Hi. I hope someone can help. I’ve spent days trying to get this to work.
I started with a clean Debian 12 install and clean FreePBX on a Linode(Akamai) VPS with no firewall other than the Sangoma within FPBX and a public IP address. I configured the trunk to Clone Line / VOISIP using the pjsip module per normal. I configured outbound routes and inbound routes.
I setup a GXP2160 Grandstream phone with a configured OpenVPN to the server. The server has 10.8.0.1 & the phone 10.8.0.2.
The very first time I dialed out, it dialed and has not done so since. It also will not ring on a dial in, just disconnects immediately.
From sngrep I can see the invite (SDP), a 401 unauthorized response, followed by an ACK. The Invite is being sent sip:5552121212@10.8.0.1 (openvpn server IP).
And that’s all it does until it eventually releases the line.
I have assumed that the use of OpenVPN should circumvent any natting and double natting issues (it does for everything else I use it for) and the FreePBX should treat the OpenVPN network as local (it looks that way in the configuration) and should use the assigned trunk route per the outbound routes.
It did just once when I first set it up. Now nothing. I’ve tried about everything I can think of at this point.
Any suggestions will be gratefully accepted.
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