Setting Up Cisco Unified IP Phones using SIP
Command or Action
Step 9
dtmf-relay [cisco-rtp] [rtp-nte] [sip-notify]
Example:
Router(config-register-pool)# dtmf-relay
rtp-nte
Step 10
end
Example:
Router(config-register-pool)# end
Examples
The following partial output from the show running-config command shows that voice register pool 12
is configured to accept all registrations from SIP IP phones with extension number 50xx from the
172.16.0.0/16 network. Autogenerated dial peers for registrations that match pool 12 have attributes
configured in this pool.
.
.
.
voice register pool 12
id network 172.16.0.0 mask 255.255.0.0
number 1 50.. preference 2
application SIP.app
preference 2
incoming called-number
cor incoming allowall default
translate-outgoing called 1
voice-class codec 1
.
.
.
Verifying SIP Registrar Configuration
To help you troubleshoot a SIP registrar and voice register pool, perform the following steps.
SUMMARY STEPS
1.
2.
3.
OL-13143-04
debug voice register errors
debug voice register events
show sip-ua status registrar
Purpose
Specifies how a SIP gateway relays dual tone
multifrequency (DTMF) tones between telephony
interfaces and an IP network. The keywords are defined as
follows:
cisco-rtp: (Optional) Forwards DTMF tones by using
•
Real-Time Transport Protocol (RTP) with a Cisco
proprietary payload type.
rtp-nte: (Optional) Forwards DTMF tones by using
•
RTP with the Named Telephone Event (NTE) payload
type.
•
sip-notify: (Optional) Forwards DTMF tones using SIP
NOTIFY messages.
Returns to privileged EXEC mode.
Cisco Unified SCCP and SIP SRST System Administrator Guide
How to Configure the SIP Registrar
115