I have an existing FreePBX system, v18.104.22.168. Set up a Sangoma S500 phone with VPN. Took the phone and it connects up and can dial and be dialed from the PBX. The problem is that when I dial another extension, the call disconnects after 32 seconds. I checked ALG on my SonicWall router, and disabled SIP Transformations, and enabled Consistant NAT. Ports are forwarded as recommended. Any other ideas out there? All other phones work great. Have two Vega Gateways on the network and 20 Sangoma phones, and 7 other PBXs tied together with IAX trunks. Other PBXs are a mix of FreePBX v13, v14 and v15.
Are you sure it is effectively connecting over the VPN? 30 seconds disconnection is almost always indication of NAT issues.
Thanks for the reply. In System Admin/VPN Server the phone is showing up in the “Clients” tab. In Asterisk Info/Peers/Chan_PjSIP Endpoints, the extension shows up as status “Avail”. It looks normal from what I can see on those screens.
Confirm that the tunnel IP address range is listed correctly in Asterisk SIP Settings -> Local Networks. If you change this you must restart (not just reload) Asterisk.
Confirm that the phone does not have any NAT settings enabled.
If you still have trouble, post a SIP trace from the PBX end.
I got it to work! I had a typo in the address in my Sangoma Template. I fixed the address typo, did a Save and Reload of the configuration, and it worked perfectly.
I needed a fresh set of eyes to look at it.
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