In order to have CUCM use a SIP trunk and SIP OPTIONS to monitor the router the minimum required is a dial-peer setup, without this the SIP OPTIONS PING from CUCM to the VGW is ignored as its TCP the response to the initial TCP SYN from CUCM is a ACK RST from the VGW.

dial-peer voice 110 voip
session protocol sipv2
incoming called-number .
dtmf-relay sip-kpml
fax rate disable
no vad

To ensure that the correct IP address is used in the SIP responses set the source interfaces;

voice service voip
bind control source-interface FastEthernet0/0.300
bind media source-interface FastEthernet0/0.300

The SIP process is “Session Application”

BR1#sh proc 366
Process ID 366 [Session Application], TTY 0
Memory usage [in bytes]
Holding: 29428, Maximum: 0, Allocated: 297292, Freed: 0
Getbufs: 0, Retbufs: 0, Stack: 19924/24000
CPU usage
PC: 63A0A050, Invoked: 6, Giveups: 0, uSec: 2000
5Sec: 0.00%, 1Min: 0.01%, 5Min: 0.00%, Average: 0.07%
Age: 16792 msec, Runtime: 12 msec
State: Waiting for Event, Priority: Normal



show sip status – will show bound media/control interfaces and seems if these are shown as disabled then the router is no accepting SIP messages.


Simon Birtles

I have been in the IT sector for over 20 years with a primary focus on solutions around networking architecture & design in Data Center and WAN. I have held two CCIEs (#20221) for over 12 years with many retired certifications with Cisco and Microsoft. I have worked in demanding and critical sectors such as finance, insurance, health care and government providing solutions for architecture, design and problem analysis. I have been coding for as long as I can remember in C/C++ and Python (for most things nowadays). Locations that I work without additional paperwork (incl. post Brexit) are the UK and the EU including Germany, Netherlands, Spain and Belgium.