Extension not registering - stumped

I am a novice contractor helping out with a site that implemented Trixbox and Asterisk and FreePBX. They use Cisco 7940 phones

I am also a novice at linux but I can find my way around the system. Although I don’t understand a lot of the configuration, most everything seems to be working and frankly I am afraid to tweak (look but don’t touch)

I have an extension setup in FreePBX 6531 which will simply not register. I can reconfigure the exact same phone for another extension and it reloads the SIP configuration and works fine, so it is not the physical phone -nor is it the associated MAC ID or the SIPxxxxxxxxxxxxxxxx.cnf file

I decided it must be something to do with the actual 6531 extension. Looking at various conf files, searching for the the 6531 string I found several exten ==> lines for 6531. I found only one anomaly.

My extension 5747 also has several exten lines but I have one which reads
exten ==> hint, SIP/5747. The same hint line for 6531 reads hint, custom/6531.

There is a warning saying not to modify this file directly as modifications should be done through asterisk. I don’t see anywhere in the FreeBBX extension creation screen where this is configured. There is dial: but both read SIP/5747 and SIP/6531 respectively. The FreePBX screens for my extension and the problem extension look nearly exactly alike (password, email config, etc…)

Lastly I was trying to telnet into the Cisco phone to try to decipher the
"E640 REG msg unsupported: in 404 request failure" -which other sites say means the phone could not register with SIP
Lots of Debug information but no clear reason why it timeouts or fails every 60 seconds.

So I don’t feel comfortable restarting any services/programs on the linux box, but I would really like to figure out why one extension works great and the other fails.

Thanks in advance


Is the extension a SIP type? Could you have chosen custom by accident?

If you do a ‘sip show peer 6521’ from Asterisk CLI do you get the peer info?

Problem solved - I looked for more SIP information on my system
There was a file called sip_additional.conf which has sip configuration blocks for all the extensions. But it was missing a block for the “bad” extension
Once that was entered in - the “bad” boy behaved.

Thanks for your input

I finally solved the problem by selecting another extension number for the problem phone. It is working perfectly now that I changed the actual extension number for the user. Some strange configuration error with the old extension I guess, anyway all is good now.

The company is probably replacing the entire phone system with Cisco next January. Still I am excited to learn more about Asterisk and SIP in general

Thanks for your comments

Hasta luego - TOM

sip_addional.conf is written by freepbx and should never have to be edited. It is created by the extensions and trunks module.

Any changes you made will be erased the next time you apply configurations in FreePBX.