[asterisk-users] issue with bridgeConference

Rusty Newton rnewton at digium.com
Fri Nov 6 07:13:00 CST 2015


On Mon, Nov 2, 2015 at 3:16 PM, hadi <almarzuki2011 at hotmail.com> wrote:
> I have configure bridgeConference. But im having some issue. I want to give
> the ability to the user when dialing from the Conference to hangup the call
> by sending dtmf tones without being hangup from the Conference. For example
> if the user call some person and that person not answering, the user has the
> ability to hangup the call by sending *9 and return back the Conference, and
> start calling again.
>
> Here is my dial plan:-
>
> exten => 200,1,Dial(SIP/200,,Hhg)
> exten => 200,n,Goto(s-${DIALSTATUS},1)
> exten => s-NOANSWER,1,Hangup
> exten => s-CONGESTION,1,Congestion
> exten => s-CANCEL,1, Busy
> exten => s-BUSY,1,Busy
> exten => s-CHANUNAVAIL,1,Playback(switchoff)
> exten => s-CHANUNAVAIL,n,Read(number,,,sn)
> exten => s-CHANUNAVAIL,n,GotoIf($["${number}" = "9"]?106)
> exten => s-CHANUNAVAIL,106,SoftHangup(${EXTEN})

I suppose by bridgeConference you mean ConfBridge?

If you require assistance you'll need to describe more than what you
*want to do*. You'll need to describe the issue you are having.
Include dialplan and logs to demonstrate the issue.

-- 
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200

Check us out at: http://digium.com & http://asterisk.org



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