dial-peer COR parameter for incoming calls to the Cisco IP phone dial peers and directory numbers in the default COR list: call-manager-fallback cor incoming LockforPhoneC default The following example shows how sub- and super-COR sets are created.
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Finally, the COR list is applied to the individual phone numbers. call-manager-fallback max-conferences 8 cor incoming engineering 1 1001 - 1001 cor incoming hr 2 1002 - 1002 cor incoming manager 3 1003 - 1008 Cisco Unified SCCP and SIP SRST System Administrator Guide OL-13143-04...
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Extension 1001 to call 734... numbers, 911, and 316..• Extension 1002 to call 734..., 1800 numbers, 911, and 316..• Extension 1003 to 1008 to call all of the possible Cisco Unified SRST router numbers • All extensions to call 316..•...
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19:00 07:00 after-hours block day thu 19:00 07:00 after-hours block day fri 19:00 07:00 after-hours block day sat 13:00 12:00 after-hours block day sun 12:00 07:00 Cisco Unified SCCP and SIP SRST System Administrator Guide OL-13143-04...
In voice register pool configuration, you can now configure several new options per pool (a pool can be one phone or a group of phones). There is also a new voice register global configuration mode for Cisco Unified SIP SRST. In voice register global mode, you can globally assign characteristics to phones.
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Example: maximum number of SIP voice register pools supported by Router(config-register-global)# max-pool 10 the Cisco Unified SIP SRST router. The upper limit of voice register pools is version- and platform-dependent; see Cisco IOS command-line interface (CLI) help. Default is 0.
SCCP IP phones may be forwarded to SIP phones. Cisco Unity or other voice messaging systems connected by a SIP trunk or SIP user agent are able to pass a message-waiting indicator (MWI) when a message is left. The SIP phone then displays the MWI when indicated by the voice messaging system.
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Configuring Call Handling How to Configure Cisco Unified SIP SRST In a typical Cisco Unified SIP SRST setup, the call-forward b2bua mailbox command is not used; however, it is likely to be used in a Cisco Unified SIP Communications Manager Express (CME) environment.
32 digits during a specified time of day, day of week, or date. Cisco Unified SIP SRST provides SIP endpoints the same time-based call blocking mechanism that is currently provided for SCCP phones. The call blocking feature supports all incoming calls, including incoming SIP and analog FXS calls.
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In SRST (call-manager-fallback configuration mode), there is no phone- or pin-based exemption to after-hours call blocking. However, in Cisco Unified SIP SRST (voice register pool mode), individual IP phones can be exempted from all call blocking using the after-hours exempt command.
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Enters voice register pool configuration mode. voice register pool tag Use this command to control which registrations are • accepted or rejected by a Cisco Unified SIP SRST Example: device. Router(config)# voice register pool 12 Cisco Unified SCCP and SIP SRST System Administrator Guide...
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19:00 07:00 after-hours day wed 19:00 07:00 after-hours day thu 19:00 07:00 after-hours day fri 19:00 07:00 The following example exempts a Cisco SIP phone pool from the configured blocking criteria: voice register pool 1 after-hour exempt Verification To verify the feature’s configuration, enter one of the following commands:...
SIP Call Hold and Resume Cisco Unified SRST supports the ability for SIP phones to place calls on hold and to resume from calls placed on hold. This also includes support for a consultative hold where A calls B, B places A on hold, B calls C, and B disconnects from C and then resumes with A.
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4 gain -6 after-hours block pattern 1 2417 after-hours date Dec 25 12:01 20:00 ! ! line con 0 exec-timeout 0 0 line aux 0 line vty 0 4 Cisco Unified SCCP and SIP SRST System Administrator Guide OL-13143-04...
Enabling three-party G.711 ad hoc conferencing involves configuring the maximum number of simultaneous three-party conferences supported by the Cisco Unified SRST router. For conferencing to be available, an IP phone must have a minimum of two lines connected to one or more buttons. See the “Configuring a Secondary Dial Tone”...
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Cisco 3800 series router: 24 • Step 3 Exits call-manager-fallback configuration mode. exit Example: Router(config-cm-fallback)# exit Examples The following example configures up to eight simultaneous three-way conferences on a router: call-manager-fallback max-conferences 8 Cisco Unified SCCP and SIP SRST System Administrator Guide OL-13143-04...
Configuring Call Handling Where to Go Next Defining XML API Schema The Cisco IOS commands in this section allow you to specify parameters associated with the XML API. For more information, see XML Provisioning Guide for Cisco CME/SRST. See the “Configuring Cisco...
• The SRST router must have a certificate; a certificate can be generated by a third party or by the Cisco IOS certificate authority (CA). The Cisco IOS CA can run on the same gateway as Cisco Unified SRST. Certificate trust lists (CTLs) on Cisco Unified Communications Manager must be enabled. For •...
Set the clock, either manually or by using Network Time Protocol (NTP). Setting the clock ensures • synchronicity with Cisco Unified Communications Manager. Enable the IP HTTP server (Cisco IOS processor) with the ip http server command, if not already • enabled. For more information on public key infrastructure (PKI) deployment, see the...
• IP Phone endpoints or from a Cisco Unified IP Phone to a gateway endpoint, a lock icon is displayed on the IP phones. The lock indicates security only for the IP leg of the call. Security of the PSTN leg is not implied.
Cisco Unified Communications Manager. Assuming that Cisco Unified Communications Manager is configured to fallback to Cisco Unified SRST, the TLS connection between the Cisco Unified IP Phones and the secure Cisco Unified SRST Router is also established. If the WAN link or Cisco Unified Communications Manager fails, call control reverts to the Cisco Unified SRST router.
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Cisco Unified Communications Manager inserts the SRST router certificate in the Cisco Unified IP Phone configuration file and downloads the configuration files to the phones. The secure Cisco Unified IP Phone uses the certificate to authenticate the SRST router during fallback operations. The credentials service runs on default TCP port 2445.
Once the certificate is generated, fill in the name of the certificate (or the name of the trustpoint in IOS) in the "trustpoint" entry. This certificate for the Credentials Server on the Secure SRST will be seamlessly exported to the Cisco Unified CM when requested in “Adding an SRST Reference to Cisco Unified Communications...
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Cisco Unified IP Phone TCP port plus 443; that is, port 2443 on a Cisco Unified SRST router. In case of WAN failure, the Cisco Unified IP Phone starts Cisco Unified SRST registration. SRST Mode The Cisco Unified IP Phone registers with the —...
Overview of the Process of Secure SRST Authentication and Encryption Process Steps Description or Detail The CA server, whether it is a Cisco IOS router CA or a third-party CA, issues a device certificate to the SRST gateway, enabling credentials service. Optionally, the certificate can be self-generated by the SRST router using a Cisco IOS CA server.
The following configuration sections ensure that the secure Cisco Unified SRST Router and the Cisco Unified IP Phones can request mutual authentication during the TLS handshake. The TLS handshake occurs when the phone registers with the Cisco Unified SRST Router, either before or after the WAN link fails.
The Cisco IOS certificate server provides a certificate generation option to users who do not have a third-party CA in their network. The Cisco IOS certificate server can run on the SRST router or on a different Cisco IOS router.
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Step 4 issuer-name DN-string Sets the CA issuer name to the specified distinguished name (DN-string). The default value is as follows: issuer-name CN=cs-label. Example: Router (cs-server)# issuer-name CN=srstcaserver Cisco Unified SCCP and SIP SRST System Administrator Guide OL-13143-04...
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Autoenrolling and Authenticating the Secure Cisco Unified SRST Router to the CA Server The secure Cisco Unified SRST Router needs to define a trustpoint; that is, it must obtain a device certificate from the CA server. The procedure is called certificate enrollment. Once enrolled, the secure Cisco Unified SRST Router can be recognized by Cisco Unified Communications Manager as a secure SRST router.
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Specifies the URL of the CA to which your • router should send certificate requests. Example: Router(ca-trustpoint)# enrollment url If you are using Cisco proprietary SCEP for enrollment, • http://10.1.1.22 url must be in the form http://CA_name, where CA_name is the host Domain Name System (DNS) name or IP address of the Cisco IOS CA.
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Configuring Secure SRST for SCCP and SIP How to Configure Secure Unified SRST Examples The following example autoenrolls and authenticates the Cisco Unified SRST router: Router(config)# crypto pki trustpoint srstca Router(ca-trustpoint)# enrollment url http://10.1.1.22 Router(ca-trustpoint)# revocation-check none Router(ca-trustpoint)# exit Router(config)# crypto pki authenticate srstca...
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Verifying Certificate Enrollment If you used the Cisco IOS certificate server as your CA, use the show running-config command to verify certificate enrollment or the show crypto pki server command to verify the status of the CA server. SUMMARY STEPS...
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<serialnum>.cer Enabling Credentials Service on the Secure Cisco Unified SRST Router Once the Cisco Unified SRST Router has its own certificate, you need to provide Cisco Unified Communications Manager the certificate. Enabling credentials service allows Cisco Unified Communications Manager to retrieve the secure SRST device certificate and place it in the configuration file of the Cisco Unified IP Phone.
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Router(config-credentials)# trustpoint srstca The trustpoint name should be the same as the one • declared in the “Autoenrolling and Authenticating the Secure Cisco Unified SRST Router to the CA Server” section on page 188. Step 4 Exits credentials configuration mode.
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• Importing Phone Certificate Files in PEM Format to the Secure SRST Router This task completes the tasks required for Cisco IP Unified Phones to authenticate secure SRST. Cisco Unified SCCP and SIP SRST System Administrator Guide OL-13143-04...
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For systems running Cisco Unified Communications Manager 4.X.X and earlier versions, the secure Cisco Unified SRST Router must retrieve phone certificates so that it can authenticate Cisco Unified IP phones during the TLS handshake. Different certificates are used for different Cisco Unified IP Phones.
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Authenticating the Imported Certificates on the Cisco Unified SRST Router To authenticate certificates on the Cisco Unified SRST router, perform these steps. Restrictions HTTP automatic enrollment from Cisco Unified Communications Manager through a virtual web server is not supported. SUMMARY STEPS...
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Step Router(config)# crypto pki authenticate CAPF What to Do Next Update the certificates in Cisco Unified CM. See the “Configuring a Secure Survivable Remote Site Telephony (SRST) Reference” chapter in the appropriate version of Cisco Unified Communications Manager Security Guide.
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End with a blank line or the word "quit" on a line by itself MIIDqDCCApCgAwIBAgIQdhL5YBU9b59OQiAgMrcjVjANBgkqhkiG9w0BAQUFADAu MRYwFAYDVQQKEw1DaXNjbyBTeXN0ZW1zMRQwEgYDVQQDEwtDQVAtUlRQLTAwMTAe Fw0wMzAyMDYyMzI3MTNaFw0yMzAyMDYyMzM2MzRaMC4xFjAUBgNVBAoTDUNpc2Nv IFN5c3RlbXMxFDASBgNVBAMTC0NBUC1SVFAtMDAxMIIBIDANBgkqhkiG9w0BAQEF AAOCAQ0AMIIBCAKCAQEArFW77Rjem4cJ/7yPLVCauDohwZZ/3qf0sJaWlLeAzBlq Rj2lFlSij0ddkDtfEEo9VKmBOJsvx6xJlWJiuBwUMDhTRbsuJz+npkaGBXPOXJmN Vd54qlpc/hQDfWlbrIFkCcYhHws7vwnPsLuy1Kw2L2cP0UXxYghSsx8H4vGqdPFQ NnYy7aKJ43SvDFt4zn37n8jrvlRuz0x3mdbcBEdHbA825Yo7a8sk12tshMJ/YdMm vny0pmDNZXmeHjqEgVO3UFUn6GVCO+K1y1dUU1qpYJNYtqLkqj7wgccGjsHdHr3a U+bw1uLgSGsQnxMWeMaWo8+6hMxwlANPweufgZMaywIBA6OBwzCBwDALBgNVHQ8E BAMCAYYwDwYDVR0TAQH/BAUwAwEB/zAdBgNVHQ4EFgQU6Rexgscfz6ypG270qSac cK4FoJowbwYDVR0fBGgwZjBkoGKgYIYtaHR0cDovL2NhcC1ydHAtMDAxL0NlcnRF bnJvbGwvQ0FQLVJUUC0wMDEuY3Jshi9maWxlOi8vXFxjYXAtcnRwLTAwMVxDZXJ0 RW5yb2xsXENBUC1SVFAtMDAxLmNybDAQBgkrBgEEAYI3FQEEAwIBADANBgkqhkiG 9w0BAQUFAAOCAQEAq2T96/YMMtw2Dw4QX+F1+g1XSrUCrNyjx7vtFaRDHyB+kobw dwkpohfkzfTyYpJELzV1r+kMRoyuZ7oIqqccEroMDnnmeApc+BRGbDJqS1Zzk4OA c6Ea7fm53nQRlcSPmUVLjDBzKYDNbnEjizptaIC5fgB/S9S6C1q0YpTZFn5tjUjy WXzeYSXPrcxb0UH7IQJ1ogpONAAUKLoPaZU7tVDSH3hD4+VjmLyysaLUhksGFrrN phzZrsVVilK17qpqCPllKLGAS4fSbkruq3r/6S/SpXS6/gAoljBKixP7ZW2PxgCU Cisco Unified SCCP and SIP SRST System Administrator Guide OL-13143-04...
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Keys generated ..... Yes (General Purpose) Issuing CA authenticated ..Yes Certificate request(s) ..Yes Cisco Unified Communications Manager 5.0 and Later Versions Example The following example shows the configuration for the four certificates (CAPF, CiscoCA, CiscoManufactureCA, and CiscoRootCA2048) that are required for systems running Cisco Unified Communications Manager 5.0:...
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Certificate has the following attributes: Fingerprint MD5: 2G3LZ6B7 2R1995ER 6KE4WE72 3E528BB8 Fingerprint SHA1: M9912245 5C130ED2 24762JBC 3E528VF8 956E8S5H % Do you accept this certificate? [yes/no]: y Trustpoint CA certificate accepted. % Certificate successfully imported Cisco Unified SCCP and SIP SRST System Administrator Guide OL-13143-04...
The following procedure describes how to add an SRST reference to Cisco Unified Communications Manager. Before following this procedure, verify that credentials service is running in the Cisco Unified SRST Router. Cisco Unified Communications Manager connects to the Cisco Unified SRST Router for its device certificate.
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For complete information about adding a device pool to Cisco Unified Communications Manager, see the “Device Pool Configuration” section in Cisco Unified Communications Manager Administration Guide for the Cisco Unified Communications Manager version that you are running. All Cisco Unified CM administration guides are at the following URL: http://www.cisco.com/en/US/products/sw/voicesw/ps556/prod_maintenance_guides_list.html...
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The Certificate Authority Proxy Function (CAPF) process allows supported devices, such as Cisco Unified Communications Manager, to request LSC certificates from Cisco Unified IP Phones. The CAPF utility generates a key pair and certificate that are specific for CAPF, and the utility copies this certificate to all Cisco Unified Communications Manager servers in the cluster.
How to Configure Secure Unified SRST Enabling SRST Mode on the Secure Cisco Unified SRST Router To configure secure SRST on the router to support the Cisco Unified IP Phone functions, use the following commands beginning in global configuration mode.
How to Configure Secure Unified SRST Command or Action Purpose Step 4 Enables the router to receive messages from the Cisco IP ip source-address ip-address [port port] Phones through the specified IP addresses and provides for strict IP address verification. The default port number Example: is 2000.
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• Verifying Phone Status and Registrations To verify or troubleshoot Cisco Unified IP Phone status and registration, complete the following steps beginning in privileged EXEC mode. You can verify Phone Status and Registrations in secure SCCP SRST after you have performed the...
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IP:10.1.1.40 32862 7970 keepalive 390 max_line 8 button 1: dn 2 number 2010 CM Fallback CH1 IDLE Step 2 Use this command to display Cisco IP Phone status show ephone offhook and quality for all phones that are off hook. In this...
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16382 to 10.1.1.40 16382 via 10.1.1.40 G711Ulaw64k 160 bytes no vad Tx Pkts 295 bytes 49468 Rx Pkts 277 bytes 46531 Lost Jitter 0 Latency 0 callingDn -1 calledDn 11 Cisco Unified SCCP and SIP SRST System Administrator Guide OL-13143-04...
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Purpose Step 3 Use this command to show the call status for all show voice call status voice ports on the Cisco Unified SRST router. This command is not applicable for calls between two Example: POTS dial peers. CallID CID ccVdb Port DSP/Ch Called # Codec Dial-peers 0x1164 2BFE 0x8619A460 50/0/35.0 2014 g711ulaw...
0x1186 2C31 0x861A56E8 50/0/36.0 2030 g711ulaw 20036/20033 0x1187 2C31 0x86185318 50/0/33.0 *2030 g711ulaw 20033/20036 18 active calls found Step 4 Use this command to debug the process of Cisco IP debug ephone register phone registration. Example: Router# debug ephone register EPHONE registration debugging is enabled *Jun 29 09:16:02.180: New Skinny socket accepted [2]...
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1 ref 7 called 6001 calling 6000 origcalled 6001 calltype 1 *Jan 11 18:33:16.047:ephone-3[3]:Call Info for chan *Jan 11 18:33:16.047:ephone-3[3]:Original Called Name 6001 *Jan 11 18:33:16.047:ephone-3[3]:6000 calling *Jan 11 18:33:16.047:ephone-3[3]:6001 *Jan 11 18:33:16.047:ephone-3[3]:Ringer Inside Ring Cisco Unified SCCP and SIP SRST System Administrator Guide OL-13143-04...
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Secure SCCP SRST: Example This section provides a configuration example to match the identified configuration tasks in the previous sections. This example does not include using a third-party CA; it assumes the use of the Cisco IOS certificate server to generate your certificates.
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28800 crypto isakmp key cisco123 address 10.1.1.13 ! The crypto key should match the key configured on Cisco Unified Communications Manager. ! The crypto IPSec configuration should match your Cisco Unified Communications Manager configuration. crypto ipsec transform-set rtpset esp-des esp-md5-hmac crypto map rtp 1 ipsec-isakmp set peer 10.1.1.13...
Restrictions for Configuring Secure SIP Call Signaling and SRTP Media with Cisco SRST SIP phones may be configured on the Cisco Unified CM with an authenticated device security mode. The Cisco Unified CM ensures integrity and authentication for the phone using a TLS connection with NULL-SHA cipher for signaling.
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Device Pool. The SRST Reference profile must have the "Is SRST Secure" checkbox selected if SIP/TLS communication is desired in the event of a WAN failure. All Cisco Unified IP Phones must have their firmware updated to version 8.5(3) or later. Devices with Note firmware earlier than 8.5(3) will need to have a separate Device Pool and SRST Reference profile created...
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Example:: if SRTP is not available. Router(config-voi-serv)# srtp fallback Step 5 (Optional) Allows connections from SIP endpoints to H.323 allow-connections sip to h323 endpoints. Example: Router(config-voi-serv)# allow-connections sip to h323 Cisco Unified SCCP and SIP SRST System Administrator Guide OL-13143-04...
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Configuring SIP options for Secure SIP SRST This section explains how to configure secure SIP SRTP. SUMMARY STEPS enable configure terminal voice service voip url sip | sips srtp negotiate cisco Cisco Unified SCCP and SIP SRST System Administrator Guide OL-13143-04...
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Example: generate URLs in SIP format for VoIP calls. Router(conf-serv-sip)# url sips Step 6 Enables a Cisco IOS SIP gateway to negotiate the sending srtp negotiate cisco and accepting of RTP profiles in response to SRTP offers. Example: Router(conf-serv-sip)# srtp negotiate cisco...
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Configuring SIP User Agent for Secure SIP SRST This section explains how the strict-cipher limits the allowed encryption algorithms. SUMMARY STEPS sip-ua registrar ipv4:destination-address expires seconds xfer target dial-peer crypto signaling default trustpoint string [strict-cipher] Cisco Unified SCCP and SIP SRST System Administrator Guide OL-13143-04...
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[strict-cipher] during the TLS handshake. The trustpoint string keyword and argument refer to the gateway’s certificate generated as part of the enrollment process, using Cisco IOS public-key Example: infrastructure (PKI) commands. The strict-cipher keyword Router(config-sip-ua)# crypto signaling default...
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FastEthernet0/0 bind media source-interface FastEthernet0/0 Cisco Unified SCCP and SIP SRST System Administrator Guide OL-13143-04...
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FastEthernet0/1 description "WAN" connection to Cluster-B ip address 10.2.0.6 255.255.255.0 duplex auto speed auto sip-ua registrar ipv4:10.2.0.10 expires 3600 xfer target dial-peer crypto signaling default trustpoint 3745-SRST Cisco Unified SCCP and SIP SRST System Administrator Guide OL-13143-04...
No new or modified MIBs are supported by this To locate and download MIBs for selected platforms, Cisco IOS feature, and support for existing MIBs has not been releases, and feature sets, use Cisco MIB Locator found at the modified by this feature. following URL: http://www.cisco.com/go/mibs...
Command Reference Command Reference The following commands are introduced or modified in the feature or features documented in this section. For information about these commands, see the Cisco IOS Voice Command Reference at http://www.cisco.com/en/US/docs/ios/voice/command/reference/vr_book.html. For information about all Cisco IOS commands, use the Command Lookup Tool at http://tools.cisco.com/Support/CLILookup...
Table 4 lists the release history for this feature. Not all commands may be available in your Cisco IOS software release. For release information about a specific command, see the command reference documentation. Use Cisco Feature Navigator to find information about platform support and software image support.
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Configuring Secure SRST for SCCP and SIP Where to Go Next Cisco Unified SCCP and SIP SRST System Administrator Guide OL-13143-04...
Cisco Unified SRST also supports incoming and outgoing Session Initiation Protocol (SIP) calls to and from Cisco Unified IP phones and router voice gateway voice ports. SIP may be used in situations where the Cisco Unified SRST Router is separate from the PSTN gateway and the SRST and PSTN gateways are linked together using SIP (instead of H.323).
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Integrating Voice Mail with Cisco Unified SRST Information About Integrating Voice Mail with Cisco Unified SCCP SRST Figure 1 Cisco Unified Communications Manager Fallback with BRI or PRI Cisco Unified Communications Cisco Unified Manager gateway SRST gateway BRI/PRI WAN failure...
Integrating Voice Mail with Cisco Unified SRST How to Integrate Voice Mail with Cisco Unified SCCP and SIP SRST How to Integrate Voice Mail with Cisco Unified SCCP and SIP SRST This section contains the following tasks: Configuring Direct Access to Voice Mail, page 237 (Required) •...
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Integrating Voice Mail with Cisco Unified SRST How to Integrate Voice Mail with Cisco Unified SCCP and SIP SRST DETAILED STEPS Command or Action Purpose Step 1 (FXO or FXS and BRI or PRI) Defines a particular dial peer, dial-peer voice tag {pots | voatm | vofr |...
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Integrating Voice Mail with Cisco Unified SRST How to Integrate Voice Mail with Cisco Unified SCCP and SIP SRST Command or Action Purpose Step 4 (Optional for FXO or FXS) Specifies which digits to forward forward-digits {num-digit | all | extra} for voice calls.
2/0:23 Configuring Message Buttons To activate the message buttons on Cisco Unified IP phones connected to the Cisco Unified SCCP and SIP SRST router during Cisco Unified Communications Manager fallback, you must program a speed-dial number to the voice-mail system. The speed-dial number is dialed when message buttons on phones connected to the Cisco Unified SCCP and SIP SRST router are pressed during Cisco Unified CM fallback.
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Integrating Voice Mail with Cisco Unified SRST How to Integrate Voice Mail with Cisco Unified SCCP and SIP SRST SUMMARY STEPS call-manager-fallback voicemail phone-number call-forward busy directory-number call-forward noan directory-number timeout seconds exit voice register pool tag call-forward b2bua busy directory-number...
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The following example specifies 1101 as the speed-dial number that is issued when message buttons are pressed on Cisco Unified IP Phones connected to the Cisco Unified SRST router. All busy and unanswered calls are configured to be forwarded to the voice-mail number (1101).
Dial 2 to select the menu option for leaving messages in the extension number’s mailbox. For Cisco Unified SCCP SRST to forward a call to a busy or unanswered number to extension 6000’s mailbox, it must be programmed to issue a sequence of 1101#6000#2. As shown in...
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A tag can be up to three character from the DTMF tone set (A to D, 0 to 9, # and *). Voice-mail systems can use limited sets of DTMF tones. For example, Cisco Unity uses all DTMF tones but A to D. Tones can be defined in multiple ways.
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“Call Routing Instructions Using DTMF Digit Patterns” section on page 243). You can find information about how Cisco Unity handles voice-mail calls in the How to Transfer a Caller Directly into a Cisco Unity Mailbox document. Additional call-handling information can be found in the “Subscriber and Operator Orientation”...
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Integrating Voice Mail with Cisco Unified SRST How to Integrate Voice Mail with Cisco Unified SCCP and SIP SRST DETAILED STEPS Command or Action Purpose Step 1 Enters voice-mail integration mode and enables voice-mail vm-integration integration with DTMF and analog voice-mail systems.
The MWI relay mechanism is initiated after someone leaves a voice-mail message on the remote voice-mail message system. MWI relay is required when one Cisco Unity Voice Mail system is shared by multiple Cisco Unified SCCP SRST routers. SCCP SRST routers use the SIP Subscribe and Notify methods for MWI.
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Integrating Voice Mail with Cisco Unified SRST How to Integrate Voice Mail with Cisco Unified SCCP and SIP SRST SUMMARY STEPS call-manager-fallback mwi relay mwi reg-e164 exit sip-ua mwi-server {ipv4:destination-address | dns:host-name} [expires seconds] [port port] [transport {tcp | udp}] [unsolicited]...
The “Dial-Peer Configuration for Integration of Voice-Mail with Cisco Unified SCCP SRST” section of the example below shows a legacy dial-peer configuration for a local voice-mail system. The “Cisco Unified SCCP SRST Voice-Mail Integration Pattern Configuration” section must be compatible with your voice-mail system configuration.
6 FDN * CGN * Configuring Central Location Voice-Mail System (FXO and FXS): Example The “Dial-Peer Configuration for Integration of Voice-Mail with Cisco Unified SCCP SRST in Central Location” section of the example shows a legacy dial-peer configuration for a central voice-mail system.
6 FDN * CGN * Configuring Voice-Mail Access over FXO and FXS: Example The following example shows how to configure the Cisco Unified SCCP SRST router to forward unanswered calls to voice mail. In this example, the voice-mail number is 1101, the voice-mail system is connected to FXS voice port 1/1/1, and the voice mailbox numbers are 3001, 3002, and 3006.
“Configuring Call Handling” section on page 123. Voice Mail number associate with SIP phone message button in SRST is configured by Cisco Note Unified Communications Manager (CUCM), and not configurable by SIP SRST. The administrator needs to know the voice mail number set by CUCM to configure proper dial peer to voice mail system in SIP SRST.
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The SIP DTMF relay method is needed in the following situations: • When SIP is used to connect a Cisco Unified SRST system to a remote SIP-based IVR or voice-mail application, such as Cisco Unity. When SIP is used to connect a Cisco Unified SRST system to a remote SIP-PSTN voice gateway •...
DTMF Relay Using SIP Notify (Nonstandard) 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.
SIP DNS SRV version:2 (rfc 2782) NAT Settings for the SIP-UA Role in SDP:NONE Check media source packets:DISABLED Maximum duration for a telephone-event in NOTIFYs:2000 ms SIP support for ISDN SUSPEND/RESUME:ENABLED Cisco Unified SCCP and SIP SRST System Administrator Guide OL-13143-04...
If you want to configure video parameters, see the “Setting Video Parameters” section on page 257. For additional information, see the “Additional References” section on page 26 in the “Cisco Unified SCCP and SIP SRST Feature Overview” chapter. Cisco Unified SCCP and SIP SRST System Administrator Guide OL-13143-04...
Cisco Unified IP phone is up. From a PC with Cisco Unified Video Advantage 1.02 or a later version installed, ensure that the line between the Cisco Unified Video Advantage and the Cisco Unified IP phone is green. For more information, see Cisco Unified Video Advantage End User Guides.
During audio-only calls, video messages are skipped. Information About Setting Video Parameters 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...
Information About Setting Video Parameters 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.
Call signaling is managed by the Cisco Unified CME router, and the media stream is directly connected between the two video-enabled SCCP endpoints on the same router. Video-related commands and flow-control messages are forwarded to the other endpoint.
How to Set Video Parameters for Cisco Unified SRST 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.
VoIP calls. Example: Router(config-serv-h323)# call start slow Verifying Cisco Unified SRST Use the following procedure to verify that the Cisco Unified SRST feature is enabled and to verify Cisco Unified IP phone configuration settings. SUMMARY STEPS enable show running config...
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Example: Router# show call-manager-fallback all Use the Settings display on the Cisco Unified IP phones in your network to verify that the default router Note IP address on the phones matches the IP address of the Cisco Unified SRST router.
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Setting Video Parameters How to Set Video Parameters for Cisco Unified SRST 7910: 34 7935: 34 7936: 34 7940: 34 7960: 34 7970: 34 Log (table parameters): max-size: 150 retain-timer: 15 transfer-system full-consult local directory service: enabled. ephone-dn 1 number 1001...
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Setting Video Parameters How to Set Video Parameters for Cisco Unified SRST preference 0 secondary 9 huntstop call-waiting beep ephone-dn 10 preference 0 secondary 9 huntstop call-waiting beep ephone-dn 11 preference 0 secondary 9 huntstop call-waiting beep ephone-dn 12 preference 0 secondary 9...
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Setting Video Parameters How to Set Video Parameters for Cisco Unified SRST timeout ringing 8 voice-port 50/0/2 station-id number 1002 station-id name 1002 timeout ringing 8 voice-port 50/0/3 voice-port 50/0/4 voice-port 50/0/5 voice-port 50/0/6 voice-port 50/0/7 voice-port 50/0/8 voice-port 50/0/9...
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Setting Video Parameters How to Set Video Parameters for Cisco Unified SRST dial-peer voice 20058 pots huntstop progress_ind setup enable 3 port 50/0/4 dial-peer voice 20059 pots huntstop progress_ind setup enable 3 port 50/0/5 dial-peer voice 20060 pots huntstop progress_ind setup enable 3...
Setting Video Parameters How to Set Video Parameters for Cisco Unified SRST port 50/0/16 dial-peer voice 20071 pots huntstop progress_ind setup enable 3 port 50/0/17 dial-peer voice 20072 pots huntstop progress_ind setup enable 3 port 50/0/18 dial-peer voice 20073 pots...
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0 to 10000000. The default is 10000000. Example: Router(conf-cm-fallback-video)# maximum bit-rate 256 Examples The following example shows the configuration for video with Cisco Unified SRST: call-manager-fallback video maximum bit-rate 384 max-conferences 2 gain -6 transfer-system full-consult ip source-address 10.0.1.1 port 2000...
• debug cch323 video: Enables video debugging trace on the H.323 SPI. – debug ephone detail: Debugs all Cisco Unified IP phones that are registered to the router and – displays error and state levels. debug h225 asn1: Displays Abstract Syntax Notation One (ASN.1) contents of H.225 messages –...
Monitoring and Maintaining Cisco Unified SRST Revised: February 3, 2011 To monitor and maintain Cisco Unified Survivable Remote Site Telephony (SRST), use the following commands in privileged EXEC mode. Command Purpose Displays the detailed configuration of all the Router# show call-manager-fallback all Cisco Unified IP phones, voice ports, and dial peers of the ...
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Router # show voice register dial-peers Displays all config voice register dn detail info. Router # show voice register dn all Displays specific voice register dn detail info. Router # show voice register dn <tag> Cisco Unified SCCP and SIP SRST System Administrator Guide OL-13143-04...
Information About Cisco Unified SIP SRST Features Using Redirect Mode, page 274 • How to Configure Cisco Unified SIP SRST Features Using Redirect Mode, page 274 • Configuration Examples for Cisco Unified SIP SRST Features Using Redirect Mode, page 278 •...
Cisco IOS voice gateway. Prior to this enhancement, an attempt by a SIP phone to contact another local SIP phone using the Cisco IOS voice gateway as if it were a SIP proxy or redirect server would fail. However, the Cisco IOS voice gateway can now act as a SIP redirect server. The voice gateway responds to the originator with a SIP Redirect message, allowing the SIP phone that originated the call to establish a call to its destination.
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Appendix A: Configuring Cisco Unified SIP SRST Features Using Redirect Mode How to Configure Cisco Unified SIP SRST Features Using Redirect Mode Configuring Call Redirect Enhancements to Support Calls Globally To enable global IP-to-IP call redirection for all VoIP dial peers, use voice service configuration mode.
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Configuring Call Redirect Enhancements to Support Calls on a Specific VoIP Dial Peer To enable IP-to-IP call redirection for a specific VoIP dial peer, configure it on an inbound dial peer in dial-peer configuration mode. The default application on Cisco Unified SIP SRST supports IP-to-IP redirection.
Router(config-dial-peer)# end Configuring Sending 300 Multiple Choice Support Prior to Cisco IOS Release 12.2(15)ZJ, when a call was redirected, the SIP gateway would send a 302 Moved Temporarily message. The first longest match route on a gateway (dial-peer destination pattern) was used in the Contact header of the 302 message.
Appendix A: Configuring Cisco Unified SIP SRST Features Using Redirect Mode Configuration Examples for Cisco Unified SIP SRST Features Using Redirect Mode DETAILED STEPS Command or Action Purpose Step 1 Enables privileged EXEC mode. enable Enter your password if prompted.
Appendix A: Configuring Cisco Unified SIP SRST Features Using Redirect Mode Configuration Examples for Cisco Unified SIP SRST Features Using Redirect Mode Cisco Unified SIP SRST: Example This section provides a configuration example to match the configuration tasks in the previous sections.
Appendix A: Configuring Cisco Unified SIP SRST Features Using Redirect Mode Where to Go Next Rule 0 94 91 ! Sets up proxy monitoring. call fallback active dial-peer cor custom name 95 name 94 name 91 ! Configures COR values to be applied to the voice register pool.
MOH eliminates the need to stream MOH across a WAN and saves bandwidth. Finding Feature Information in This Module Your Cisco Unified CME version may not support all of the features documented in this module. For a list of the versions in which each feature is supported, see the “Feature Information for Cisco Unified SRST...
Appendix B: Integrating Cisco Unified Communications Manager and Cisco Unified SRST to Use Cisco Unified SRST Prerequisites for Using Cisco Unified SRST Gateways as a Multicast MOH Resource Prerequisites for Using Cisco Unified SRST Gateways as a Multicast MOH Resource Multicast MOH for H.323 and MGCP is supported on Cisco Unified CM 3.1.1 and higher versions.
Cisco Unified SRST gateways are configured to send audio packets from their flash files to port number 16384 and IP address 239.1.1.1. Cisco Unified CM and the IP phones are spoofed and behave as if Cisco Unified CM were originating the multicast MOH.
G.711 is the only codec used by a central Cisco Unified CM and three branches. In some cases, a Cisco Unified CM system may use additional codecs. For example, for bandwidth savings, Cisco Unified CM may use G.711 for multicast MOH and G.729 for phone conversations.
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Cisco Unified SRST and other gateways are not. When the central site and Branch 3 phone users are put on hold by other IP phones in the Cisco Unified CM system, MOH is originated by Cisco Unified CM. When Branch 1 and Branch 2 phone users are put on hold by other phone users in the Cisco Unified CM system, MOH is originated by the Cisco Unified SRST gateways.
MOH live feed provides live feed MOH streams from an audio device connected to an E&M or FXO port to Cisco IP phones in SRST mode. Music from a live feed is from a fixed source and is continuously fed into the MOH playout buffer instead of being read from a flash file.
Cisco Unified Communications Manager MOH audio files must reach the WAN and another set must not. Audio packets from the central Cisco Unified CM must cross the WAN to reach branches running Cisco Unified CM. For branches running Cisco Unified SRST, the packets must not reach the WAN.
WAN and the remote phones, the configured Cisco Unified CM MOH IP port and address information are still used by Cisco Unified CM to tell the phones which multicast IP address to listen to for MOH (for the MOH sourced by SRST).
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The MOH audio source is a file from which Cisco Unified CM transmits RTP packets. You can create an audio file or use the default audio file. For Cisco Unified SRST multicast MOH, only one audio source can be used, even if, for example, one out of 500 sites uses Cisco Unified SRST multicast MOH. In...
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The simplest way to create an audio source is to use the default audio source. Whether you use a default Cisco Unified CM MOH audio source or you create one, the MOH audio source must be configured for multicasting in the MOH Audio Source Configuration window.
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239.1.1.1 and 16386, and so forth. It is important to configure a Cisco Unified CM port number and IP address that use a G.711 audio source for Cisco Unified SRST multicast MOH. If Cisco Unified CM multicast MOH is also being used on gateways that do not have Cisco Unified SRST and use a different codec, such as G.729, ensure that the...
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IP address to remain unchanged. Choose IP Address if you want IP addresses to be incremented and the port number to remain unchanged. If all of your branches run Cisco Unified SRST and thus use G.711 for MOH, use either setting •...
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Appendix B: Integrating Cisco Unified Communications Manager and Cisco Unified SRST to Use Cisco Unified SRST How to Use Cisco Unified SRST Gateways as a Multicast MOH Resource access-list access-list-number deny ip-address Configures the access list mechanism for filtering frames by IP address. For the ip-address argument, enter the MOH IP address that you want to prevent from going over the WAN.
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Appendix B: Integrating Cisco Unified Communications Manager and Cisco Unified SRST to Use Cisco Unified SRST How to Use Cisco Unified SRST Gateways as a Multicast MOH Resource Use the following procedure to create an MRG and MRGL, to enable MOH multicast, and to configure gateways.
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Configure the Region Configuration window. If the Cisco Unified CM system uses G.711 only, all of the central sites and their constituent branches for the MOH region must be set to G.711. If a Cisco Unified CM system has a combination of branches that do and do not run Cisco Unified SRST multicast MOH and the branches that do not run Cisco Unified SRST require a different codec for Cisco Unified Communications Manager multicast MOH, they must be configured accordingly.
MOH is heard. If multicast is not enabled on the WAN, place an IP phone on the same subnet as the Cisco Unified Communications Manager MOH server and verify that MOH can be heard. Because the IP phone and the MOH server are on the same subnet, no multicast routing capabilities in the network are required.
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The Cisco Unified SRST gateways must run Cisco IOS Release 12.2(15)ZJ2 or a later release. • The flash memory in each of the Cisco Unified SRST gateways must have an MOH audio file. The • MOH file can be in .wav or .au file format, but must contain 8-bit 8-kHz data, such as an a-law or mu-law data format.
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Appendix B: Integrating Cisco Unified Communications Manager and Cisco Unified SRST to Use Cisco Unified SRST How to Use Cisco Unified SRST Gateways as a Multicast MOH Resource max-dn max-directory-number moh filename multicasting-enabled multicast moh multicast-address port port [route ip-address-list]...
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Appendix B: Integrating Cisco Unified Communications Manager and Cisco Unified SRST to Use Cisco Unified SRST How to Use Cisco Unified SRST Gateways as a Multicast MOH Resource Command or Action Purpose Step 9 (Optional if the route keyword is not used in the ip source-address ip-address [port port] multicast moh command.
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Appendix B: Integrating Cisco Unified Communications Manager and Cisco Unified SRST to Use Cisco Unified SRST How to Use Cisco Unified SRST Gateways as a Multicast MOH Resource Command or Action Purpose Step 14 Enables multicast of MOH from a branch office...
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Appendix B: Integrating Cisco Unified Communications Manager and Cisco Unified SRST to Use Cisco Unified SRST How to Use Cisco Unified SRST Gateways as a Multicast MOH Resource Verifying Basic Cisco Unified SRST Multicast MOH Streaming Use the following procedure to verify that multicast MOH packets are configured with the multicast moh command.
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Rebooting may also clear the error. Verifying Cisco Unified SRST MOH to PSTN Use the following procedure to verify Cisco Unified CM control of MOH (the WAN link is up) and that multicast MOH packets transmit over a public switched telephone network (PSTN).
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If the PSTN caller hears tone on hold (TOH) instead of MOH, two problems are probable: • Cisco Unified CM has failed to activate MOH and has used TOH as a fallback. To verify that – this is the case, see the “Verifying Cisco Unified Communications Manager Multicast MOH”...
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• If the POTS caller on hold does not hear a sound, Cisco Unified CM has successfully completed the multicast MOH handshaking with the Cisco Unified SRST gateway, and the gateway is failing to pick up the locally generated multicast RTP stream.
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Appendix B: Integrating Cisco Unified Communications Manager and Cisco Unified SRST to Use Cisco Unified SRST How to Use Cisco Unified SRST Gateways as a Multicast MOH Resource The Incoming Interface field shows where the Cisco Unified SRST gateway is to receive the –...
If no MOH is heard and the Cisco Unified SRST MOH signaling is multicasting, connect a sniffer to the PC port on the back of IP phone. If the IP phone and Cisco Unified SRST gateway are connected to the same subnet, multicast RTP packets must be detected at all times, even when the IP phone was not placed on hold.
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The Cisco Unified SRST router uses the audio stream from the call as the source for the MOH stream, displacing any audio stream that is available from a flash file. An example of an MOH stream received over an incoming call is an external H.323-based server device that calls the directory number to deliver...
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Appendix B: Integrating Cisco Unified Communications Manager and Cisco Unified SRST to Use Cisco Unified SRST How to Use Cisco Unified SRST Gateways as a Multicast MOH Resource SUMMARY STEPS voice-port port input gain decibels auto-cut-through (E&M only) operation 4-wire (E&M only) signal immediate (E&M only)
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Appendix B: Integrating Cisco Unified Communications Manager and Cisco Unified SRST to Use Cisco Unified SRST How to Use Cisco Unified SRST Gateways as a Multicast MOH Resource Setting Up the Directory Numbers on the Cisco Unified SRST Gateway After setting up the voice port, create a dial peer and give the voice port a directory number with the destination-pattern command.
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Appendix B: Integrating Cisco Unified Communications Manager and Cisco Unified SRST to Use Cisco Unified SRST How to Use Cisco Unified SRST Gateways as a Multicast MOH Resource DETAILED STEPS Command or Action Purpose Step 1 Enters call-manager-fallback configuration mode.
Appendix B: Integrating Cisco Unified Communications Manager and Cisco Unified SRST to Use Cisco Unified SRST Configurations Examples for Cisco Unified SRST Gateways Command or Action Purpose Step 4 Specifies that this telephone number is to be used for an...
Feature Information for Cisco Unified SRST as a Multicast MOH Resource multicast moh 239.1.1.1 port 16384 route 172.21.51.143 10.1.1.1 The multicast IP address and port must match the IP address and the port number that Cisco Unified CM Note is configured to use for multicast MOH. If you are using different codecs for MOH, these might not be the base IP address and port, but an incremented IP address or port number.
Appendix B: Integrating Cisco Unified Communications Manager and Cisco Unified SRST to Use Cisco Unified SRST Where to Go Next Table 3 lists the Cisco Unified SRST version that introduced support for a given feature. Unless noted Note otherwise, subsequent versions of Cisco Unified SRST software also support that feature.
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Appendix B: Integrating Cisco Unified Communications Manager and Cisco Unified SRST to Use Cisco Unified SRST Where to Go Next Cisco Unified SCCP and SIP SRST System Administrator Guide OL-13143-04...
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Call Blocking by Time and Date call waiting call blocking configuration enabling on dual-line phone called number ccm-manager fallback-mgcp command 65, 67 digit translation rules cdn (called number) Cisco Unified SCCP and SIP SRST System Administrator Guide OL-13143-04...
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Cisco IP Phone 7940 message button for voice mail language display outgoing calls Cisco IP Phone 7941G and Cisco IP Phone 7941GE ringing timeout default Cisco IP Phone 7960 ring sound language display SIP proxy...
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Cisco SRST V2.1 in pattern direct command for converting abbreviated extension numbers to E.164 numbers FECC (far-end camera control) digit collect kpml command firmware, for video digit translation rules flow-around mode Cisco Unified SCCP and SIP SRST System Administrator Guide OL-13143-04...
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FXO setting up for Cisco IP phone display hookflash limit-dn command analog transfer using local call transfer host command configuring hunting local consultation dial peer configuring huntstop Cisco Unified SCCP and SIP SRST System Administrator Guide OL-13143-04...
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CallManager gateway for voice mail with operation command BRI/PRI access option 150 ip command 70, 71 redirect ip2ip command outgoing calls registrar server command configuring SIP networks remote call transfer Cisco Unified SCCP and SIP SRST System Administrator Guide OL-13143-04...
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147, 149 show ephone-dn command transfer patterns show ephone-dn loopback command transfer-system command show ephone-dn summary command translate command show ephone offhook command 270, 271 translation-profile command show ephone phone-load command Cisco Unified SCCP and SIP SRST System Administrator Guide OL-13143-04...
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Cisco SRST handles routing of calls voicemail command 241, 243 voice register global command voice register pool command 109, 112, 164, 167, 169 voice service voip command Cisco Unified SCCP and SIP SRST System Administrator Guide OL-13143-04...
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Index Cisco Unified SCCP and SIP SRST System Administrator Guide OL-13143-04...
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