Cisco 7961G IP phone cannot registry to Asterisk server

Dear all

I tried to use Cisco 7961G connect to Asterisk at Raspberry pi. The phone 's firmware was refreshed to SIP version. However, it cannot registry to Asterisk (prompt “registering” long long time) . And the report page of FreePBX shown no online.

I had created the extension already.

I refer some guideline told remove everything between , so I remove “true” on the true. Now the section became:










Yes, the "registering " no longer prompt, but the report page of FreePBX still shown no online.

I tried to install a software SIP app in my notebook, and it sucess registry to Asterisk (show 1 online).

How to let the Asterisk server know the assigned Extension number for a Cisco 7061G IP phone ?

Thanks in advance.

I spent many days and hours trying to get a 7961 to register to asterisk using pjsip.conf. I finally changed the 'modules.conf ’ file to load ‘chan_sip.so’. I then configured sip.conf and it now registers. My conclusion is that the 7961 will not register using pjsip. Looking at the 7961 phone logs when it was failing it appears that the pjsip 401 response has too many characters for the 7961 to handle.

You don’t need an extension to register in Asterisk (FreePBX uses the term differently).

You need to provide the protocol log from Asterisk (full log, enabled, and with pjsip set logger on issued at the CLI (sip set debug on, if using the obsolete driver.).

We might also need the relevant parts of pjsip.conf and its includes (or sip.conf and its includes).

Make sure you either either markup logs an configurations as preformatted text, for the forum, or use a pastebin server, as the forum will, otherwise, garble them.

However attempting to use Cisco phones, reflashed for SIP, usually ends in tears. In particular, they do not handle any sort of NAT well.

Correction: I finally got two 7961s registered with pjsip thanks to a post where the following line has to be added in the pjsip.conf file under the endpoint location, fprce_rport=no. I am running asterisk 16 with SIP41.8.5-4S load.