WebRTC browser calls FreePBX 13 (Stable)

I have the same problem. WebRTC extension connects via websocket and the sip "extension" is reachable according to sip show peers on the asterisk cli. When an incoming call comes in:

-- SIP/995051-0000000c is ringing
-- Redirecting update to SIP/IVS2-0000000b prevented. ( <-- this is my trunk)
-- SIP/995051-0000000c is busy

FreePBX 13.0.35
WebRTC module - 13.0.9
Asterisk - 13.6.0