I am running Asterisk 11.8.1 and FreePBX 2.11.0.37. Basically, it’s the FreePBX distro.
I enabled WebRTC on my extension per instructions. I go to the User Portal for that extension and bring up the WebRTC Phone and the status sometimes goes to “Registered to SIP Server” briefly (10 seconds) but then it switches to “Disconnected”. I does not do anything when a call comes in on my extension.
I don’t see anything in the Log related to WebRTC (perhaps I am looking at the wrong file or don’t have correct log options).
I don’t use this feature yet (obviously), but think it’s a cool new feature.
Any idea what the problem could be.
FWIW, the phone extension is a Digium D40.