Integrating Voice Mail with Cisco Unified SRST
How to Configure DTMF Relay for SIP Applications and Voice Mail
253
Cisco Unified SCCP and SIP SRST System Administrator Guide
OL-13143-04
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:
•
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.
Note
The need to use out-of-band DTMF relay conversion is limited to SCCP phones. SIP phones natively
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.
dial-peer voice
tag
voip
2.
dtmf-relay rtp-nte
3.
exit
4.
sip-ua
5.
notify telephone-event max-duration
time
6.
exit
DETAILED STEPS
Command or Action
Purpose
Step 1
dial-peer voice
tag
voip
Example:
Router(config)# dial-peer voice 2 voip
Enters dial-peer configuration mode.
Step 2
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.
Step 3
exit
Example:
Router(config-dial-peer)# exit
Exits dial-peer configuration mode.
Step 4
sip-ua
Example:
Router(config)# sip-ua
Enables SIP user-agent configuration mode.