FreePBX V14, not registering Cisco 7960 phones which worked with earlier version

We recently upgraded to FreePBX V14 from FreePBX V13 and our Cisco 7960 phones stop working. Can someone please help?
You can see the detailed description with XML file at:

I use FreePBX 13 but to get those phones to work I have to SSH into the FreePBX system and go to /etc/asterisk and modify pjsip.endpoint_custom_post.conf
and add a stanza for each extension that says:

[extension#](+)  ; (there's brackets around extension# and it's the actual number not the word extension)

do you do this on version 14?

Did you set tcpenable=yes in Asterisk SIP settings?

EDIT: ooh…you use EPM…still…I would check if your extensions switched to pjsip…you might try to set them to sip…

another thing…use virtualbox for your freePBX server, so you can always return to a previous (working) state!

The Force rport paramater is a user settable option on the Advanced tab when editing pjsip extensions. There should be no need to set this manually.

[[email protected] ~]# asterisk -x "pjsip show endpoint 2003" | grep rport
 force_rport                        : true

Thanks Lorne but for whatever reason when I create an extension, that defaults to YES however the Cisco phones will not register unless I put it in endpoint_custom and reboot the PBX. In fact I just checked all my extensions and it’s set to yes even on the extensions that go to Polycom phones which supposedly don’t need it and do not have entries in endpoint_custom. This is FreePBX 13.

As I did not know this option was in Advanced if you like I will troubleshoot it further and verify that the option is indeed being ignored.

Another trick I have found (although the need for this may have been due to router firmware issues on my VPN router which have since been resolved) is that it helped stability to configure the Cisco phones as NAT and configure FreePBX to NOT list the remote subnet they are operating on as a Local subnet (thus configuring the PBX as NAT for those extensions) even though there was no actual address translation going in in between the PBX and the phones. Doing it the other way where the phone was configured as not nat, and the PBX was configured as local subnet, the phones would lose registration after a while.

I concur with the ALG thing but many routers even if they are doing ALG have no means for the admin to disable it. There’s a lot of garbage routers out there. But even if ALG is disabled the translation code still seems to have to do extra stuff to translate SIP.

With customer setups that are production I never set them up to run hard extensions through a translator although that does mean for the customer that even if they have a single remote office with ONE extension, that they will have to get a static IP for that office and install a VPN router. My preference in those cases is the Cisco ASA, as I have found that IPSec VPN’s between Cisco ASA devices appear almost bulletproof. I only run them through NAT when the extension is a soft phone on a smartphone or some similar mobile device.

This topic was automatically closed 14 days after the last reply. New replies are no longer allowed.