WebRTC Disconnect Status when brought up

I am running Asterisk 11.8.1 and FreePBX 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.

You need to bring up your javascript console in your browser to see what the error is

I’ll have to wait til I’m on my home network, just to make sure things are as simple as possible. Not being a Javascript person, I don’t know exactly what to look for in the log, but I will duplicate the problem and post the results.

This is a bug in the browser and Asterisk with WebRTC and they have no clue when they may fix it.

Is there some sort of Bug Report for this problem that I can look at and track?