I’m using asterisk 1.2.18 with SIP protocol (and behind a NAT)
Problem occurs when, on an incoming call, the caller sends dtmf: the audio caller->callee stops.
This only happens on incoming calls, and ,this problem excepted, everything runs fine.
I searched a lot but couldn’t find any solution.
Thanks for any help !
PS: I can join any logs and/or conf files snippets if you tell me what would be revelant.