Asterisk not receiving DTMF 0


#1

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.
My problem is only when sending the DTMF 0 (zero), all other DTMF’s works perfectly. I’ve already checked the JavaScript functions who sends the dtmf, the SIP traffic using sngrep and Sip Debug in console of Asterisk and the only thing different is that when it’s the DTMF 0, in the console it prints only 8 lines about this SIP Signaling, and when it is any other DTMF it prints 13 lines.
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.

https://pastebin.com/tM7uHSBt

If there is anything more relevant to the issue let me know so I provide it here. Thanks in advance!