Cisco 7965 Registration - SIP45.8-5-4S


I’m trying to get a Cisco 7965 registered to my Asterisk server:

Asterisk SVN-branch-12-r400020

All things seem to be working fine but the phone does not register as the last step. I’m wondering if anyone has some experience to make a suggestion.

What I have confirmed working:

  • Phone uses hard-coded option 150 which points to the Asterisk TFTP server
  • Phone is able to download it’s SEPXXXXXX.cnf.xml file and obtains the settings within
  • Phone is able to download and install updated firmware (updated to SIP45.8-5-4S, which was confirmed working on another post here)
  • Phone attempts to register with the Asterisk server and Asterisk replies 200 OK

The phone is stuck in a registration loop. When I watch the SIP debugs on the Asterisk console it’s a series of Register and 200 OK messages between the phone and server. To me it appears there’s an issue with the 200 OK message getting back to the phone, or being accepted by the phone. I have an NTP source defined in the phones config file and even though it doesn’t appear to be using that NTP server, it should use the 200 OK message from the registration request to update it’s time, which it doesn’t do. I’m pretty confident that the messages are getting back to the phone from Asterisk since the tftp communication is working ,which also uses UDP.

Any thoughts on a next-step or something missing from the above?


I pulled a wireshark trace from the span port on the phone and I see the Asterisk messages arriving fine. Also, when I reset the phone with verbose debugging on Asterisk console it tells me that the phone has unregistered so it seems like even Asterisk thinks that the phone is registered, but I only get a red x on the line appearance on the phone with no dial-tone. I can post relevant sip.conf/phone xml configs if needed.

What hardware version of the phone were you using? My version did not allow loading firmware below 9.3(1)SR1. My symptoms are similar to yours but i was able to get out of the registration loop by emptying out the element. I did get a dial tone but could not make calls or receive calls and AsteriskNOW never received any sip messages from the phone.


UPDATE: Got my 7965g working with the latest 9.x firmware and latest version of asterisk now. had to configure to UDP and the USECALLMANAGER configuration in the proxy from a response in another thread:

with versions 9.x you should use the keyword USECALLMANAGER instead of IP address in proxy tag in line section. eg instead of use USECALLMANAGER

WARNING: what ever you do. no NOT empty out the element as i was doing. it needs to be populated and was part the issue i was having.