WebRTC calls failing

1

Asterisk (Ver. 11.9.0)
FreePBX 2.11.0.38

Calls to an extension with WebRTC phone enabled fail.

When logging into the user panel, the browser uses websocket with the following SIP Register info. X.X.X.X = asterisk server IP

REGISTER sip:X.X.X.X SIP/2.0\r\n
Via: SIP/2.0/WS l8rjaoqq2mn1.invalid;branch=z9hG4bK7826469\r\n
Max-Forwards: 69\r\n
To: sip:[email protected]\r\n
From: sip:[email protected];tag=rsg82pli58\r\n

The INVITE getting sent to the browser contains the following Req-URI. The browser interface never shows an incoming call.
INVITE sip:[email protected];transport=ws SIP/2.0\r\n
Via: SIP/2.0/WS X.X.X.X:0;branch=z9hG4bK271eaabe;rport\r\n
To: sip:[email protected];transport=ws;tag=4ejhkh6pci\r\n
Max-Forwards: 70\r\n

Then the browser sends back a 488.

SIP/2.0 488 Not Acceptable Here\r\n
Via: SIP/2.0/WS X.X.X.X:0;branch=z9hG4bK271eaabe;rport\r\n
To: sip:[email protected];transport=ws;tag=4ejhkh6pci\r\n

2

You will not get this to work in 2.11. It’s broken forever. It is fixed in 12 however.

This is why I removed it from 2.11 a few weeks ago

3

What’s the best procedure for going from 2.11 to 12.0.1 Beta?

4