Setting Up Cisco Unified IP Phones using SIP
How to Configure the SIP Registrar
115
Cisco Unified SCCP and SIP SRST System Administrator Guide
OL-13143-04
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.
debug voice register errors
2.
debug voice register events
3.
show sip-ua status registrar
Step 9
dtmf-relay
[
cisco-rtp
] [
rtp-nte
] [
sip-notify
]
Example:
Router(config-register-pool)# dtmf-relay
rtp-nte
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.
Step 10
end
Example:
Router(config-register-pool)# end
Returns to privileged EXEC mode.
Command or Action
Purpose