FreePBX 14 one-way audio over OpenVPN connection with Grandstream

We recently upgraded some FreePBX servers from version 12 to 14. Some remote Grandstream phones using an OpenVPN connection suddenly stopped working by not passing along audio to the remote phones in both intercom and external calls. The NAT settings are correct as all other phones work just fine with both internal and external calls. Interestingly enough, the same phones and configuration works with FreePBX 12, just not 14. Also, using an OpenVPN connection on a PC with the X-Lite softphone by CounterPath also works fine with FreePBX 14. The OpenVPN local network is listed in “Asterisk SIP Settings” for NAT’d connections to make external calls (even though in this case we get no audio back to the remote phone for internal calls either.) Does anyone know what might have changed in FreePBX 14 that would prevent the Grandstream GXP21xx phones from working remotely over an OpenVPN connection?

EDIT – I have some additional information from packet captures on FreePBX 14. It seems that the Grandstream phones put the ethernet IP address into the SIP information instead of the OpenVPN connection IP address unlike the X-Lite softphone that uses the correct one. That OpenVPN IP address is only found in the TCP packet headers for the Grandstream phone communications. I am wondering if Asterisk 13 used in FreePBX 14 gets the originating IP address from the SIP information whereas Asterisk 11 in FreePBX 12 just uses the one in the TCP headers.

Sounds like a Grandstream bug. I’m hoping that you can work around it by setting it up as a chan_sip extension and setting NAT Mode for the extenstion to Yes (force_rport, comedia).

If no luck, post a SIP trace.

Excellent! That was the issue. FreePBX 14 sets the NAT Mode to “No” by default whereas FreePBX 12 sets it to what you indicated as the default when creating an extension.

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