Cisco CP-7911G-CH1 System Administrator Manual page 253

Integrating Voice Mail with Cisco Unified SRST
versions provide conversion from the out-of-band SCCP digit indication to the SIP standard for DTMF
relay, which is RFC 2833. You select this method in the SIP VoIP dial peer using the dtmf-relay rtp-nte
command.
The SIP DTMF relay method is needed in the following situations:
•
•
The need to use out-of-band DTMF relay conversion is limited to SCCP phones. SIP phones natively
Note
support in-band DTMF relay as specified in RFC 2833.
To enable SIP DTMF relay using RFC 2833, the commands in this section must be used on both
originating and terminating gateways.
SUMMARY STEPS
1.
2.
3.
4.
5.
6.
DETAILED STEPS
Command or Action
Step 1
dial-peer voice tag voip
Example:
Router(config)# dial-peer voice 2 voip
Step 2
dtmf-relay rtp-nte
Example:
Router(config-dial-peer)# dtmf-relay rtp-nte
Step 3
exit
Example:
Router(config-dial-peer)# exit
Step 4
sip-ua
Example:
Router(config)# sip-ua
OL-13143-04
When SIP is used to connect a Cisco Unified SRST system to a remote SIP-based IVR or voice-mail
application, such as Cisco Unity.
When SIP is used to connect a Cisco Unified SRST system to a remote SIP-PSTN voice gateway
that goes through the PSTN to a voice-mail or IVR application.
dial-peer voice tag voip
dtmf-relay rtp-nte
exit
sip-ua
notify telephone-event max-duration time
exit
How to Configure DTMF Relay for SIP Applications and Voice Mail
Purpose
Enters dial-peer configuration mode.
Forwards DTMF tones by using Real-Time Transport
Protocol (RTP) with the Named Telephone Event
(NTE) payload type.
Exits dial-peer configuration mode.
Enables SIP user-agent configuration mode.
Cisco Unified SCCP and SIP SRST System Administrator Guide
253