Configuring SIP Trunk Support CCME

This procedure enables four SIP trunk support parameters:

Call forwarding over SIP networkscall-forward pattern and calling-number local commands
Call transfer over SIP networkstransfer-system and transfer-pattern commands
DTMF relaydtmf-relay rtp-nte or dtmf-relay sip-notify command and notify telephone-event
max-duration command
SIP registrarregistrar, 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 Ciscoproprietary call forwarding for backward compatibility.
patternDigits 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 nonlocal 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-blindCalls are transferred without consultation using H.450.2 standard methods.
full-consultCalls 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-patternString of digits for permitted call transfers. Wildcards are allowed.
Note When defining transfers to nonlocal 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 timeTime 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-nameDomain name server that resolves the name of the dial peer to receive calls.
ipv4:ip-addressIP address of the dial peer to receive calls.
expires secondsDefault 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.
numberNumber 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.
timeWaiting 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