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Configuring SIP Trunk Support CCME

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-eventmax-duration command

•SIP registrar—registrar, retry, and timers commands

SUMMARY STEPS1. enable2. configure terminal3. telephony-service4. call-forward pattern pattern

5. calling-number local6. transfer-system {full-blind | full-consult}7. transfer-pattern transfer-pattern8. exit9. dial-peer voice tag voip10. dtmf-relay rtp-nte11. dtmf-relay sip-notify

12. exit13. sip-ua14. notify telephone-event max-duration time15. registrar {dns:host-name | ipv4:ip-address} expires seconds [tcp] [secondary]16. retry register number17. timers register time18. exit

DETAILED STEPSCommand or ActionPurpose

Step 1enable

Example:Router> enable

Enables privileged EXEC mode.•Enter your password if prompted.

Step 2

configure terminalExample:Router# configure terminal

Enters global configuration mode.



telephony-service

Step 3telephony-service

Example:Router(config)# telephony-service

Enters telephony-service configuration mode.

Step 4call-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-partynumbers 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 redirectionmethod. A pattern of .T matches all calling-party numbers.

Note When defining forwards to nonlocal numbers, it is important to note that pattern-digit matching isperformed before translation-rule operations. Therefore, you should specify in this command the digitsactually entered by phone users before they are translated. For more information, see the "VoiceTranslation Rules and Profiles" section on page 117.

Step 5calling-number local

Example:Router(config-telephony)# calling-number local

(Optional) Replaces a calling-party number and name with the forwarding-party (local) number andname.

Step 6transfer-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 moreinformation, 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 7transfer-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 notransfer 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 nonlocal numbers, it is important to note that transfer-pattern digitmatching 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 moreinformation, see the "Voice Translation Rules and Profiles" section on page 117.

Step 8exit

Example:Router(config-telephony)# exit

Exits telephony-service configuration mode.

Configuration EXAMPLE:

te lephony-service

load 7960-7940 P0030702T023

max-epho nes 24

max-dn 24

ip sou rce-address 172.24.34.160 port 2000

time-fo rmat 24

date-format dd -mm-yy

max-co nferen ces 12 gain -6

moh music-on -hold.au

web admin syst em name xxx secret xxx

dn-webedit

t ime-webedit

transfer-system ful l -consu lt

create cnf -fi les versio n-stamp 7960 May 11 2006 17:52:56

dial-peer voiceStep 9dial-peer voice tag voip

Example:Router(config)# dial-peer voice 2 voip

Enters dial-peer configuration mode.

Step 10dtmf-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 11dtmf-relay sip-notify

Example:

Router(config-dial-peer)# dtmf-relay sip-notify

Forwards DTMF tones using SIP NOTIFY messages.



Step 12exit

Example:Router(config-dial-peer)# exit

Exits dial-peer configuration mode.


dial-peer voice 20 vo ip

desti nation -pattern 004219[01]T

session protoco l s ipv2

sess ion target ip v4:172.24.34.169

dtmf-relay sip -noti fy

cod ec g711ulaw

no vad

sip-ua

Step 13sip-ua

Example:Router(config)# sip-ua

Enters SIP user-agent configuration mode.

Step 14notify 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 asingle 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 15registrar {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 voiceports (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 redundancypurposes.



Step 16retry 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 17timers 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 18exit

Example:Router(config-sip-ua)# exit

Exits SIP user-agent configuration mode.


sip-ua

!

!

!

voice register global

mode cme

source-address 172.24.34.160 por t 5060

load 7960-7940 P0S3-07-4-00create prof il e 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 potsdestination-pattern 9.Tport 1/0/0

!



dial-peer voice 4000 voipdestination-pattern 4...session protocol sipv2session-target ipv4:1.1.1.1

!telephony-servicecall-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 phonesserviced by the router:!dial-peer voice 100 potsdestination-pattern 9.Tport 1/0/0

!dial-peer voice 4000 voipdestination-pattern 4...session protocol sipv2session-target ipv4:1.1.1.1

!telephony-servicetransfer-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 usingdial peer 2.

dial-peer voice 2 voipdtmf-relay rtp-nte

sip-uanotify 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 usingdial peer 4.

dial-peer voice 4 voipdtmf-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 withan external SIP registrar.

sip-uaregistrar ipv4:10.8.17.40 expires 3600 secondaryretry register 10timers register 500



Troubleshooting SIP Trunk SupportFeaturesStep 1 The show sip-ua status command output displays the time interval between consecutiveNOTIFY messages for a telephone event. In the following example, the time interval is 2000 ms.

Router#show sip-ua statusSIP User Agent StatusSIP User Agent for UDP :ENABLEDSIP User Agent for TCP :ENABLEDSIP User Agent bind status(signaling):DISABLEDSIP User Agent bind status(media):DISABLEDSIP early-media for 180 responses with SDP:ENABLEDSIP max-forwards :6SIP DNS SRV version:2 (rfc 2782)NAT Settings for the SIP-UA

Role in SDP:NONECheck media source packets:DISABLEDMaximum duration for a telephone-event in NOTIFYs:2000 msSIP support for ISDN SUSPEND/RESUME:ENABLEDRedirection (3xx) message handling:ENABLED

SDP application configuration:Version line (v=) requiredOwner line (o=) requiredTimespec line (t=) requiredMedia supported:audio imageNetwork types supported:IN Address types supported:IP4Transport types supported:RTP/AVP udptl

Step 2 Use the show sip-ua timers command to show the waiting time before Register requests aresent; 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

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