Cisco Unified Survivable Remote Site Telephony Version 4.0 System Administrator Guide February 2006 Corporate Headquarters Cisco Systems, Inc. 170 West Tasman Drive San Jose, CA 95134-1706 USA http://www.cisco.
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C O N T E N T S Cisco Unified Survivable Remote Site Telephony Feature Roadmap Contents 1 1 Documentation Organization 1 Feature Roadmap 3 Information About New Features in Cisco Unified SRST V4.0 7 Information About New Features in Cisco SRST V3.4 9 Information About New Features in Cisco SRST V3.3 9 Information About New Features in Cisco SRST V3.2 10 Information About New Features in Cisco SRST V3.1 13 Information About New Features in Cisco SRST V3.
Contents Related Documents 35 Standards 37 MIBs 37 RFCs 37 Technical Assistance 37 Setting Up the Network Contents 39 39 Information About Setting Up the Network 39 How to Set Up the Network 40 Enabling IP Routing 40 Enabling SRST on an MGCP Gateway 40 Configuring DHCP for Cisco Unified SRST Phones 42 Specifying Keepalive Intervals 45 Configuring Cisco Unified SRST to Support Phone Functions Verifying That Cisco Unified SRST Is Enabled 48 Where to Go Next 49 Setting Up Cisco Unified IP Phones Conten
Contents H.323 VoIP Call Preservation Enhancements for WAN Link Failures Where to Go Next 97 Configuring Additional Call Features Contents 97 99 99 Information About Configuring Additional Call Features 99 How to Configure Additional Call Features 99 Enabling Three-Party G.711 Ad Hoc Conferencing 100 Configuring MOH for G.
Contents RFCs 145 Technical Assistance 145 Integrating Voice Mail with Cisco Unified SRST Contents 147 147 Information About Integrating Voice Mail with Cisco Unified SRST 147 How to Integrate Voice Mail with Cisco Unified SRST 149 Configuring Direct Access to Voice Mail 149 Configuring Message Buttons 152 Redirecting to Cisco Unified CallManager Gateway 154 Configuring Call Forwarding to Voice Mail 154 Configuring Message Waiting Indication 159 Configuration Examples 161 Configuring Local Voice-Mai
Contents Monitoring and Maintaining Cisco Unified SRST 179 Appendix A: Preparing Cisco Unified SRST Support for SIP Contents 181 181 DTMF Relay for SIP Applications and Voice Mail DTMF Relay Using SIP RFC 2833 182 DTMF Relay Using SIP Notify (Nonstandard) 181 183 INDEX Cisco IOS Survivable Remote Site Telephony Version 4.
Contents Cisco IOS Survivable Remote Site Telephony Version 4.
Cisco Unified Survivable Remote Site Telephony Feature Roadmap This chapter contains a list of Cisco Unified Survivable Remote Site Telephony (Cisco Unified SRST) features and the location of feature documentation. Note Prior to version 4.0, the name of this product was Cisco SRST. Use Cisco Feature Navigator to find information about platform support and Cisco IOS software image support. Access Cisco Feature Navigator at http://www.cisco.com/go/fn. You must have an account on Cisco.com.
Cisco Unified Survivable Remote Site Telephony Feature Roadmap Documentation Organization Table 1 Cisco Unified SRST Configuration Sequence Chapter or Appendix Description Overview of Cisco Unified SRST Provides a summary of SRST.
Cisco Unified Survivable Remote Site Telephony Feature Roadmap Feature Roadmap Table 1 Cisco Unified SRST Configuration Sequence (continued) Chapter or Appendix Description Setting Up Secure Survivable Remote Site Telephony Describes the Media and Signaling Authentication and Encryption feature for Cisco IOS MGCP gateways in SRST mode.
Cisco Unified Survivable Remote Site Telephony Feature Roadmap Feature Roadmap Table 2 Cisco Unified SRST Features by Cisco IOS Release (continued) Cisco Unified SRST Version Cisco IOS Release Version 3.3 Version 3.2 Version 3.1 Version 3.0 12.3(14)T 12.3(11)T 12.3(7)T 12.3(4)T 12.2(15)ZJ Modifications • Secure SRST, page 10.
Cisco Unified Survivable Remote Site Telephony Feature Roadmap Feature Roadmap Table 2 Cisco Unified SRST Features by Cisco IOS Release (continued) Cisco Unified SRST Version Cisco IOS Release Version 2.1 12.2(15)T1 12.2(15)T 12.2(11)YT Version 2.02 Version 2.01 12.2(13)T 12.
Cisco Unified Survivable Remote Site Telephony Feature Roadmap Feature Roadmap Table 2 Cisco Unified SRST Features by Cisco IOS Release (continued) Cisco Unified SRST Version Cisco IOS Release Version 2.0 Modifications 12.2(8)T1 Cisco Unified SRST was implemented on the Cisco 2600XM and Cisco 2691 routers. 12.2(8)T Cisco Unified SRST was integrated into Cisco IOS Release 12.2(8)T and implemented on the Cisco 3725 and Cisco 3745 routers and the Cisco MC3810-V3 concentrators. 12.
Cisco Unified Survivable Remote Site Telephony Feature Roadmap Feature Roadmap Table 2 Cisco Unified SRST Features by Cisco IOS Release (continued) Cisco Unified SRST Version Cisco IOS Release Version 1.0 12.1(5)YD1 12.1(5)YD Modifications Support was added for 144 Cisco IP phones on the Cisco 3660 multiservice routers. • Cisco Unified SRST introduced on the Cisco 2600 series and Cisco 3600 series multiservice routers and the Cisco IAD2420 series integrated access devices.
Cisco Unified Survivable Remote Site Telephony Feature Roadmap Feature Roadmap Additional Cisco Unified IP Phone Support The following IP phones are supported with Cisco Unified SRST systems: • Cisco Unified IP Phone 7911G • Cisco Unified IP Phone 7941G and Cisco Unified IP Phone 7941G-GE • Cisco Unified IP Phone 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.
Cisco Unified Survivable Remote Site Telephony Feature Roadmap Feature Roadmap Call preservation is useful when a gateway and the other endpoint (typically a Cisco Unified IP phone) are collocated at the same site and the call agent is remote and therefore more likely to experience connectivity failures. For configuration information see the “Configuring H.323 Gateways” chapter in the Cisco IOS H.323 Configuration Guide, Release 12.4T at http://www.cisco.
Cisco Unified Survivable Remote Site Telephony Feature Roadmap Feature Roadmap 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 CallManager using the WAN. But if the WAN link or Cisco Unified CallManager goes down, all communication through the remote phones becomes nonsecure.
Cisco Unified Survivable Remote Site Telephony Feature Roadmap Feature Roadmap • Enhancement to the cor Command • Enhancement to the pickup Command • Enhancement to the user-locale Command • Increased the Number of Cisco Unified IP Phones Supported on the Cisco 3845 • MOH Live-Feed Support • No Timeout for Call Preservation • RFC 2833 DTMF Relay Support • Translation Profile Support Enhancement to the alias Command The alias command has been enhanced as follows: • The cfw keyword was added
Cisco Unified Survivable Remote Site Telephony Feature Roadmap Feature Roadmap Increased the Number of Cisco Unified IP Phones Supported on the Cisco 3845 The Cisco 3845 now supports 720 phones and up to 960 ephone-dns or virtual voice ports. For more information, see Cisco IOS Survivable Remote Site Telephony (SRST) 3.2 Specifications for Cisco IOS Software Release 12.3(11)T. MOH Live-Feed Support Cisco SRST has been enhanced with the new moh-live command.
Cisco Unified Survivable Remote Site Telephony Feature Roadmap Feature Roadmap Information About New Features in Cisco SRST V3.1 Cisco SRST V3.1 introduced the new features described in the following sections: • Cisco Unified IP Phone 7920 Support • Cisco Unified IP Phone 7936 Support Cisco Unified IP Phone 7920 Support The Cisco Unified Wireless IP Phone 7920 is an easy-to-use IEEE 802.
Cisco Unified Survivable Remote Site Telephony Feature Roadmap Feature Roadmap • System Log Messages for Phone Registrations • Three-Party G.711 Ad Hoc Conferencing • Support for Cisco VG248 Analog Phone Gateway Version 1.2(1) and Higher Additional Language Options for IP Phone Display Displays for the Cisco Unified Unified IP Phone 7940G and Cisco Unified Unified IP Phone 7960G can be configured with additional ISO-3166 codes for Denmark, The Netherlands, Norway, and Sweden.
Cisco Unified Survivable Remote Site Telephony Feature Roadmap Feature Roadmap Dual-Line Mode A new keyword that has been added to the max-dn command allows you to set IP phones to dual-line mode. Each dual-line IP phone must have one voice port and two channels to handle two independent calls. This mode enables call waiting, call transfer, and conference functions on a single ephone-dn (ephone directory number). There is a maximum number of DNs available during Cisco SRST fallback.
Cisco Unified Survivable Remote Site Telephony Feature Roadmap Feature Roadmap • New Zealand • Paraguay • Peru • Philippines • Saudi Arabia • Singapore • South Africa (Panaftel variant) • Telmex corporation (Mexico) • Telnor corporation (Mexico) • Thailand • Uruguay • Venezuela • Vietnam 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) For config
Cisco Unified Survivable Remote Site Telephony Feature Roadmap Feature Roadmap Secondary Dial Tone A secondary dial tone is available for Cisco Unified IP Phones running Cisco SRST. The secondary dial tone is generated when a user dials a predefined PSTN access prefix. An example would be the different dial tone heard when a designated number is pressed to reach an outside line. The secondary dial tone is created through the secondary dialtone command.
Cisco Unified Survivable Remote Site Telephony Feature Roadmap Feature Roadmap Information About Features That Were New in Cisco SRST V2.1 Cisco SRST V2.
Cisco Unified Survivable Remote Site Telephony Feature Roadmap Feature Roadmap Cisco Unified IP Phone 7902G Support The Cisco Unified IP Phone 7902G is an entry-level IP phone that addresses the voice communications needs of a lobby, laboratory, manufacturing floor, hallway, or other area where only basic calling capability is required.
Cisco Unified Survivable Remote Site Telephony Feature Roadmap Feature Roadmap Enhancement to the dialplan-pattern Command A new keyword has been added to the dialplan-pattern command. The extension-pattern keyword sets an extension number’s leading digit pattern when it is different from the E.164 telephone number’s leading digits defined in the pattern variable. This enhancement allows manipulation of IP phone abbreviated extension number prefix digits.
Cisco Unified Survivable Remote Site Telephony Feature Roadmap Feature Roadmap Table 3 Increases in Directory Numbers in Cisco IOS Release 12.2(11)T (continued) Increase in Maximum Directory Number Cisco Platform Maximum Cisco IP Phones From To Cisco 3725 routers 144 432 576 Cisco 3745 routers 240 720 960 Unity Voice Mail Integration Using In-Band DTMF Signaling Across the PSTN and BRI/PRI Unity Voice Mail and other voice-mail systems can be integrated with Cisco SRST.
Cisco Unified Survivable Remote Site Telephony Feature Roadmap Feature Roadmap Cisco Unified Survivable Remote Site Telephony Version 4.
Overview of Cisco Unified SRST This chapter describes Cisco Unified Survivable Remote Site Telephony (Cisco Unified SRST) and what it does. It also includes information about Cisco Unified IP Phone, platform, and Cisco Unified CallManager version support; specifications; features; restrictions; and where to find additional reference documents. Note Prior to version 4.0, the name of this product was Cisco SRST.
Overview of Cisco Unified SRST Cisco Unified SRST Description Cisco Unified CallManager supports Cisco IP phones at remote sites attached to Cisco multiservice routers across the WAN. Prior to Cisco Unified SRST, when the WAN connection between a router and the Cisco Unified CallManager failed or when connectivity with Cisco Unified CallManager was lost for some reason, Cisco IP phones on the network became unusable for the duration of the failure.
Overview of Cisco Unified SRST Cisco Unified SRST Description Cisco Unified CallManager fallback mode. When the Cisco Unified CallManager is restored, the message goes away and full Cisco IP phone functionality is restored. While in Cisco Unified CallManager fallback mode, Cisco IP phones periodically attempt to reestablish a connection with Cisco Unified CallManager at the central office.
Overview of Cisco Unified SRST Cisco Unified SRST Description Figure 2 Branch Office Cisco IP Phones Operating in SRST Mode Telephone Telephone Fax Central Cisco CallManager PSTN V Cisco Unified SRST router IP IP Cisco IP phones WAN disconnected PCs 146613 IP IP network H.323 Gateways and SRST On H.323 gateways, when the WAN link fails, active calls from Cisco Unified IP Phones to the PSTN are not maintained by default.
Overview of Cisco Unified SRST Support for Cisco Unified IP Phones, Platforms, Cisco Unified CallManager, Signals, Languages, and Switches Note The commands listed above are ineffective unless both commands are configured. For instance, your configuration will not work if you only configure the ccm-manager fallback-mgcp command.
Overview of Cisco Unified SRST Support for Cisco Unified IP Phones, Platforms, Cisco Unified CallManager, Signals, Languages, and Switches Cisco Unified IP Phone Support For the most up-to-date information about Cisco Unified IP Phone support, see the Cisco Unified SRST 4.0 Supported Firmware, Platforms, Memory, and Voice Products at http://www.cisco.com/en/US/customer/products/sw/voicesw/ps2169/prod_installation_guide09186a00 805f6f1b.html. The following IP phones are supported by Cisco Unified SRST 4.
Overview of Cisco Unified SRST Support for Cisco Unified IP Phones, Platforms, Cisco Unified CallManager, Signals, Languages, and Switches Note During Cisco Unified CallManager fallback, Cisco Unified SRST considers the Cisco VG248 to be a group of Cisco Unified IP Phones. Cisco Unified SRST counts each of the 48 ports on the Cisco VG248 as a separate Cisco IP phone. Support for Cisco VG248 Version 1.2(1) and higher is available as of Cisco SRST Version 2.1.
Overview of Cisco Unified SRST Support for Cisco Unified IP Phones, Platforms, Cisco Unified CallManager, Signals, Languages, and Switches Language Support Cisco SRST 3.2 and later supports the following languages: Note • Danish • Dutch • English • French • German • Italian • Japanese Katakana (available under Cisco Unified CallManager 4.0 or later).
Overview of Cisco Unified SRST Prerequisites for Configuring Cisco Unified SRST Prerequisites for Configuring Cisco Unified SRST Before configuring Cisco Unified SRST you must do the following: • You have an account on Cisco.com to download software. To obtain an account on Cisco.com, go to www.cisco.com and click Register at the top of the screen. • You have purchased a Cisco Unified SRST license. To purchase a license, go to http://www.cisco.com/cgi-bin/tablebuild.pl/ip-key.
Overview of Cisco Unified SRST Prerequisites for Configuring Cisco Unified SRST Cisco SRST and Cisco Unified SRST can be configured to support continuous multicast output of music on hold (MOH) from a flash MOH file in flash memory. For more information, see the “Configuring MOH from Flash Files” section on page 102. If you plan use music on hold, go to the Technical Support Software Download site at http://www.cisco.com/cgi-bin/tablebuild.pl/ip-iostsp and copy the music-on-hold.
Overview of Cisco Unified SRST Restrictions for Configuring Cisco Unified SRST If You Have Cisco CallManager Prior to V3.3 If you have firmware versions that enable Cisco Unified SRST by default, no additional configuration is required on Cisco CallManager to support Cisco Unified SRST. If your firmware versions disable Cisco Unified SRST by default, you must enable Cisco Unified SRST for each phone configuration. Step 1 Go to the Cisco CallManager Phone Configuration page. a.
Overview of Cisco Unified SRST Restrictions for Configuring Cisco Unified SRST Table 4 History of Restrictions from Cisco SRST V1.0 to the Present Cisco Unified SRST Version Cisco SRST Version Cisco IOS Release Version 4.0 12.4(4)XC • All of the restrictions in Cisco SRST Version 1.0. Version 3.4 12.4(4)T • Call transfer is supported only on the following: Version 3.3 12.3(14)T Version 3.2 12.3(11)T Version 3.1 12.3(7)T – FXO and FXS loop-start (analog) Version 3.0 12.
Overview of Cisco Unified SRST Where to Go Next Where to Go Next The next chapters of this guide describe how to configure Cisco Unified SRST. As shown in Table 5, each chapter takes you through these 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 is configured correctly for Cisco Unified SRST.
Overview of Cisco Unified SRST Additional References Related Topic Documents Configuring SRST and MGCP Fallback • Configuring MGCP Gateway Support for Cisco Unified CallManager • MGCP Gateway Fallback Transition to Default H.
Overview of Cisco Unified SRST Additional References Standards Standard Title No new or modified standards are supported by this — feature, and support for existing standards has not been modified by this feature. MIBs MIB MIBs Link No new or modified MIBs are supported by this feature, and support for existing MIBs has not been modified by this feature.
Overview of Cisco Unified SRST Additional References Cisco Unified Survivable Remote Site Telephony Version 4.
Setting Up the Network 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 CallManager fallback. Note Prior to version 4.0, the name of this product was Cisco SRST. Note The Cisco IOS Voice Configuration Library includes a standard library preface, glossary, and feature and troubleshooting documents and is located at http://www.cisco.
Setting Up the Network How to Set Up the Network How to Set Up the Network This section contains the following tasks: • Enabling IP Routing, page 40 (Required) • Enabling SRST on an MGCP Gateway (Required) • Configuring DHCP for Cisco Unified SRST Phones, page 42 (Required) • Specifying Keepalive Intervals, page 45 (Optional) • Configuring Cisco Unified SRST to Support Phone Functions, page 46 (Required) • Verifying That Cisco Unified SRST Is Enabled, page 48 (Optional) Enabling IP Routing For
Setting Up the Network How to Set Up the Network DETAILED STEPS Step 1 Command or Action Purpose enable Enables privileged EXEC mode. • Enter your password when prompted. Example: Router> enable Step 2 Enters global configuration mode.
Setting Up the Network How to Set Up the Network Configuring DHCP for Cisco Unified SRST Phones To perform this task, you must have your network configured with DHCP. For further details about DHCP configuration, see the Cisco IOS DHCP Server document and refer to your Cisco Unified CallManager documentation. When a Cisco IP phone is connected to the Cisco Unified SRST system, it automatically queries for a DHCP server.
Setting Up the Network How to Set Up the Network DETAILED STEPS Step 6 Command or Action Purpose ip dhcp pool pool-name Creates a name for the DHCP server address pool and enters DHCP pool configuration mode. Example: Router(config)# ip dhcp pool mypool Step 7 network ip-address [mask | prefix-length] Example: Specifies the IP address of the DHCP address pool and the optional mask or number of bits in the address prefix, preceded by a forward slash. Router(config-dhcp)# network 10.0.0.0 255.255.
Setting Up the Network How to Set Up the Network DETAILED STEPS Step 1 Command or Action Purpose ip dhcp pool pool-name Creates a name for the DHCP server address pool and enters DHCP pool configuration mode. Example: Router(config)# ip dhcp pool pool2 Step 2 host ip-address subnet-mask Specifies the IP address that you want the phone to use. Example: Router(config-dhcp)# host 10.0.0.0 255.255.0.0 Step 3 option 150 ip ip-address Example: Router(config-dhcp)# option 150 ip 10.0.22.
Setting Up the Network How to Set Up the Network DETAILED STEPS Step 1 Command or Action Purpose service dhcp Enables the Cisco IOS DHCP Server feature on the router. Example: Router(config)# service dhcp Step 2 interface type number Example: Router(config)# interface serial 0 Step 3 Router(config-if)# ip helper-address 10.0.22.1 Specifies the helper address for any unrecognized broadcast for TFTP server and Domain Name System (DNS) requests.
Setting Up the Network How to Set Up the Network DETAILED STEPS Step 1 Command or Action Purpose call-manager-fallback Enters call-manager-fallback configuration mode. Example: Router(config)# call-manager-fallback Step 2 keepalive seconds Sets the time interval, in seconds, between keepalive messages that are sent to the router by Cisco IP phones. • Example: seconds—Range is 10 to 65535. Default is 30.
Setting Up the Network How to Set Up the Network DETAILED STEPS Step 1 Command or Action Purpose call-manager-fallback Enters call-manager-fallback configuration mode. Example: Router(config)# call-manager-fallback Step 2 ip source-address ip-address [port port] [any-match | strict-match] Example: Enables the router to receive messages from the Cisco IP phones through the specified IP addresses and provides for strict IP address verification. The default port number is 2000.
Setting Up the Network How to Set Up the Network Step 4 Command or Action Purpose max-ephones max-phones Configures the maximum number of Cisco IP phones that can be supported by the router. The default is 0. The maximum number is platform dependent. See the “Platform and Memory Support” section on page 29 for further details.
Setting Up the Network Where to Go Next b. Enter the no form of the appropriate access-list command to restore normal service for the phone. c. Use the debug ephone register command to observe the registration process of the Cisco IP phone on the Cisco Unified SRST router. d. Use the show ephone command to display the Cisco IP phones that have registered to the Cisco Unified SRST router.
Setting Up the Network Where to Go Next Cisco Unified Survivable Remote Site Telephony Version 4.
Setting Up Cisco Unified IP Phones 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 CallManager fallback. Note Prior to Cisco Unified Survivable Remote Site Telephony (SRST) 4.0, the name of this product was Cisco SRST. Note The Cisco IOS Voice Configuration Library includes a standard library preface, glossary, and feature and troubleshooting documents and is located at http://www.cisco.
Setting Up Cisco Unified IP Phones How to Set Up Cisco Unified IP Phones How to Set Up Cisco Unified IP Phones This section contains the following tasks: • Configuring IP Phone Clock, Date, and Time Formats, page 52 (Optional) • Configuring IP Phone Language Display, page 53 (Optional) • Configuring Customized System Messages for Cisco Unified IP Phones, page 55 (Optional) • Configuring a Secondary Dial Tone, page 57 (Optional) • Configuring Dual-Line Phones, page 58 (Required Under Certain Condit
Setting Up Cisco Unified IP Phones How to Set Up Cisco Unified IP Phones DETAILED STEPS Step 1 Command or Action Purpose clock timezone zone hours-offset [minutes-offset] Sets the time zone for display purposes. • zone—Name of the time zone to be displayed when standard time is in effect. The length of the zone argument is limited to 7 characters. • hours-offset—The number of hour difference from Coordinated Universal Time (UTC). • minutes-offset—(Optional) Minutes difference from UTC.
Setting Up Cisco Unified IP Phones How to Set Up Cisco Unified IP Phones Note This configuration option is available in Cisco SRST V2.1 and later running under Cisco CallManager V3.2 and later. Systems with software prior to Cisco SRST V2.1 and Cisco CallManager V3.2 can use the default country, United States (US), only. SUMMARY STEPS 1. call-manager-fallback 2. user-locale country-code 3.
Setting Up Cisco Unified IP Phones How to Set Up Cisco Unified IP Phones Examples The following example offers a configuration for the Portugal user locale. call-manager-fallback user-locale PT Configuring Customized System Messages for Cisco Unified IP Phones The system message command is used to customize the system message displayed on all Cisco UnifiedIP Phone 7910, Cisco Unified IP Phone 7940G, and Cisco Unified IP Phone 7960G units during Cisco Unified CallManager fallback.
Setting Up Cisco Unified IP Phones How to Set Up Cisco Unified IP Phones DETAILED STEPS Step 1 Command or Action Purpose call-manager-fallback Enters call-manager-fallback configuration mode. Example: Router(config)# call-manager-fallback Step 2 system message {primary primary-string | secondary secondary-string} Declares the text for the system display message on IP phones in fallback mode.
Setting Up Cisco Unified IP Phones 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 exit Configuring a Secondary Dial Tone A secondary dial tone can be generated when a phone user dials a predefined PSTN access prefix and can be terminated when additional digits are dialed.
Setting Up Cisco Unified IP Phones How to Set Up Cisco Unified IP Phones Configuring Dual-Line Phones Dual-line phone configuration is required for dual-line phone operation during Cisco Unified CallManager fallback. Consultative transfer is also required (see the “Enabling Consultative Call Transfer and Forward Using H.450.2 and H.450.3 with Cisco SRST 3.0” section on page 82). Dual-line IP phones are supported during Cisco Unified CallManager fallback using the max-dn command.
Setting Up Cisco Unified IP Phones How to Set Up Cisco Unified IP Phones DETAILED STEPS Step 1 Command or Action Purpose call-manager-fallback Enters call-manager-fallback configuration mode. Example: Router(config)# call-manager-fallback Step 2 max-dn max-directory-numbers [dual-line] [preference preference-order] Example: Sets the maximum number of directory numbers (DNs) or virtual voice ports that can be supported by the router and activates the dual-line mode.
Setting Up Cisco Unified IP Phones How to Set Up Cisco IP Communicator for Cisco Unified SRST How to Set Up Cisco IP Communicator for Cisco Unified SRST Cisco IP Communicator is a software-based application that delivers enhanced telephony support on personal computers. Cisco IP Communicator appears on a user’s computer monitor as a graphical, display-based IP phone with a color screen, a key pad, feature buttons, and soft keys.
Setting Up Cisco Unified IP Phones Where to Go Next Verifying Cisco IP Communicator Step 1 Use the show running-config command to display ephone-dn and ephone information associated with this phone. Step 2 After Cisco IP Communicator registers with Cisco Unified CME, it displays the phone extensions and soft keys in its configuration. Verify that these are correct. Step 3 Make a local call from the phone and ask someone to call you. Verify that you have a two-way voice path.
Setting Up Cisco Unified IP Phones Where to Go Next Cisco Unified Survivable Remote Site Telephony Version 4.
Setting Up Call Handling This chapter describes how to configure Cisco Unified Survivable Remote Site Telephony (SRST) for incoming calls and outgoing calls. Note Prior to version 4.0, the name of this product was Cisco SRST. Note The Cisco IOS Voice Configuration Library includes a standard library preface, glossary, and feature and troubleshooting documents and is located at http://www.cisco.com/en/US/products/ps6441/prod_configuration_guide09186a0080565f8a.html.
Setting Up Call Handling How to Set Up Call Handling for Incoming and Outgoing Calls Configuring Incoming Calls Incoming call configuration can include the following tasks: • Call Forwarding and Rerouting – Configuring Call Forwarding During a Busy Signal or No Answer, page 64 (Optional) – Configuring Call Rerouting, page 66 (Optional) – Configuring Call Pickup, page 69 (Optional) • Phone Number Conversion and Translation – Configuring Global Prefixes, page 71 (Optional) – Enabling Digit Translation Rul
Setting Up Call Handling How to Set Up Call Handling for Incoming and Outgoing Calls DETAILED STEPS Step 1 Command or Action Purpose call-manager-fallback Enters call-manager-fallback configuration mode. Example: Router(config)# call-manager-fallback Step 2 call-forward busy directory-number Configures call forwarding to another number when the Cisco IP phone is busy. • Example: Router(config-cm-fallback)# call-forward busy 50..
Setting Up Call Handling How to Set Up Call Handling for Incoming and Outgoing Calls Configuring Call Rerouting Note The alias command obsoletes the default-destination command and is recommended over the default-destination command. The alias command provides a mechanism for rerouting calls to telephone numbers that are unavailable during fallback. Up to 50 sets of rerouting alias rules can be created for calls to telephone numbers that are unavailable during Cisco Unified CallManager fallback.
Setting Up Call Handling How to Set Up Call Handling for Incoming and Outgoing Calls In this example, you have created a second dial peer for 1001 to route calls to 1001, but that has preference 1 and call forwarding to 2001. Because the preference on the dial peer created by the alias command is now a lower numeric value than the preference that the dial peer first created, all calls come initially to the dial peer created by the alias command.
Setting Up Call Handling How to Set Up Call Handling for Incoming and Outgoing Calls DETAILED STEPS Step 1 Command or Action Purpose call-manager-fallback Enters call-manager-fallback configuration mode. Example: Router(config)# call-manager-fallback Step 2 alias tag number-pattern to alternate-number [preference preference-value] [cfw number timeout timeout-value] [huntstop] Example: Router(config-cm-fallback)# alias 1 60..
Setting Up Call Handling How to Set Up Call Handling for Incoming and Outgoing Calls Step 4 Command or Action Purpose end Returns to privileged EXEC mode. Example: Router(config-cm-fallback)# end Step 5 show dial-peer voice summary Displays information for voice dial peers. • Example: Router# show dial-peer voice summary If you suspect a problem with the dial peers, use this command to display the dial peers created by the alias command.
Setting Up Call Handling How to Set Up Call Handling for Incoming and Outgoing Calls DETAILED STEPS Step 1 Command or Action Purpose call-manager-fallback Enters call-manager-fallback configuration mode. Example: Router(config)# call-manager-fallback Step 2 Disables huntstop.
Setting Up Call Handling How to Set Up Call Handling for Incoming and Outgoing Calls pickup 8005550100 When a DID incoming call to 800 555-0100 is received, the alias command routes the call at random to one of the four extensions (5001 to 5004). Because the pickup command is configured, if the DID call rings on extension 5002, the call can be answered from any of the other extensions (5001, 5003, 5004) by pressing the PickUp soft key.
Setting Up Call Handling How to Set Up Call Handling for Incoming and Outgoing Calls DETAILED STEPS Step 1 Command or Action Purpose call-manager-fallback Enters call-manager-fallback configuration mode. Example: Router(config)# call-manager-fallback Step 2 dialplan-pattern tag pattern extension-length length [extension-pattern extension-pattern] [no-reg] Example: Router(config-cm-fallback)# dialplan-pattern 1 4085550100 extension-length 3 extension-pattern 4..
Setting Up Call Handling How to Set Up Call Handling for Incoming and Outgoing Calls In the following example, the leading prefix digit for the 3-digit extension numbers is transformed from 0 to 4, so that the extension-number range becomes 400 to 499. call-manager-fallback dialplan-pattern 1 40855500.. extension-length 3 extension-pattern 4.. In the following example, the dialplan-pattern command creates dial-plan pattern 2 for extensions 801 to 899 with the telephone prefix starting with 4085559.
Setting Up Call Handling How to Set Up Call Handling for Incoming and Outgoing Calls DETAILED STEPS Step 1 Command or Action Purpose call-manager-fallback Enters call-manager-fallback configuration mode. Example: Router(config)# call-manager-fallback Step 2 translate {called | calling} translation-rule-tag Example: Applies a translation rule to modify the phone number dialed or received by any Cisco Unified IP Phone user while CallManager fallback is active.
Setting Up Call Handling How to Set Up Call Handling for Incoming and Outgoing Calls In the configuration below, the voice translation-rule and the rule command allow you to set and define how a number is to be manipulated. The translate command in voice translation-profile mode defines the type of number you are going to manipulate; such as a called, calling, or a redirecting number.
Setting Up Call Handling How to Set Up Call Handling for Incoming and Outgoing Calls DETAILED STEPS Step 1 Command or Action Purpose voice translation-rule number Defines a translation rule for voice calls and enters voice translation-rule configuration mode. Example: • Router(config)# voice translation-rule 1 Step 2 rule precedence/match-pattern/ /replace-pattern/ Example: Router(cfg-translation-rule)# rule 1/^9/ // Step 3 number—Number that identifies the translation rule.
Setting Up Call Handling How to Set Up Call Handling for Incoming and Outgoing Calls Step 8 Command or Action Purpose translation-profile {incoming | outgoing} name Assigns a translation profile for incoming or outgoing call legs on a Cisco IP phone. Example: • incoming—Applies the translation profile to incoming calls. • outgoing—Applies the translation profile to outgoing calls. • name—The name of the translation profile.
Setting Up Call Handling How to Set Up Call Handling for Incoming and Outgoing Calls Use this command to test your translation profiles. See the test voice translation-rule command in the Cisco IOS Voice Command Reference, Release 12.3 T for more information.
Setting Up Call Handling How to Set Up Call Handling for Incoming and Outgoing Calls DETAILED STEPS Step 1 Command or Action Purpose call-manager-fallback Enters call-manager-fallback configuration mode. Example: Router(config)# call-manager-fallback Step 2 Sets the huntstop attribute for the dial peers associated with the Cisco Unified IP Phone dial peers created during CallManager fallback.
Setting Up Call Handling How to Set Up Call Handling for Incoming and Outgoing Calls DETAILED STEPS Step 1 Command or Action Purpose call-manager-fallback Enters call-manager-fallback configuration mode. Example: Router(config)# call-manager-fallback Step 2 timeouts busy seconds Sets the amount of time after which calls are disconnected when they are transferred to busy destinations. • Example: Router(config-cm-fallback)# timeouts busy 20 Note Step 3 seconds—Number of seconds.
Setting Up Call Handling How to Set Up Call Handling for Incoming and Outgoing Calls DETAILED STEPS Step 1 Command or Action Purpose call-manager-fallback Enters call-manager-fallback configuration mode. Example: Router(config)# call-manager-fallback Step 2 Sets the ringing timeout default, in seconds. The range is from 5 to 60000. There is no default value. timeouts ringing seconds Example: Router(config-cm-fallback)# timeouts ringing 30 Step 3 Exits call-manager-fallback configuration mode.
Setting Up Call Handling How to Set Up Call Handling for Incoming and Outgoing Calls SUMMARY STEPS 1. call-manager-fallback 2. transfer-pattern transfer-pattern 3. exit DETAILED STEPS Step 1 Command or Action Purpose call-manager-fallback Enters call-manager-fallback configuration mode. Example: Router(config)# call-manager-fallback Step 2 transfer-pattern transfer-pattern Example: Router(config-cm-fallback)# transfer-pattern 52540..
Setting Up Call Handling How to Set Up Call Handling for Incoming and Outgoing Calls Prerequisites • Call transfer with consultation is available only when a second line or call instance is supported by the IP phone. Please see the dual-line keyword in the max-dn command. • All voice gateway routers in the VoIP network must support the H.450 standard. • All voice gateway routers in the VoIP network must be running the following software: – Cisco IOS Release 12.3(2)T or a later release – Cisco SRST 3.
Setting Up Call Handling How to Set Up Call Handling for Incoming and Outgoing Calls Step 3 Command or Action Purpose transfer-system {blind | full-blind | full-consult | local-consult} Defines the call-transfer method for all lines served by the Cisco Unified SRST router. • Example: Router(config-cm-fallback)# transfer-system full-consult Step 4 transfer-pattern transfer-pattern Example: Router(config-cm-fallback)# transfer-pattern 52540..
Setting Up Call Handling How to Set Up Call Handling for Incoming and Outgoing Calls Step 8 Command or Action Purpose h450 h450-2 timeout {T1 | T2 | T3 | T4} milliseconds (Optional) Sets timeouts for supplementary service timers, in milliseconds. This command is used primarily when the default settings for these timers do not match your network delay parameters. See the ITU-T H.450.2 specification for more information on these timers.
Setting Up Call Handling How to Set Up Call Handling for Incoming and Outgoing Calls Enabling Analog Transfer Using Hookflash and the H.450.2 Standard with Cisco SRST 3.0 or Earlier Analog call transfer using hookflash and the H.450.2 standard allows analog phones to transfer calls with consultation by using the hookflash to initiate the transfer.
Setting Up Call Handling How to Set Up Call Handling for Incoming and Outgoing Calls Restrictions • When a consultative transfer is made by an analog FXS phone using hookflash, the consultation call itself cannot be further transferred (that is, it cannot become a recursive or chained transfer) until after the initial transfer operation has been completed and the transferee and transfer-to parties are connected.
Setting Up Call Handling How to Set Up Call Handling for Incoming and Outgoing Calls Step 3 Command or Action Purpose call application voice application-name set-location language category location Defines the location and category of the audio files that are used by the application for dynamic prompts. Example: Router(config)# call application voice transfer_app set-location en 0 flash:/prompts • application-name—Name of the Tcl IVR application.
Setting Up Call Handling How to Set Up Call Handling for Incoming and Outgoing Calls Step 9 Command or Action Purpose application application-name Loads the application named in Step 1 onto the dial peer. Example: Router(config-dial-peer)# application transfer_app Step 10 Exits dial-peer configuration mode. exit Timesaver Example: Router(config-dial-peer)# exit Before exiting dial-peer configuration mode, configure any other dial-peer parameters that you need to set for this dial peer.
Setting Up Call Handling How to Set Up Call Handling for Incoming and Outgoing Calls 3. exit DETAILED STEPS Step 1 Command or Action Purpose call-manager-fallback Enters call-manager-fallback configuration mode.
Setting Up Call Handling How to Set Up Call Handling for Incoming and Outgoing Calls Note This value setting is important when using variable-length dial-peer destination patterns (dial plans). For more information on setting dial plans, see the “Configuration Dial Plans, Dial Peers, and Digit Manipulation” chapter of the Cisco IOS Voice, Video, and Fax Configuration Guide, Release 12.2. SUMMARY STEPS 1. call-manager-fallback 2. timeouts interdigit seconds 3.
Setting Up Call Handling How to Set Up Call Handling for Incoming and Outgoing Calls You can have up to 20 COR lists for each incoming and outgoing call. A default COR is assigned to directory numbers that do not match any COR list numbers or number ranges. An assigned COR is invoked for the dial peers and created for each directory number automatically during CallManager fallback registration.
Setting Up Call Handling How to Set Up Call Handling for Incoming and Outgoing Calls 2. cor {incoming | outgoing} cor-list-name {cor-list-number starting-number - ending-number | default} 3. exit DETAILED STEPS Step 1 Command or Action Purpose call-manager-fallback Enters call-manager-fallback configuration mode.
Setting Up Call Handling How to Set Up Call Handling for Incoming and Outgoing Calls dial-peer cor custom name 911 name 1800 name 1900 name local_call In the following configuration example, COR lists are created and applied to the dial peer.
Setting Up Call Handling How to Set Up Call Handling for Incoming and Outgoing Calls 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 The sample configuration allows for the following: • Extension 1001 to call 734... numbers, 911, and 316.... • Extension 1002 to call 734..., 1800 numbers, 911, and 316....
Setting Up Call Handling How to Set Up Call Handling for Incoming and Outgoing Calls DETAILED STEPS Step 1 Command or Action Purpose call-manager-fallback Enters call-manager-fallback configuration mode. Example: Router(config)# call-manager-fallback Step 2 after-hours block pattern tag pattern [7-24] Example: Defines a pattern of outgoing digits to be blocked. Up to 32 patterns can be defined, using individual commands.
Setting Up Call Handling H.323 VoIP Call Preservation Enhancements for WAN Link Failures Example The following example defines several patterns of digits for which outgoing calls are blocked. Patterns 1 and 2, which block calls to external numbers that begin with “1” and “011,” are blocked on Monday through Friday before 7 a.m. and after 7 p.m., on Saturday before 7 a.m. and after 1 p.m., and all day Sunday. Pattern 3 blocks calls to 900 numbers 7 days a week, 24 hours a day.
Setting Up Call Handling Where to Go Next Cisco Unified Survivable Remote Site Telephony Version 4.
Configuring Additional Call Features This chapter describe how to configure three-party G.711 ad hoc conferencing and music on hold (MOH) for Cisco Unified Survivable Remote Site Telephony (SRST). Note Prior to version 4.0, the name of this product was Cisco SRST. Note The Cisco IOS Voice Configuration Library includes a standard library preface, glossary, and feature and troubleshooting documents and is located at http://www.cisco.com/en/US/products/ps6441/prod_configuration_guide09186a0080565f8a.html.
Configuring Additional Call Features How to Configure Additional Call Features • Configuring MOH for G.711 VoIP and PSTN Calls, page 101 (Optional) • Configuring MOH from Flash Files, page 102 (Optional) • Defining XML API Schema (Optional) Enabling Three-Party G.711 Ad Hoc Conferencing 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.
Configuring Additional Call Features How to Configure Additional Call Features Examples The following example configures up to eight simultaneous three-way conferences on a router. call-manager-fallback max-conferences 8 Configuring MOH for G.711 VoIP and PSTN Calls MOH configuration works with G.711 VoIP and PSTN calls only. For all other calls, such as internal calls between Cisco Unified IP Phones, a tone is heard. The MOH file can be in .wav or .au file format.
Configuring Additional Call Features How to Configure Additional Call Features Example The following example enables the playing of an audio file called classical.au on G.711, on-net VoIP, and PSTN calls: call-manager-fallback moh classical.
Configuring Additional Call Features Where to Go Next DETAILED STEPS Step 1 Command or Action Purpose call-manager-fallback Enters call-manager-fallback configuration mode. Example: Router(config)# call-manager-fallback Step 2 Specifies the URL for an XML API schema to be used with this Cisco Unified SRST system. xmlschema schema-url • Example: Router(config-cm-fallback)# xmlschema http://server2.example.com/ schema/schema1.xsd Step 3 schema-url—Local or remote URL as defined in RFC 2396.
Configuring Additional Call Features Where to Go Next Cisco Unified Survivable Remote Site Telephony Version 4.
Setting Up Secure Survivable Remote Site Telephony This chapter describes new Secure Survivable Remote Site Telephony (SRST) security features such as authentication, integrity, and media encryption. Note Prior to Cisco Unified SRST 4.0, the name of this product was Cisco SRST.
Setting Up Secure Survivable Remote Site Telephony Restrictions for Setting Up Secure SRST • Gateway routers that run secure SRST must support voice- and security-enabled Cisco IOS images (a “k9” cryptographic software image). The following two images are supported: – Advanced IP Services. This image includes a number of advanced security features. – Advanced Enterprise Services. This image includes full Cisco IOS software.
Setting Up Secure Survivable Remote Site Telephony Information About Setting Up Secure SRST http://www.cisco.com/wwl/export/crypto/tool/ If you require further assistance, please contact us by sending e-mail to export@cisco.com. • When a Secure Real-Time Transport Protocol (SRTP) encrypted call is made between Cisco Unified 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.
Setting Up Secure Survivable Remote Site Telephony Information About Setting Up Secure SRST activates when the WAN link or Cisco Unified CallManager goes down. When the WAN link or Cisco Unified CallManager is restored, Cisco Unified CallManager resumes secure call-handling capabilities. Secure SRST provides new SRST security features such as authentication, integrity, and media encryption. Authentication provides assurance to one party that another party is whom it claims to be.
Setting Up Secure Survivable Remote Site Telephony Information About Setting Up Secure SRST SRST Routers and PKI The transfer of certificates between an SRST router and Cisco Unified CallManager is mandatory for secure SRST functionality. Public key infrastructure (PKI) commands are used to generate, import, and export the certificates for secure SRST. Table 7 shows the secure SRST supported Cisco Unified IP Phones and the appropriate certificate for each phone.
Setting Up Secure Survivable Remote Site Telephony Information About Setting Up Secure SRST Secure SRST Authentication and Encryption Figure 4 illustrates the process of secure SRST authentication and encryption, and Table 8 describes the process. Figure 4 Secure SRST Authentication and Encryption CAPF Cisco IOS router CA or third-party CA TFTP 4 Cisco Unified CallManager SRST cert 2 4 5 3 1 SRST cert 7940/7960 LSC SEPMACxxxx.cnf.
Setting Up Secure Survivable Remote Site Telephony Information About Setting Up Secure SRST Table 8 Overview of the Process of Secure SRST Authentication and Encryption (continued) Process Steps Description or Detail Cisco Unified CallManager provides the PEM format files that contain phone certificate information to the SRST router. Providing the PEM files to the SRST router is done manually; see SRST Routers and PKI, page 109 for more information. 5.
Setting Up Secure Survivable Remote Site Telephony Information About Setting Up Secure SRST Interworking of Credentials Server on SRST Router, Cisco Unified CallManager, and Cisco Unified IP Phone Cisco Unified CallManager/ client 1. Cisco Unified CallManager requests the SRST certificate from the credentials server. WAN Credentials server running on secure SRST router 155100 Figure 5 2. The credentials server responds with the certificate. 3.
Setting Up Secure Survivable Remote Site Telephony How to Configure Secure SRST How to Configure Secure SRST The following configuration sections ensure that the secure SRST router and the Cisco IP phones can request mutual authentication during the TLS handshake. The TLS handshake occurs when the phone registers with the SRST router, either before or after the WAN link fails.
Setting Up Secure Survivable Remote Site Telephony How to Configure Secure SRST DETAILED STEPS Step 1 Command or Action Purpose crypto pki server cs-label Enables the certificate server and enters certificate server configuration mode. Example: Note Router (config)# crypto pki server srstcaserver If you manually generated an RSA key pair, the cs-label argument must match the name of the key pair. For more information on the certificate server, see the Cisco IOS Certificate Server documentation.
Setting Up Secure Survivable Remote Site Telephony How to Configure Secure SRST Step 5 Command or Action Purpose grant auto Allows an automatic certificate to be issued to any requestor. • Example: Router (cs-server)# grant auto Step 6 This command is used only during enrollment and will be removed in the “Disabling Automatic Certificate Enrollment” section on page 118. Enables the Cisco IOS certificate server.
Setting Up Secure Survivable Remote Site Telephony How to Configure Secure SRST SUMMARY STEPS 1. crypto pki trustpoint name 2. enrollment url url 3. revocation-check method1 4. exit 5. crypto pki authenticate name 6. crypto pki enroll name DETAILED STEPS Step 1 Command or Action Purpose crypto pki trustpoint name Declares the CA that your router should use and enters ca-trustpoint configuration mode.
Setting Up Secure Survivable Remote Site Telephony How to Configure Secure SRST Step 5 Command or Action Purpose crypto pki authenticate name Authenticates the CA (by getting the certificate from the CA). • Example: Takes the name of the CA as the argument. Router(config)# crypto pki authenticate srstca Step 6 Obtains the SRST router certificate from the CA. crypto pki enroll name • Takes the name of the CA as the argument.
Setting Up Secure Survivable Remote Site Telephony How to Configure Secure SRST Disabling Automatic Certificate Enrollment The command grant auto allows certificates to be issued and was activated in the optional task documented in the “Configuring a Certificate Authority Server on a Cisco IOS Certificate Server” section on page 113. Note A security best practice is to disable the grant auto command so that certificates cannot be continually granted. SUMMARY STEPS 1. crypto pki server cs-label 2.
Setting Up Secure Survivable Remote Site Telephony How to Configure Secure SRST SUMMARY STEPS 1. show running-config 2. show crypto pki server DETAILED STEPS Step 1 show running-config Use the show running-config command to verify the creation of the CA server (01) and device (02) certificates. This example shows the enrolled certificates. Router# show running-config . . . ! SRST router device certificate.
Setting Up Secure Survivable Remote Site Telephony How to Configure Secure SRST Certificate Server srstcaserver: Status: enabled Server's configuration is locked (enter "shut" to unlock it) Issuer name: CN=srstcaserver CA cert fingerprint: AC9919F5 CAFE0560 92B3478A CFF5EC00 Granting mode is: auto Last certificate issued serial number: 0x2 CA certificate expiration timer: 13:46:57 PST Dec 1 2007 CRL NextUpdate timer: 14:54:57 PST Jan 19 2005 Current storage dir: nvram Database Level: Complete - all issued
Setting Up Secure Survivable Remote Site Telephony How to Configure Secure SRST DETAILED STEPS Step 1 Command or Action Purpose credentials Provides the SRST router certificate to Cisco Unified CallManager and enters credentials configuration mode. Example: Router(config)# credentials Step 2 ip source-address ip-address [port port] Example: Router(config-credentials)# ip source-address 10.1.1.
Setting Up Secure Survivable Remote Site Telephony How to Configure Secure SRST DETAILED STEPS Step 1 show credentials Use the show credentials command to display the credential settings on the SRST router that are supplied to Cisco Unified CallManager for use during secure SRST fallback. Router# show credentials Credentials IP: 10.1.1.22 Credentials PORT: 2445 Trustpoint: srstca Step 2 debug credentials Use the debug credentials command to set debugging on the credential settings of the SRST router.
Setting Up Secure Survivable Remote Site Telephony How to Configure Secure SRST Prerequisites You must have certificates available when the last configuration command (crypto pki authenticate), issues the following prompt: Enter the base 64 encoded CA certificate. End with a blank line or the word "quit" on a line by itself Cisco Unified CallManager 4.X.X and Earlier For Cisco Unified CallManager 4.X.
Setting Up Secure Survivable Remote Site Telephony How to Configure Secure SRST DETAILED STEPS Step 1 Command or Action Purpose crypto pki trustpoint name Declares the CA that your router should use and enters ca-trustpoint configuration mode. • Example: Router (config)# crypto pki trustpoint 7970 Step 2 revocation-check method1 Example: Router(ca-trustpoint)# revocation-check none Checks the revocation status of a certificate.
Setting Up Secure Survivable Remote Site Telephony How to Configure Secure SRST MRYwFAYDVQQKEw1DaXNjbyBTeXN0ZW1zMRQwEgYDVQQDEwtDQVAtUlRQLTAwMjAe Fw0wMzEwMTAyMDE4NDlaFw0yMzEwMTAyMDI3MzdaMC4xFjAUBgNVBAoTDUNpc2Nv IFN5c3RlbXMxFDASBgNVBAMTC0NBUC1SVFAtMDAyMIIBIDANBgkqhkiG9w0BAQEF AAOCAQ0AMIIBCAKCAQEAxCZlBK19w/2NZVVvpjCPrpW1cCY7V1q9lhzI85RZZdnQ 2M4CufgIzNa3zYxGJIAYeFfcRECnMB3f5A+x7xNiEuzE87UPvK+7S80uWCY0Uhtl AVVf5NQgZ3YDNoNXg5MmONb8lT86F55EZyVac0XGne77TSIbIdejrTgYQXGP2MJx Qhg+ZQlGFDRzbHfM84Duv2Msez+l+SqmqO80kIckq
Setting Up Secure Survivable Remote Site Telephony How to Configure Secure SRST MRYwFAYDVQQKEw1DaXNjbyBTeXN0ZW1zMRQwEgYDVQQDEwtDQVAtUlRQLTAwMTAe Fw0wMzAyMDYyMzI3MTNaFw0yMzAyMDYyMzM2MzRaMC4xFjAUBgNVBAoTDUNpc2Nv IFN5c3RlbXMxFDASBgNVBAMTC0NBUC1SVFAtMDAxMIIBIDANBgkqhkiG9w0BAQEF AAOCAQ0AMIIBCAKCAQEArFW77Rjem4cJ/7yPLVCauDohwZZ/3qf0sJaWlLeAzBlq Rj2lFlSij0ddkDtfEEo9VKmBOJsvx6xJlWJiuBwUMDhTRbsuJz+npkaGBXPOXJmN Vd54qlpc/hQDfWlbrIFkCcYhHws7vwnPsLuy1Kw2L2cP0UXxYghSsx8H4vGqdPFQ NnYy7aKJ43SvDFt4zn37n8jrvlRuz0x3mdbcBEdHb
Setting Up Secure Survivable Remote Site Telephony How to Configure Secure SRST Certificate request(s) ..... None Trustpoint srstcaserver: Issuing CA certificate configured: Subject Name: cn=srstcaserver Fingerprint MD5: 6AF5B084 79C93F2B 76CC8FE6 8781AF5E Fingerprint SHA1: 47D30503 38FF1524 711448B4 9763FAF6 3A8E7DCF State: Keys generated ............. Yes (General Purpose) Issuing CA authenticated ....... Yes Certificate request(s) .....
Setting Up Secure Survivable Remote Site Telephony How to Configure Secure SRST Enter the base 64 encoded CA certificate.
Setting Up Secure Survivable Remote Site Telephony How to Configure Secure SRST FR5umgIJFq0roIlgX9p7L6owEAYJKwYBBAGCNxUBBAMCAQAwDQYJKoZIhvcNAQEF BQADggEBAJ2dhISjQal8dwy3U8pORFBi71R803UXHOjgxkhLtv5MOhmBVrBW7hmW Yqpao2TB9k5UM8Z3/sUcuuVdJcr18JOagxEu5sv4dEX+5wW4q+ffy0vhN4TauYuX cB7w4ovXsNgOnbFp1iqRe6lJT37mjpXYgyc81WhJDtSd9i7rp77rMKSsH0T8lasz Bvt9YAretIpjsJyp8qS5UwGH0GikJ3+r/+n6yUA4iGe0OcaEb1fJU9u6ju7AQ7L4 CYNu/2bPPu8Xs1gYJQk0XuPL1hS27PKSb3TkL4Eq1ZKR4OCXPDJoBYVL0fdX4lId kxpUnwVwwEpxYB5DC2Ae/qPOgRnhCzU= quit Cer
Setting Up Secure Survivable Remote Site Telephony How to Configure Secure SRST b. Check the box asking if the SRST gateway is secure. c. Enter the certificate provider (credentials service) port number. Credentials service runs on default port 2445. Figure 6 SRST Reference Configuration Window Step 4 To add the new SRST reference, click Insert. The message “Status: Insert completed” displays. Step 5 To add more SRST references, repeat Steps 2 through 4.
Setting Up Secure Survivable Remote Site Telephony How to Configure Secure SRST DETAILED STEPS Step 1 In the menu bar in Cisco Unified CallManager, choose CCMAdmin > System > Device Pool. Step 2 Use one of the following methods to add a device pool: Step 3 • If a device pool already exists with settings that are similar to the one that you want to add, choose the existing device pool to display its settings, click Copy, and modify the settings as needed. Continue with Step 4.
Setting Up Secure Survivable Remote Site Telephony How to Configure Secure SRST Configuring CAPF on Cisco Unified CallManager The Certificate Authority Proxy Function (CAPF) process allows supported devices, such as Cisco Unified CallManager, 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 CallManager servers in the cluster.
Setting Up Secure Survivable Remote Site Telephony How to Configure Secure SRST Step 3 Command or Action Purpose transfer-system {blind | full-blind | full-consult | local-consult} Defines the call-transfer method for all lines served by the Cisco Unified Unified SRST router. • blind—Calls are transferred without consultation with a single phone line using the Cisco proprietary method. • full-blind—Calls are transferred without consultation using H.450.2 standard methods.
Setting Up Secure Survivable Remote Site Telephony How to Configure Secure SRST Examples The following example enables SRST mode on your router. Router(config)# call-manager-fallback Router(config-cm-fallback)# secondary-dialtone 9 Router(config-cm-fallback)# transfer-system full-consult Router(config-cm-fallback)# ip source-address 10.1.1.22 port 2000 Router(config-cm-fallback)# max-ephones 15 Router(config-cm-fallback)# max-dn 30 Router(config-cm-fallback)# transfer-pattern .....
Setting Up Secure Survivable Remote Site Telephony How to Configure Secure SRST Step 2 show ephone offhook Use this command to display Cisco IP phone status and quality for all phones that are off hook. In this example, authentication and encryption status is active with a TLS connection, and there is an active secure call. Router# show ephone offhook ephone-1 Mac:1000.1111.
Setting Up Secure Survivable Remote Site Telephony How to Configure Secure SRST 0x117F 2C25 0x85C6BE6C 0x1180 2C28 0x860ADFF0 0x1181 2C28 0x8618FBBC 0x1182 2C2B 0x860C3B1C 0x1183 2C2B 0x860590EC 0x1184 2C2E 0x8617A090 0x1185 2C2E 0x8606E234 0x1186 2C31 0x861A56E8 0x1187 2C31 0x86185318 18 active calls found Step 4 50/0/1.0 *2006 g711ulaw 20001/20006 50/0/13.0 2029 g711ulaw 20013/20034 50/0/34.0 *2029 g711ulaw 20034/20013 50/0/15.0 2036 g711ulaw 20015/20005 50/0/5.0 *2036 g711ulaw 20005/20015 50/0/32.
Setting Up Secure Survivable Remote Site Telephony How to Configure Secure SRST *Jan 11 18:33:16.039:ephone-2[2]:Call Info DN 2 line 1 ref 6 called 6001 calling 6000 origcalled 6001 calltype 2 *Jan 11 18:33:16.039:ephone-2[2]:Call Info for chan 1 *Jan 11 18:33:16.039:ephone-2[2]:Original Called Name 6001 *Jan 11 18:33:16.039:ephone-2[2]:6000 calling *Jan 11 18:33:16.039:ephone-2[2]:6001 *Jan 11 18:33:16.047:ephone-3[3]:SetCallState line 1 DN 4(4) chan 1 ref 7 TsRingIn *Jan 11 18:33:16.
Setting Up Secure Survivable Remote Site Telephony Configuration Examples for Secure SRST *Jan 11 18:33:21.095:ephone-2[2]:OpenReceiveChannelAck:IP 1.1.1.8, port=25552, dn_index=2, dn=2, chan=1 *Jan 11 18:33:21.095:ephone-3[3]:StartMedia 1.1.1.8 port=25552 *Jan 11 18:33:21.095:DN 2 chan 1 codec 4:G711Ulaw64k duration 20 ms bytes 160 *Jan 11 18:33:21.095:ephone-3[3]:Send Encryption Key ! Ephone 3 sends its encryption key. *Jan 11 18:33:21.347:ephone-3[3]:OpenReceiveChannelAck:IP 1.1.1.
Setting Up Secure Survivable Remote Site Telephony Configuration Examples for Secure SRST database level complete database url nvram issuer-name CN=srstcaserver ! crypto pki trustpoint srstca enrollment url http://10.1.1.22:80 revocation-check none ! crypto pki trustpoint srstcaserver revocation-check none rsakeypair srstcaserver ! ! Define CTL/7970 trustpoint.
Setting Up Secure Survivable Remote Site Telephony Configuration Examples for Secure SRST crypto pki certificate chain srstcaserver certificate ca 01 30820207 30820170 A0030201 02020101 300D0609 2A864886 17311530 13060355 0403130C 73727374 63617365 72766572 31323139 34353136 5A170D30 37303431 32313934 3531365A 55040313 0C737273 74636173 65727665 7230819F 300D0609 01050003 818D0030 81890281 8100C3AF EE1E4BB1 9922A8DA 1051C9FE 32A971B3 3C336635 74691954 98E765B1 059E24B6 9619993F CC72C525 7357EBAC E6335A32 2
Setting Up Secure Survivable Remote Site Telephony Configuration Examples for Secure SRST 55DE78AA 5A5CFE14 037D695B AC816409 C6211F0B 3BBF09CF B0BBB2D4 AC362F67 0FD145F1 620852B3 1F07E2F1 AA74F150 367632ED A289E374 AF0C5B78 CE7DFB9F C8EBBE54 6ECF4C77 99D6DC04 47476C0F 36E58A3B 6BCB24D7 6B6C84C2 7F61D326 BE7CB4A6 60CD6579 9E1E3A84 8153B750 5527E865 423BE2B5 CB575453 5AA96093 58B6A2E4 AA3EF081 C7068EC1 DD1EBDDA 53E6F0D6 E2E0486B 109F1316 78C696A3 CFBA84CC 7094034F C1EB9F81 931ACB02 0103A381 C33081C0 300B060
Setting Up Secure Survivable Remote Site Telephony Configuration Examples for Secure SRST ip address 10.1.1.22 255.255.255.0 duplex auto speed auto crypto map rtp ! interface FastEthernet0/1 no ip address shutdown duplex auto speed auto ! ip classless ! ip http server no ip http secure-server ! ! ! Define traffic to be encrypted by IPSec. access-list 116 permit ip host 10.1.1.22 host 10.1.1.
Setting Up Secure Survivable Remote Site Telephony Configuration Examples for Secure SRST forward-digits all ! dial-peer voice 81234 pots application mgcpapp destination-pattern 81234 port 1/0/0 ! dial-peer voice 999100 pots application mgcpapp port 1/0/0 ! dial-peer voice 999110 pots application mgcpapp port 1/1/0 ! ! ! Enable credentials service on the gateway. credentials ip source-address 10.1.1.22 port 2445 trustpoint srstca ! ! ! Enable SRST mode.
Setting Up Secure Survivable Remote Site Telephony Where to Go Next ! Define aggregate control plane service for the active Route Processor. control-plane service-policy input control-plane-policy . . . Where to Go Next If you require voice mail, see the voice-mail configuration instructions in the “Integrating Voice Mail with Cisco Unified SRST” chapter. You may also want to read the “Monitoring and Maintaining Cisco Unified SRST” chapter.
Setting Up Secure Survivable Remote Site Telephony Additional References Related Topic Documents Command reference and configuration information for voice and telephony commands • Cisco IOS Voice Command Reference • Cisco IOS Debug Command Reference Go to http://www.cisco.com/en/US/products/sw/iosswrel/tsd_product s_support_category_home.html and click the appropriate Cisco IOS Software Release and Command References.
Setting Up Secure Survivable Remote Site Telephony Additional References Cisco Unified Survivable Remote Site Telephony Version 4.
Integrating Voice Mail with Cisco Unified SRST This chapter describes how to make your existing voice-mail system run on phones connected to a Cisco Unified (SRST) router during Cisco CallManager fallback. Note Prior to version 4.0, the name of this product was Cisco SRST. Note The Cisco IOS Voice Configuration Library includes a standard library preface, a glossary, and feature and troubleshooting documents and is located at http://www.cisco.
Integrating Voice Mail with Cisco Unified SRST Information About Integrating Voice Mail with Cisco Unified SRST Figure 8 IP Cisco Unified CallManager Fallback with BRI or PRI Cisco Unified SRST gateway Cisco Unified CallManager gateway BRI/PRI IP WAN failure Cisco Unified CallManager Voice-mail server WAN Figure 9 146615 IP Cisco Unified CallManager Fallback with PSTN Cisco Unified CallManager gateway IP FXS FXO PSTN IP IP WAN Voice-mail server 155102 Cisco Unified CallManager WAN fail
Integrating Voice Mail with Cisco Unified SRST How to Integrate Voice Mail with Cisco Unified SRST How to Integrate Voice Mail with Cisco Unified SRST This section contains the following tasks: • Configuring Direct Access to Voice Mail, page 149 (Required) • Configuring Message Buttons, page 152 (Required) • Redirecting to Cisco Unified CallManager Gateway, page 154 (Required for BRI or PRI)) • Configuring Call Forwarding to Voice Mail, page 154 (Required FXO or FXS) • Configuring Message Waiting
Integrating Voice Mail with Cisco Unified SRST How to Integrate Voice Mail with Cisco Unified SRST DETAILED STEPS Step 1 Command or Action Purpose dial-peer voice tag {pots | voatm | vofr | voip} (FXO or FXS and BRI or PRI) Defines a particular dial peer, specifies the method of voice encapsulation, and enters dial-peer configuration mode. The dial-peer command provides different syntax for individual routers. This example is syntax for Cisco 3600 series routers.
Integrating Voice Mail with Cisco Unified SRST How to Integrate Voice Mail with Cisco Unified SRST Step 4 Command or Action Purpose forward-digits {num-digit | all | extra} (Optional for FXO or FXS) Specifies which digits to forward for voice calls. Example: • num-digit—The number of digits to be forwarded. If the number of digits is greater than the length of a destination phone number, the length of the destination number is used. Range is 0 to 32.
Integrating Voice Mail with Cisco Unified SRST How to Integrate Voice Mail with Cisco Unified SRST Examples The following FXO and FXS example sets up a POTS dial peer named 1102, matches dial-peer 1102 to voice-mail extension 1101, and assigns dial-peer 1102 to voice-port 1/1/1 where the voice-mail system is connected. Other dial peers are configured for direct access to voice mail.
Integrating Voice Mail with Cisco Unified SRST How to Integrate Voice Mail with Cisco Unified SRST SUMMARY STEPS 1. call-manager-fallback 2. voicemail phone-number 3. call-forward busy directory-number 4. call-forward noan directory-number timeout seconds 5. exit DETAILED STEPS Step 1 Command or Action Purpose call-manager-fallback Enters call-manager-fallback configuration mode.
Integrating Voice Mail with Cisco Unified SRST How to Integrate Voice Mail with Cisco Unified SRST Examples 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).
Integrating Voice Mail with Cisco Unified SRST How to Integrate Voice Mail with Cisco Unified SRST Call Routing Instructions Using DTMF Digit Patterns Cisco Unified SRST call-routing instructions are required so that forwarded calls can be sent to the correct voice mailboxes. These instructions consist of DTMF digits configured in patterns that match the dial sequences required by the voice-mail system to get to a particular voice-mail location.
Integrating Voice Mail with Cisco Unified SRST How to Integrate Voice Mail with Cisco Unified SRST How Numbers Are Extracted from Tokens (cgn=calling number) IP (fdn=forwarding number) 1000 calls 2000 ext. 1000 IP (cdn=called number) Cisco CallManager 1000 is forwarded ext. 2000 ext.
Integrating Voice Mail with Cisco Unified SRST How to Integrate Voice Mail with Cisco Unified SRST 5. pattern trunk-to-ext busy tag1 {CGN | CDN | FDN} [tag2 {CGN | CDN | FDN}] [tag3 {CGN | CDN | FDN}] [last-tag] 6.
Integrating Voice Mail with Cisco Unified SRST How to Integrate Voice Mail with Cisco Unified SRST Step 5 Command or Action Purpose pattern trunk-to-ext busy tag1 {CGN | CDN | FDN} [tag2 {CGN | CDN | FDN}] [tag3 {CGN | CDN | FDN}] [last-tag] Configures the DTMF digit pattern forwarding necessary to activate the voice-mail system once an external trunk call reaches a busy extension and the call is forwarded to voice mail. For argument and keyword information, see Step 2.
Integrating Voice Mail with Cisco Unified SRST How to Integrate Voice Mail with Cisco Unified SRST Configuring Message Waiting Indication 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 SRST routers. SRST routers use the SIP Subscribe and Notify methods for MWI.
Integrating Voice Mail with Cisco Unified SRST How to Integrate Voice Mail with Cisco Unified SRST Step 6 Command Purpose mwi-server {ipv4:destination-address | dns:host-name} [expires seconds] [port port] [transport {tcp | udp}] [unsolicited] Configures voice-mail server settings on a voice gateway or user agent. The IP address and port for the SIP-based MWI server should be in the same LAN as the voice-mail server. The MWI server is a Cisco Unified SRST router.
Integrating Voice Mail with Cisco Unified SRST Configuration Examples Configuration Examples This section provides the following configuration examples: • Configuring Local Voice-Mail System (FXO and FXS): Example, page 161 • Configuring Central Location Voice-Mail System (FXO and FXS): Example, page 162 • Configuring Voice-Mail Access over FXO and FXS: Example, page 162 • Configuring Voice-Mail Access over BRI and PRI: Example, page 163 Configuring Local Voice-Mail System (FXO and FXS): Example Th
Integrating Voice Mail with Cisco Unified SRST Configuration Examples Configuring Central Location Voice-Mail System (FXO and FXS): Example The “Dial-Peer Configuration for Integration of Voice-Mail with Cisco Unified SRST in Central Location” section of the example shows a legacy dial-peer configuration for a central voice-mail system. The “Cisco Unified SRST Voice-Mail Integration Pattern Configuration” section must be compatible with your voice-mail system configuration.
Integrating Voice Mail with Cisco Unified SRST Where to Go Next call-forward busy 1101 call-forward noan 1101 timeout 3 moh minuet.au vm-integration pattern direct * CGN pattern ext-to-ext no-answer # FDN #2 pattern ext-to-ext busy # FDN #2 pattern trunk-to-ext no-answer # FDN #2 pattern trunk-to-ext busy # FDN #2 Configuring Voice-Mail Access over BRI and PRI: Example The following example shows how to configure the Cisco Unified SRST router to forward unanswered calls to voice mail.
Integrating Voice Mail with Cisco Unified SRST Where to Go Next Cisco Unified Survivable Remote Site Telephony Version 4.
Setting Video Parameters This chapter describes how to set video parameters for a Cisco Unified Survivable Remote Site Telephony (SRST) router. Note Prior to version 4.0, the name of this product was Cisco SRST.
Setting Video Parameters Restrictions for Setting Video Parameters • Perform basic Cisco Unified SRST configuration. For more information, see Cisco Unified SRST V4.0: Setting Up the Network. • Perform basic ephone configuration. For more information, see Cisco Unified SRST V4.0: Setting Up Cisco Unified IP Phones. Restrictions for Setting Video Parameters • This feature supports only the following video codecs: – H.261 – H.
Setting Video Parameters Information About Setting Video Parameters To set video parameters, you should understand the following concepts: • Matching Endpoint Capabilities, page 167 • Retrieving Video Codec Information, page 167 • Call Fallback to Audio-Only, page 167 • Call Setup for Video Endpoints, page 167 • Flow of the RTP Video Stream, page 168 Matching Endpoint Capabilities Endpoint capabilities are stored in the Cisco Unified SRST during phone registration.
Setting Video Parameters Information About Setting Video Parameters During call setup for video, media setup handling determines if a video-media path is required or not. If so, the corresponding video-media-path setup actions are taken. • For an SCCP endpoint, video-media-path setup includes sending messages to the endpoints to open a multimedia path and start the multimedia transmission. • For an H.
Setting Video Parameters How to Set Video Parameters for Cisco Unified SRST Router# show voip rtp connections VoIP RTP active connections : No. CallId dstCallId LocalRTP RmtRTP LocalIP 1 102 103 18714 18158 10.1.1.1 2 105 104 17252 19088 10.1.1.1 Found 2 active RTP connections ============================ RemoteIP 192.168.1.1 192.168.1.
Setting Video Parameters How to Set Video Parameters for Cisco Unified SRST DETAILED STEPS Step 1 Command or Action Purpose enable Enables privileged EXEC mode. • Enter your password if prompted. Example: Router> enable Step 2 configure terminal Enters global configuration mode. Example: Router# configure terminal Step 3 voice service voip Enters voice-service configuration mode. Example: Router(config)# voice service voip Step 4 Enters H.323 voice-service configuration mode.
Setting Video Parameters How to Set Video Parameters for Cisco Unified SRST DETAILED STEPS Step 1 Command or Action Purpose enable Enables privileged EXEC mode. • Enter your password if prompted. Example: Router> enable Step 2 Displays the entire contents of the running configuration file.
Setting Video Parameters How to Set Video Parameters for Cisco Unified SRST Limit number of DNs per phone: 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.
Setting Video Parameters How to Set Video Parameters for Cisco Unified SRST ephone-dn 9 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 huntstop call-waiting beep ephone-dn 13 preference 0 secondary 9 huntstop call-waiting beep ephone-dn 14 preference 0 secondary 9 huntstop call-waiting beep ephone-dn 15 preference 0 secondary 9 huntstop
Setting Video Parameters How to Set Video Parameters for Cisco Unified SRST voice-port 50/0/1 station-id number 1001 station-id name 1001 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 ! voice-port 50/0/10 ! voice-port 50/0/11 ! voice-port 50/0/12 ! voice-port 50/0/13 ! voice-port 50/0/14 ! voice-port 50/0/15 ! voice
Setting Video Parameters How to Set Video Parameters for Cisco Unified SRST huntstop progress_ind setup enable 3 port 50/0/3 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 port 50/0/6 dial-peer voice 20061 pots huntstop progress_ind setup enable 3 port 50/0/7 dial-peer voice 20062 pots huntstop progress_ind setup enable 3 port 50/0/8 dial-p
Setting Video Parameters How to Set Video Parameters for Cisco Unified SRST dial-peer voice 20070 pots huntstop progress_ind setup enable 3 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 huntstop progress_ind setup enable 3 port 50/0/19 dial-peer voice 20074 pots huntstop progress_ind setup enable 3 port 50/0/20 tftp-server system:/its/SEPDEFAULT.
Setting Video Parameters How to Set Video Parameters for Cisco Unified SRST Setting Video Parameters for Cisco Unified SRST Using the following procedure to set the maximum bit rate for all video-capable phones in a Cisco Unified SRST system. SUMMARY STEPS 1. enable 2. configure terminal 3. call-manager-fallback 4. video 5. maximum bit-rate value DETAILED STEPS Step 1 Command or Action Purpose enable Enables privileged EXEC mode. • Enter your password if prompted.
Setting Video Parameters Troubleshooting Video for Cisco Unified SRST max-dn 110 dialplan-pattern 1 4084442... extension-length 4 transfer-pattern .T keepalive 45 voicemail 6001 call-forward pattern .T call-forward busy 6001 call-forward noan 6001 timeout 3 moh music-on-hold.au time-format 24 date-format dd-mm-yy ! Troubleshooting Video for Cisco Unified SRST Use the following commands to troubleshoot Video for Cisco Unified SRST.
Monitoring and Maintaining Cisco Unified SRST To monitor and maintain Cisco Unified Survivable Remote Site Telephony (SRST), use the following commands in the privileged EXEC and mode. Note Prior to version 4.0, the name of this product was Cisco SRST. Command Purpose Router# show running-config Displays the configuration. Router# show call-manager-fallback all Displays the detailed configuration of all the Cisco IP phones, voice ports, and dial peers of the Cisco Unified SRST router.
Monitoring and Maintaining Cisco Unified SRST Cisco Unified Survivable Remote Site Telephony Version 4.
Appendix A: Preparing Cisco Unified SRST Support for SIP Cisco Unified Survivable Remote Site Telephony (SRST) supports incoming and outgoing Session Initiation Protocol (SIP) calls to and from IP phones and router voice gateway voice ports, but does not support direct attachment of SIP phones to Cisco Unified SRST. SIP may be used in situations where the SRST router is separate from the PSTN gateway and the SRST and PSTN gateways are linked together using SIP (instead of H.323).
Appendix A: Preparing Cisco Unified SRST Support for SIP DTMF Relay for SIP Applications and Voice Mail DTMF Relay Using SIP RFC 2833 Cisco Skinny Client Control Protocol (SCCP) phones, such as those used with Cisco Unified SRST systems, provide only out-of-band DTMF digit indications. To enable SCCP phones to send digit information to remote SIP-based IVR and voice-mail applications, Cisco SRST 3.
Appendix A: Preparing Cisco Unified SRST Support for SIP DTMF Relay for SIP Applications and Voice Mail Step 3 Command or Action Purpose exit Exits dial-peer configuration mode. Example: Router(config-dial-peer)# exit Step 4 Enables SIP user-agent configuration mode.
Appendix A: Preparing Cisco Unified SRST Support for SIP DTMF Relay for SIP Applications and Voice Mail 6. exit DETAILED STEPS Step 1 Command or Action Purpose dial-peer voice tag voip Enters dial-peer configuration mode. Example: Router(config)# dial-peer voice 2 voip Step 2 dtmf-relay sip-notify Forwards DTMF tones using SIP NOTIFY messages. Example: Router(config-dial-peer)# dtmf-relay sip-notify Step 3 Exits dial-peer configuration mode.
Appendix A: Preparing Cisco Unified SRST Support for SIP DTMF Relay for SIP Applications and Voice Mail Check media source packets:DISABLED Maximum duration for a telephone-event in NOTIFYs:2000 ms SIP support for ISDN SUSPEND/RESUME:ENABLED Redirection (3xx) message handling:ENABLED SDP application configuration: Version line (v=) required Owner line (o=) required Timespec line (t=) required Media supported:audio image Network types supported:IN Address types supported:IP4 Transport types supported:RTP/AV
Appendix A: Preparing Cisco Unified SRST Support for SIP DTMF Relay for SIP Applications and Voice Mail Cisco Unified Survivable Remote Site Telephony Version 4.
I N D EX call forwarding A 82 during busy signal or no answer access codes trunk to voice mail 89 After Hours Call Blocking 95 after-hours date command 96 after-hours day command 96 96 call-forward pattern command digit translation rules CallManager gateway 101 call setup, for video 66 ANI (answer number indication) application command 166 call transfer 73 analog phones 87 blind area codes and prefix codes 73 86 84 consultative 167 82 consultative using H.450.
Index CIF (common intermediate format) codecs, for video 166 Cisco CallManager common intermediate format, see CIF behavior when WAN is down installing enabling on dual-line phone 31 versions supported for video three-party G.
Index debug ephone message command debug ephone register command debug ephone video command DTMF relay using SIP RFC 2833 178 dual-line mode 178 about 178 15 debug h225 asn1 command 178 dual-line phone debug h245 asn1 command 178 configuring debug voip ccapi inout command default-router command three-party G.711 ad hoc conferencing 178 149 DHCP (Dynamic Host Configuration Protocol) 42 100 E E.
Index voice mail Cisco SRST with Cisco CallManager 147 voice mail with Cisco SRST interface command G 42, 43 ip helper-address command MOH (music on hold) 44 IP routing 101 three-party ad hoc conferencing enabling 100 40 ip source-address command global prefixes configuring 147 44 ip dhcp pool command G.711 32 46 ISDN (Integrated Services Digital Network) 71 voice mail 147 H K H.261 video codec 166 H.263 video codec 166 H.323 endpoint H.450.
Index configuring direct access to voice mail MIBs (Management Information Bases) supported by Cisco SRST preference rules 37, 145 MOH (music on hold) from flash files moh command 66 prefix codes and area codes for G.711, on-net VoIP, and PSTN calls voice mail configuration voice mail N network network command 42 147 PSTN (public switched telephone network) 179 MOH (music on hold) 39 97 PRI (Primary Rate Interface) 101 about setting up 73 preservation, call preservation for H.
Index system log messages S 17 system message command SCCP endpoint secure SRST 168 for configuring customized system messages on Cisco IP phone displays 55 105 secure SRST authentication and encryption service dhcp command 110 44 T SETUP message to Cisco CallManager show call active video command 154 tag 178 show call-manager-fallback all command time format 171, 179 show call-manager-fallback dial-peer command 155 179 setting up on Cisco IP phone display show call-manager-fallback e
Index bandwidth settings call setup 168 codecs supported endpoints 166 167 firmware version 165 formats supported media path video support 166 168 troubleshooting video codecs 177 178 167 165 vm-integration command 156 voice mail call forwarding 154 configuring direct access to how Cisco SRST handles routing of calls 149 147 148 voicemail command 153, 155 voice service voip command 83 VoIP, on-net MOH (music on hold) 101 W WAN when WAN connection is down 23, 39 X xmlschema com
Index Cisco IOS Survivable Remote Site Telephony Version 4.
