Configuring SIP Trunk Support Cisco Call Manager Express with 2N VoiceBlue Lite
Released on = July 3, 2006, 7:17 am
Press Release Author = Nicole Konstantinidisova
Industry = Telecommunications
Press Release Summary = This procedure enables four SIP trunk support parameters
Press Release Body = This 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
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.
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.
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.
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.
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.
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:
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.
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
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