Hello all, I’m facing a bug that I’ve never seen before. First of all I’ll explain a bit my scenario:
It’s a GoAutoDial v4 server, it uses Asterisk 13.17.2, Kamailio and WebRTC. The client uses a softphone in the browser via WebRTC connecting in the Kamailio, then the Kamailio has a SIP Peer to the asterisk server.
The Sip Debug is in the PasteBin attached to this post.
I’ve already changed the dtmfmode of all the peers involved on this exchange, rfc2833, inband and auto, but nothing has changed.
If there is anything more relevant to the issue let me know so I provide it here. Thanks in advance!