I am on Asterisk certified/13.13-cert3 and running into this problem.
I have a dialplan like so:
[extensions] exten => _X.,1,noop same => n,ConfBridge(123)
I have a conference profile with these options. Basically enabling DTMF passthrough so everyone in the conference can hear them.
[default_user] type=user dtmf_passthrough=yes
I have two SIP soft phones set up for testing
[user1] type=friend host=dynamic username=user1 secret=password context=extensions host=dynamic qualify=yes [user2] type=friend username=user1 host=dynamic secret=password context=extensions qualify=yes
I then join both these users to the conference.
Now with AMI, when I try to mute one party:
Action: MuteAudio Channel: "SIP/user1-00000000" Direction: "all" State: "on"
I'm finding this works fine for spoken audio, but the DTMF digits are still being transmitted.
Is this a bug or by design? how can I prevent
SIP/user1 from hearing anything including the DTMF?
Thanks for any help.