Cisco Unified SIP SRST 4.0 System Administrator Guide Cisco IOS Release 12.4(4)XC 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 SIP SRST Feature Roadmap Contents 1 Documentation Organization Feature Roadmap 1 3 Cisco Unified SIP SRST Feature Overview Contents 1 5 5 Cisco Unified SIP SRST Description 5 Support for Cisco Unified IP Phones and Platforms 7 Finding Cisco IOS Software Releases That Support Cisco Unified SRST Cisco Unified IP Phone Support 8 Platform and Memory Support 8 Prerequisites for Configuring Cisco Unified SIP SRST Restrictions for Configuring Cisco Unified SIP SRST Where to
Contents Prerequisites for Configuring the SIP Registrar Restrictions for Configuring the SIP Registrar Information About Configuring the SIP Registrar 23 23 24 How to Configure the SIP Registrar 24 Configuring the SIP Registrar 24 Configuring Backup Registrar Service to SIP Phones 26 Configuring Backup Registrar Service to SIP Phones (Using Optional Commands) Verifying SIP Registrar Configuration 33 Verifying Proxy Dial-Peer Configuration 34 30 Configuring Cisco Unified SIP SRST Features Using Redirec
Contents Cisco Unified SIP SRST: Example 59 INDEX Cisco Unified SIP SRST 4.
Contents Cisco Unified SIP SRST 4.
Cisco Unified SIP SRST Feature Roadmap Note Prior to version 4.0, the name of this product was Cisco SIP SRST. This chapter contains a summary of Cisco Unified Session Initiation Protocol (SIP) Survivable Remote Site Telephony (SRST) features and the location of feature documentation. Use Cisco Feature Navigator to find information about platform support and Cisco IOS software image support. Access Cisco Feature Navigator at http://www.cisco.com/go/fn. You must have an account on Cisco.com.
Cisco Unified SIP SRST Feature Roadmap Documentation Organization Table 1 Cisco Unified SIP SRST Configuration Sequence Chapter or Appendix Description Cisco Unified SIP SRST Feature Overview Gives a brief description of Cisco Unified SIP SRST and provides information on the supported platforms and Cisco Unified IP phones. In addition, it describes any prerequisites or restrictions that should be addressed before Cisco Unified SIP SRST is configured.
Cisco Unified SIP SRST Feature Roadmap Feature Roadmap Feature Roadmap Table 2 provides a summary of Cisco Unified SIP SRST features by release. Table 2 Cisco Unified SIP SRST Features by Cisco IOS Release Cisco SIP SRST Version Cisco IOS Release Modifications Version 4.0 12.4(4)XC — Version 3.4 12.4(4)T Cisco SIP SRST 3.4 includes the following features: • Getting Started • Configuring Cisco Unified SIP SRST Features Using Redirect Mode (for Version 3.
Cisco Unified SIP SRST Feature Roadmap Feature Roadmap Cisco Unified SIP SRST 4.
Cisco Unified SIP SRST Feature Overview Note Prior to version 4.0, the name of this product was Cisco SIP SRST. This chapter includes information about supported Cisco IP phones and platforms. It also includes information on Cisco Unified Session Initiation Protocol (SIP) Survivable Remote Site Telephony (SRST) specifications, features, prerequisites, restrictions, and where to find additional reference documents.
Cisco Unified SIP SRST Feature Overview Cisco Unified SIP SRST Description Cisco Unified SIP SRST can support SIP phones with standard RFC 3261 feature support locally and across SIP WAN networks. With Cisco Unified SIP SRST, SIP phones can place calls across SIP networks in the same way as SCCP phones.
Cisco Unified SIP SRST Feature Overview Support for Cisco Unified IP Phones and Platforms Figure 2 Call Proceeds with Cisco Unified SIP SRST, When WAN Is Down PSTN SIP proxy server IP Dual registration IP 146133 SIP SRST registrar (B2BUA router) WAN SIP phone Support for Cisco Unified IP Phones and Platforms The following sections provide information about Cisco Feature Navigator and the histories of Cisco Unified IP Phone and platform support from Cisco SRST 3.0 to the present version.
Cisco Unified SIP SRST Feature Overview Prerequisites for Configuring Cisco Unified SIP SRST Cisco Unified IP Phone Support For the most up-to-date information about Cisco Unified IP Phone support, see 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 Cisco UnifiedIP Phone 7940G and Cisco Unified IP Phone 7960G are fully supported if dual registration is enabled.
Cisco Unified SIP SRST Feature Overview Prerequisites for Configuring Cisco Unified SIP SRST Note When the WAN goes down, for each outgoing call the SIP phone continues to send the SIP proxy server up to seven Invite messages. If the Invite messages are not acknowledged, the SIP phone switches to Cisco Unified SIP SRST to route the call. Thus, there may be a few seconds delay before SIP SRST takes over call processing from the SIP proxy server.
Cisco Unified SIP SRST Feature Overview Restrictions for Configuring Cisco Unified SIP SRST Restrictions for Configuring Cisco Unified SIP SRST Table 3 provides a history of restrictions from Cisco SIP SRST 3.0 to the present version. Table 3 History of Restrictions from Cisco SIP SRST Version 3.0 to the Present Version Cisco SRST Version Cisco IOS Release Version 4.0 12.4(4)XC Version 3.4 12.4(4)T Version 3.2 12.3(11)T Version 3.1 12.3(7)T Version 3.0 12.2(15)ZJ 12.
Cisco Unified SIP SRST Feature Overview Where to Go Next Where to Go Next The next chapters of this book describe how to configure Cisco Unified SIP SRST. As shown in Table 4, 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.
Cisco Unified SIP SRST Feature Overview Additional References Related Topic Documents Cisco Unified IP Phones • Cisco IP Phone 7902 Quick Start Guide • Cisco IP Phone 7902G Quick Start Guide • At a Glance Cisco IP Phone 7912G • Cisco IP Phone 7960 and 7940 Series User Guide • Cisco IP Phone 7970 Guide • Cisco SIP IP Phone 7960 Administrator Guide, Version 5.
Cisco Unified SIP SRST Feature Overview Additional References Technical Assistance Description Link http://www.cisco.com/techsupport The Cisco Technical Support website contains thousands of pages of searchable technical content, including links to products, technologies, solutions, technical tips, and tools. Registered Cisco.com users can log in from this page to access even more content. Cisco Unified SIP SRST 4.
Cisco Unified SIP SRST Feature Overview Additional References Cisco Unified SIP SRST 4.
Getting Started Note Prior to version 4.0, the name of this product was Cisco SIP SRST. This chapter describes the main tasks necessary for the following: • Running Cisco Session Initiation Protocol (SIP) Survivable Remote Site Telephony (SRST) 3.0 for the first time • Running Cisco Unified SIP SRST 4.0 for the first time • Upgrading from Cisco SIP SRST 3.0 to Cisco Unified SIP SRST 4.0 Note that upgrades from Cisco SIP SRST 3.4 to Cisco Unified SIP SRST 4.
Getting Started Configuration and Upgrade Tasks Cisco Unified SIP SRST V4.0, Cisco IOS Release 12.4(4)XC With Cisco Unified SIP SRST 4.0, a SIP redirect server is not necessary. Instead, a back-to-back user agent (B2BUA) server routes the call as desired. A B2BUA is a separate call agent that has more features than a redirect server, which can accept and forward calls only. With a B2BUA you can also configure call blocking and call forwarding.
Getting Started Configuration and Upgrade Tasks Cisco Unified SIP SRST Version Instructions and Procedures If you are interested in Cisco Unified SIP SRST 4.0 (using a B2BUA) and have never used Cisco Unified SIP SRST in the past, complete these procedures. VoIP-to-VoIP connections permit the termination and reorigination of transferred and forwarded calls over the VoIP network.
Getting Started How to Upgrade from Cisco SIP SRST 3.0 to Cisco Unified SIP SRST 4.0 How to Upgrade from Cisco SIP SRST 3.0 to Cisco Unified SIP SRST 4.0 This section contains the following procedures: • Disabling Call Redirection, page 18 (required) • Enabling SIP-to-SIP Connection Capabilities, page 21 (required) Disabling Call Redirection Because Version 4.0 uses a B2BUA and not a redirect server, call redirection must be disabled if it was previously enabled.
Getting Started How to Upgrade from Cisco SIP SRST 3.0 to Cisco Unified SIP SRST 4.0 Step 3 Command or Action Purpose voice service voip Enters voice service configuration mode. Example: Router(config)# voice service voip Step 4 no redirect ip2ip Disables redirection of SIP phone calls to SIP phone calls globally using the Cisco IOS voice gateway. Example: Router(config-voi-srv)# no redirect ip2ip Step 5 Returns to privileged EXEC mode.
Getting Started How to Upgrade from Cisco SIP SRST 3.0 to Cisco Unified SIP SRST 4.0 Step 3 Command or Action Purpose dial-peer voice tag voip Enters dial-peer configuration mode. • tag—A number that uniquely identifies the dial peer (this number has local significance only). • voip—Indicates that this is a VoIP peer using voice encapsulation on the POTS network and is used for configuring redirect.
Getting Started How to Upgrade from Cisco SIP SRST 3.0 to Cisco Unified SIP SRST 4.0 Enabling SIP-to-SIP Connection Capabilities VoIP-to-VoIP connections permit the termination and reorigination of transferred and forwarded calls over the VoIP network. For Cisco Unified SIP SRST 4.0 we enable SIP-to-SIP connections for hairpin call routing. The B2BUA that routes the call uses the SIP-to-SIP connection.
Getting Started How to Upgrade from Cisco SIP SRST 3.0 to Cisco Unified SIP SRST 4.0 What to Do Next SIP registrar functionality in Cisco IOS software is a required part of Cisco Unified SIP SRST. By default, Cisco Unified SIP SRST is not enabled and cannot accept SIP register messages. To configure the SIP registrar to accept incoming SIP Register messages, see the “Configuring the SIP Registrar” chapter.
Configuring the SIP Registrar Note Prior to version 4.0, the name of this product was Cisco SIP SRST. 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 2543, a SIP registrar is a server that accepts Register requests and is typically collocated with a proxy or redirect server. A SIP registrar may also offer location services.
Configuring the SIP Registrar Information About Configuring the SIP Registrar Information About Configuring the SIP Registrar Cisco Unified SIP SRST provides backup to an external SIP proxy server by providing basic registrar and redirect services. These services are used by a SIP IP phone in the event of a WAN connection outage when the SIP phone is unable to communicate with its primary SIP proxy. The Cisco Unified SIP SRST device also provides PSTN gateway access for placing and receiving PSTN calls.
Configuring the SIP Registrar How to Configure the SIP Registrar Prerequisites The SIP endpoints (IP phones) must support dual concurrent registration, which is registering with the main SIP proxy and the Cisco Unified SIP SRST device (redirect server) at the same time. If this requirement is not met, the Cisco Unified SIP SRST device cannot route incoming calls to the SIP phone. For configuration instructions, see the Cisco IP Phone Documentation for Session Initiation Protocol (SIP). SUMMARY STEPS 1.
Configuring the SIP Registrar How to Configure the SIP Registrar Step 5 Command or Action Purpose registrar server [expires [max sec] [min sec]] Enables SIP registrar functionality. The keywords and arguments are defined as follows: Example: • expires: (Optional) Sets the active time for an incoming registration. • max sec: (Optional) Maximum expiration time for a registration, in seconds. The range is from 600 to 86400. The default is 3600.
Configuring the SIP Registrar How to Configure the SIP Registrar Restrictions Note • The id command identifies the individual SIP IP phone or sets of SIP IP phones that are to be configured. Thus, the id command configured in Step 5 is required and must be configured before any other voice register pool commands.
Configuring the SIP Registrar How to Configure the SIP Registrar Step 3 Command or Action Purpose call fallback active (Optional) Enables a call request to fall back to alternate dial peers in case of network congestion. Example: • Router(config)# call fallback active Step 4 voice register pool tag Example: Enters voice register pool configuration mode for SIP phones.
Configuring the SIP Registrar How to Configure the SIP Registrar Step 7 Command or Action Purpose proxy ip-address [preference value] [monitor probe {icmp-ping | rtr} [alternate-ip-address]] Autogenerates additional VoIP dial peers to reach the main 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: Example: Router(config-register-pool)# proxy 10.2.161.
Configuring the SIP Registrar How to Configure the SIP Registrar What to Do Next There are several more voice register pool commands that add functionality, but that are not required. See the “Configuring Backup Registrar Service to SIP Phones (Using Optional Commands)” section on page 30 for these commands. Configuring Backup Registrar Service to SIP Phones (Using Optional Commands) The prior configurations set up a basic voice register pool.
Configuring the SIP Registrar How to Configure the SIP Registrar 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 register pool tag Enters voice register pool configuration mode.
Configuring the SIP Registrar How to Configure the SIP Registrar Step 6 Command or Action Purpose cor {incoming | outgoing} cor-list-name {cor-list-number starting-number [ending-number] | default} Configures a class of restriction (COR) on the VoIP dial peers associated with directory numbers. COR specifies which incoming dial peers can use which outgoing dial peers to make a call. Each dial peer can be provisioned with an incoming and outgoing COR list.
Configuring the SIP Registrar How to Configure the SIP Registrar Step 9 Command or Action Purpose dtmf-relay [cisco-rtp] [rtp-nte] [sip-notify] Specifies how a SIP gateway relays dual tone multifrequency (DTMF) tones between telephony interfaces and an IP network. The keywords are defined as follows: Example: Router(config-register-pool)# dtmf-relay rtp-nte Step 10 • cisco-rtp: (Optional) Forwards DTMF tones by using Real-Time Transport Protocol (RTP) with a Cisco proprietary payload type.
Configuring the SIP Registrar How to Configure the SIP Registrar DETAILED STEPS Step 1 debug voice register errors Use this command to debug errors that happen during registration, for example: Router# debug voice register errors *Apr *Apr *Apr *Apr *Apr 22 22 22 22 22 11:52:54.523 11:52:54.539 11:52:54.539 11:52:54.559 11:53:04.
Configuring the SIP Registrar How to Configure the SIP Registrar SUMMARY STEPS 1. configure terminal 2. voice register pool tag 3. proxy ip-address [preference value] [monitor probe {icmp-ping | rtr} [alternate-ip-address]] 4. end 5. show voice register dial-peers 6. show dial-peer voice DETAILED STEPS Step 1 configure terminal Use this command to enter global configuration mode.
Configuring the SIP Registrar How to Configure the SIP Registrar Step 6 show dial-peer voice Finally, use the show dial-peer voice command on dial peer 40036, and notice the monitor probe status. Note Also highlighted is the output of the cor and incoming called-number commands.
Configuring the SIP Registrar How to Configure the SIP Registrar Playout Mode is set to adaptive, Initial 60 ms, Max 300 ms Playout-delay Minimum mode is set to default, value 40 ms Fax nominal 300 ms Max Redirects = 1, signaling-type = cas, VAD = enabled, Poor QOV Trap = disabled, Source Interface = NONE voice class sip url = system, voice class sip rel1xx = system, redirect ip2ip = enabled monitor probe method: icmp-ping ip address: 10.2.161.
Configuring the SIP Registrar How to Configure the SIP Registrar Cisco Unified SIP SRST 4.
Configuring Cisco Unified SIP SRST Features Using Redirect Mode (for Version 3.0 Only) Note Prior to version 4.0, the name of this product was Cisco SIP SRST. This chapter describes Cisco Unified Session Initiation Protocol (SIP) Survivable Remote Site Telephony (SRST) features using redirect mode. 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.
Configuring Cisco Unified SIP SRST Features Using Redirect Mode (for Version 3.0 Only) Restrictions for Cisco Unified SIP SRST Features Using Redirect Mode Restrictions for Cisco Unified SIP SRST Features Using Redirect Mode See the restrictions documented in the “Restrictions for Configuring Cisco Unified SIP SRST” section in the “Cisco Unified SIP SRST Feature Overview” chapter.
Configuring Cisco Unified SIP SRST Features Using Redirect Mode (for Version 3.0 Only) 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 The call redirect enhancement supports calls from a local SIP phone to another local SIP phone through the Cisco IOS voice gateway.
Configuring Cisco Unified SIP SRST Features Using Redirect Mode (for Version 3.0 Only) How to Configure Cisco Unified SIP SRST Features Using Redirect Mode Step 3 Command or Action Purpose voice service voip Enters voice service configuration mode. Example: Router(config)# voice service voip Step 4 Redirects SIP phone calls to SIP phone calls globally on a gateway using the Cisco IOS voice gateway.
Configuring Cisco Unified SIP SRST Features Using Redirect Mode (for Version 3.0 Only) How to Configure Cisco Unified SIP SRST Features Using Redirect Mode 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 dial-peer voice tag voip Enters dial-peer configuration mode.
Configuring Cisco Unified SIP SRST Features Using Redirect Mode (for Version 3.0 Only) How to Configure Cisco Unified SIP SRST Features Using Redirect Mode SUMMARY STEPS 1. enable 2. configure terminal 3. voice service voip 4. sip 5. redirect contact order [best-match | longest-match] 6. end DETAILED STEPS Step 1 Command or Action Purpose enable Enables privileged EXEC mode. • Enter your password if prompted.
Configuring Cisco Unified SIP SRST Features Using Redirect Mode (for Version 3.0 Only) Configuration Examples for Cisco Unified SIP SRST Features Using Redirect Mode Configuration Examples for Cisco Unified SIP SRST Features Using Redirect Mode This section provides the following configuration example. • Note Cisco Unified SIP SRST: Example IP addresses and hostnames in examples are fictitious.
Configuring Cisco Unified SIP SRST Features Using Redirect Mode (for Version 3.0 Only) Where to Go Next ! voice register pool 4 id network 10.2.161.0 mask 255.255.255.0 number 1 94... preference 1 preference 5 cor incoming everywhere default cor outgoing everywhere default proxy 10.2.161.187 preference 1 max registrations 2 voice-class codec 1 ! ! Configures translation rules to be applied in the voice register pools. ! translation-rule 1 Rule 0 94 91 ! ! Sets up proxy monitoring.
Configuring Cisco Unified SIP SRST Features Using Back-to-Back User Agent Mode (for Version 3.4 and Version 4.0 Only) Note Prior to version 4.0, the name of this product was Cisco SIP SRST. This chapter describes Cisco Unified Cisco Unified Survivable Remote Site Telephony (SRST) support for standardized RFC 3261 features for SIP phones. Features include call blocking and call forwarding.
Configuring Cisco Unified SIP SRST Features Using Back-to-Back User Agent Mode (for Version 3.4 and Version 4.0 Restrictions for Cisco Unified SIP SRST Features Using Back-to-Back User Agent Mode • Complete the necessary tasks found in the “Getting Started” chapter. Specific tasks include the required task that is documented in the “Enabling SIP-to-SIP Connection Capabilities” section on page 21. • Configure the SIP registrar.
Configuring Cisco Unified SIP SRST Features Using Back-to-Back User Agent Mode (for Version 3.4 and Version 4.0 Information About Cisco Unified SIP SRST Features Using Back-to-Back User Agent Mode Procedures for configuring Cisco Unified SIP CME and complete descriptions of all CME and voice register dn commands are found in the Cisco CallManager Express Version 3.4 documentation. Note Table 5 is not all-inclusive; additional commands may exist. Table 5 Version 3.
Configuring Cisco Unified SIP SRST Features Using Back-to-Back User Agent Mode (for Version 3.4 and Version 4.0 Information About Cisco Unified SIP SRST Features Using Back-to-Back User Agent Mode Table 5 Version 3.
Configuring Cisco Unified SIP SRST Features Using Back-to-Back User Agent Mode (for Version 3.4 and Version 4.
Configuring Cisco Unified SIP SRST Features Using Back-to-Back User Agent Mode (for Version 3.4 and Version 4.0 How to Configure Cisco Unified SIP SRST Step 3 Command or Action Purpose voice register global tag Enters voice register global configuration mode to set global parameters for all supported Cisco SIP IP phones in a Cisco Unified SIP SRST environment.
Configuring Cisco Unified SIP SRST Features Using Back-to-Back User Agent Mode (for Version 3.4 and Version 4.0 How to Configure Cisco Unified SIP SRST Step 10 Command or Action Purpose codec codec-type [bytes] Specifies the codec supported by a single SIP phone or a VoIP dial peer in a Cisco Unified SIP SRST environment. The codec-type argument specifies the preferred codec and can be one of the following: Example: Router(config-register-pool)# codec g729r8 • g711alaw—G.711 a–law 64,000 bps.
Configuring Cisco Unified SIP SRST Features Using Back-to-Back User Agent Mode (for Version 3.4 and Version 4.0 How to Configure Cisco Unified SIP SRST SUMMARY STEPS 1. enable 2. configure terminal 3. voice register pool tag 4. call-forward b2bua all directory-number 5. call-forward b2bua busy directory-number 6. call-forward b2bua mailbox directory-number 7. call-forward b2bua noan directory-number timeout seconds 8.
Configuring Cisco Unified SIP SRST Features Using Back-to-Back User Agent Mode (for Version 3.4 and Version 4.0 How to Configure Cisco Unified SIP SRST Step 6 Command or Action Purpose call-forward b2bua mailbox directory-number Controls the specific voice-mail box selected in a voice-mail system at the end of a call forwarding exchange. directory-number—Telephone number to which calls are forwarded when the forwarded destination is busy or does not answer. Represents a fully qualified E.164 number.
Configuring Cisco Unified SIP SRST Features Using Back-to-Back User Agent Mode (for Version 3.4 and Version 4.0 How to Configure Cisco Unified SIP SRST When a user attempts to place a call to digits that match a pattern that has been specified for call blocking during a time period that has been defined for call blocking, the call is immediately terminated and the caller hears a fast busy.
Configuring Cisco Unified SIP SRST Features Using Back-to-Back User Agent Mode (for Version 3.4 and Version 4.0 How to Configure Cisco Unified SIP SRST Step 5 Command or Action Purpose after-hours day day start-time stop-time Defines a recurring time period based on the day of the week during which calls are blocked to outgoing dial patterns that are defined using the after-hours block pattern command.
Configuring Cisco Unified SIP SRST Features Using Back-to-Back User Agent Mode (for Version 3.4 and Version 4.0 How to Configure Cisco Unified SIP SRST Step 9 Command or Action Purpose after-hour exempt Specifies that for a particular voice register pool, none its outgoing calls are blocked even though call blocking is enabled. Example: Router(config-register-pool)# after-hour exempt Step 10 Returns to privileged EXEC mode.
Configuring Cisco Unified SIP SRST Features Using Back-to-Back User Agent Mode (for Version 3.4 and Version 4.0 Configuration Examples for Cisco Unified SIP SRST Features Using Back-to-Back User Agent Mode Note Music on hold (MOH) is not supported for call hold invoked from a SIP phone. A caller hears only silence when placed on hold by a SIP phone. Configuration Examples for Cisco Unified SIP SRST Features Using Back-to-Back User Agent Mode This section provides the following configuration example.
Configuring Cisco Unified SIP SRST Features Using Back-to-Back User Agent Mode (for Version 3.4 and Version 4.0 Configuration Examples for Cisco Unified SIP SRST Features Using Back-to-Back User Agent Mode allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip sip registrar server expires max 600 min 60 ! ! ! voice register global max-dn 10 max-pool 10 ! ! Define call forwarding under a voice register pool voice register pool 1 id mac 0012.7F57.
Configuring Cisco Unified SIP SRST Features Using Back-to-Back User Agent Mode (for Version 3.4 and Version 4.0 Configuration Examples for Cisco Unified SIP SRST Features Using Back-to-Back User Agent Mode max-conferences 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 login ! scheduler allocate 20000 1000 ntp server 10.0.2.10 ! end Cisco Unified SIP SRST 4.
Configuring Cisco Unified SIP SRST Features Using Back-to-Back User Agent Mode (for Version 3.4 and Version 4.0 Configuration Examples for Cisco Unified SIP SRST Features Using Back-to-Back User Agent Mode Cisco Unified SIP SRST 4.
I N D EX cor command A after-hour exempt command 30 58 after-hours block pattern command D 56 after-hours date command 57 description of SIP SRST after-hours day command 57 documentation references alias command 30 dtmf-relay command allow-connections command application command 5 11 30 21 27, 51 E external ring command C call blocking configuration 55 call-forward b2bua all command F 54 call-forward b2bua busy command 54 feature roadmap call-forward b2bua mailbox command call
Index voice-class codec command N 27 voice register global command number command 30 voice register pool command VoIP-to-VoIP connections configuring P platforms supported by each SRST version preference command 8 27 prerequisites for configuring Cisco SIP SRST proxy command 8 27 R redirect contact order command redirect ip2ip command 44 18, 41 registrar server command SIP networks 25 restrictions for each Cisco SRST version 10 RFCs supported by Cisco SIP SRST 12 S show dial-peer voice
