Re: [asterisk-users] issue with bridgeConference
> On Mon, Nov 2, 2015 at 3:16 PM, hadi 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. Hi Rusty, I solved my problem. Thank you for your support. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] issue with bridgeConference
On Mon, Nov 2, 2015 at 3:16 PM, hadi 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] issue with bridgeConference
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}) Regards -Hadi.Salem -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users