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Configuring SIP Trunk Support Cisco Call Manager Express with 2N VoiceBlue LiteThis procedure enables four SIP trunk support parameters: •Call forwarding over SIP networks—call-forward pattern and calling-number local commands •Call transfer over SIP networks—transfer-system and transfer-pattern commands •DTMF relay—dtmf-relay rtp-nte or dtmf-relay sip-notify command and notify telephone-event max-duration command •SIP registrar—registrar, retry, and timers commands SUMMARY STEPS 1. enable 2. configure terminal 3. telephony-service 4. call-forward pattern pattern 5. calling-number local 6. transfer-system {full-blind | full-consult} 7. transfer-pattern transfer-pattern 8. exit 9. dial-peer voice tag voip 10. dtmf-relay rtp-nte 11. dtmf-relay sip-notify 12. exit 13. sip-ua 14. notify telephone-event max-duration time 15. registrar {dns:host-name | ipv4:ip-address} expires seconds [tcp] [secondary] 16. retry register number 17. timers register time 18. exit DETAILED STEPS Command or Action Purpose Step 1 enable Example: Router> enable Enables privileged EXEC mode. •Enter your password if prompted. Step 2 configure terminal Example: Router# configure terminal Enters global configuration mode. telephony-service Step 3 telephony-service Example: Router(config)# telephony-service Enters telephony-service configuration mode. Step 4 call-forward pattern pattern Example: Router(config-telephony)# call-forward pattern 4... Specifies the H.450.3 standard or SIP 302 redirection method for call forwarding. Calling-party numbers that do not match the patterns defined with this command are forwarded using Cisco-proprietary call forwarding for backward compatibility. •pattern—Digits to match for call forwarding using the H.450.3 standard or SIP 302 redirection method. A pattern of .T matches all calling-party numbers. Note when defining forwards to non local numbers, it is important to note that pattern-digit matching is performed before translation-rule operations. Therefore, you should specify in this command the digits actually entered by phone users before they are translated. For more information, see the "Voice Translation Rules and Profiles" section on page 117. Step 5 calling-number local Example: Router(config-telephony)# calling-number local (Optional) Replaces a calling-party number and name with the forwarding-party (local) number and name. Step 6 transfer-system {full-blind | full-consult} Example: Router(config-telephony)# transfer-system full-consult Defines the call transfer method for all lines served by the router. Note For SIP networks, use only the full-blind keyword or the full-consult keyword. For more information, see the Cisco IOS SIP Configuration Guide. •full-blind—Calls are transferred without consultation using H.450.2 standard methods. •full-consult—Calls are transferred with consultation using H.450.2 standard methods and a second phone line if available. The calls fall back to full-blind if the second line is unavailable. Step 7 transfer-pattern transfer-pattern Example: Router(config-telephony)# transfer-pattern 52540.. Allows transfer of telephone calls by Cisco Unified IP phones to specified phone number patterns. If no transfer pattern is set, the default is that transfers are permitted only to other local IP phones. •transfer-pattern—String of digits for permitted call transfers. Wildcards are allowed. Note when defining transfers to non local numbers, it is important to note that transfer-pattern digit matching is performed before translation-rule operations. Therefore, you should specify in this command the digits that are actually entered by phone users before they are translated. For more information, see the "Voice Translation Rules and Profiles" section on page 117. Step 8 exit Example: Router(config-telephony)# exit Exits telephony-service configuration mode. Configuration EXAMPLE: telephony-service load 7960-7940 P0030702T023 max-ephones 24 max-dn 24 ip source-address 172.24.34.160 port 2000 time-format 24 date-format dd-mm-yy max-conferences 12 gain -6 moh music-on-hold.au web admin system name xxx secret xxx dn-webedit time-webedit transfer-system full-consult create cnf-files version-stamp 7960 May 11 2006 17:52:56 dial-peer voice Step 9 dial-peer voice tag voip Example: Router(config)# dial-peer voice 2 voip Enters dial-peer configuration mode. Step 10 dtmf-relay rtp-nte Example: Router(config-dial-peer)# dtmf-relay rtp-nte Forwards DTMF tones by using Real-Time Transport Protocol (RTP) with the Named Telephone Event (NTE) payload type. This enables DTMF relay using the RFC 2833 standard method. Step 11 dtmf-relay sip-notify Example: Router(config-dial-peer)# dtmf-relay sip-notify Forwards DTMF tones using SIP NOTIFY messages. Step 12 exit Example: Router(config-dial-peer)# exit Exits dial-peer configuration mode. Configuration EXAMPLE: dial-peer voice 20 voip destination-pattern 004219[01]T session protocol sipv2 session target ipv4:172.24.34.169 dtmf-relay sip-notify codec g711ulaw no vad sip-ua Step 13 sip-ua Example: Router(config)# sip-ua Enters SIP user-agent configuration mode. Step 14 notify telephone-event max-duration time Example: Router(config-sip-ua)# notify telephone-event max-duration 2000 Configures the maximum time interval allowed between two consecutive NOTIFY messages for a single DTMF event. •max-duration time—Time interval between consecutive NOTIFY messages for a single DTMF event, in milliseconds. Range is from 500 to 3000. Default is 2000. Step 15 registrar {dns:host-name | ipv4:ip-address} expires seconds [tcp] [secondary] Example: Router(config-sip-ua)# registrar ipv4:10.8.17.40 expires 3600 secondary Registers E.164 numbers on behalf of analog telephone voice ports (FXS) and IP phone virtual voice ports (EFXS) with an external SIP proxy or SIP registrar server. •dns:host-name—Domain name server that resolves the name of the dial peer to receive calls. •ipv4:ip-address—IP address of the dial peer to receive calls. •expires seconds—Default registration time, in seconds. •tcp—(Optional) Sets the transport layer protocol to TCP. UDP is the default. •secondary—(Optional) Specifies registration with a secondary SIP proxy or registrar for redundancy purposes. Step 16 retry register number Example: Router(config-sip-ua)# retry register 10 Sets the total number of SIP Register messages that the gateway should send. •number—Number of Register message retries. Range is from 1 to 10. Default is 10. Step 17 timers register time Example: Router(config-sip-ua)# timers register 500 Sets how long the SIP user agent (UA) waits before sending Register requests. •time—Waiting time, in milliseconds. Range is from 100 to 1000. Default is 500. Step 18 exit Example: Router(config-sip-ua)# exit Exits SIP user-agent configuration mode. Configuration EXAMPLE: sip-ua ! ! ! voice register global mode cme source-address 172.24.34.160 port 5060 load 7960-7940 P0S3-07-4-00 create profile sync 0052141959334142 Verifying SIP Trunk Support Features Step 1. Use the show running-config command to verify dial-peer, telephony-service, and SIP UA parameter values. Call Forwarding over SIP Networks: Example The following example enables call forwarding using the H.450.3 standard or SIP 302 response: dial-peer voice 100 pots destination-pattern 9.T port 1/0/0 ! dial-peer voice 4000 voip destination-pattern 4... session protocol sipv2 session-target ipv4:1.1.1.1 ! telephony-service call-forward pattern 4... Call Transfer over SIP Networks: Example The following example specifies transfer with consultation using the H.450.2 standard for all IP phones serviced by the router: ! dial-peer voice 100 pots destination-pattern 9.T port 1/0/0 ! dial-peer voice 4000 voip destination-pattern 4... session protocol sipv2 session-target ipv4:1.1.1.1 ! telephony-service transfer-pattern 4... transfer-system full-consult DTMF Relay using RFC 2833: Example The following example specifies use of the RFC 2833 method for in-band DTMF relay for calls using dial peer 2. dial-peer voice 2 voip dtmf-relay rtp-nte sip-ua notify telephone-event max-duration 2000 DTMF Relay using SIP Notify:Example The following example specifies use of the SIP notify method for in-band DTMF relay for calls using dial peer 4. dial-peer voice 4 voip dtmf-relay sip-notify sip-ua notify telephone-event max-duration 2000 SIP Register Support: Example The following example sets up the gateway to register the gateway's E.164 telephone numbers with an external SIP registrar. sip-ua registrar ipv4:10.8.17.40 expires 3600 secondary retry register 10 timers register 500 Troubleshooting SIP Trunk Support Features Step 1 The show sip-ua status command output displays the time interval between consecutive NOTIFY messages for a telephone event. In the following example, the time interval is 2000 ms. Router# show sip-ua status SIP User Agent Status SIP User Agent for UDP :ENABLED SIP User Agent for TCP :ENABLED SIP User Agent bind status(signaling):DISABLED SIP User Agent bind status(media):DISABLED SIP early-media for 180 responses with SDP:ENABLED SIP max-forwards :6 SIP DNS SRV version:2 (rfc 2782) NAT Settings for the SIP-UA Role in SDP:NONE Check media source packets:DISABLED Maximum duration for a telephone-event in NOTIFYs:2000 ms SIP support for ISDN SUSPEND/RESUME:ENABLED 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/AVP udptl Step 2 Use the show sip-ua timers command to show the waiting time before Register requests are sent; that is, the value that has been set with the timers register command. Step 3 Use the show sip-ua register status command to show the status of local E.164 registrations. Step 4 Use the show sip-ua statistics command to show the Register messages that have been sent. Feature History for SIP Trunk Features Cisco Unified CME Version Modification |