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almarzuki2011 at hotma...
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 Posted: Mon Nov 02, 2015 4:17 pm Post subject: [asterisk-users] issue with bridgeConference |
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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
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rnewton at digium.com
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| Posted: Fri Nov 06, 2015 8:15 am Post subject: [asterisk-users] issue with bridgeConference |
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On Mon, Nov 2, 2015 at 3:16 PM, hadi <almarzuki2011@hotmail.com> wrote:
| Quote: | 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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_____________________________________________________________________
-- 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:
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almarzuki2011 at hotma...
Guest
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| Posted: Fri Nov 06, 2015 2:39 pm Post subject: [asterisk-users] issue with bridgeConference |
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| Quote: | On Mon, Nov 2, 2015 at 3:16 PM, hadi <almarzuki2011@hotmail.com> wrote:
| Quote: | 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-${DIALSTATUS},1) exten => s-NOANSWER,1,Hangup
exten => s-CONGESTION,1,Congestion exten => s-CANCEL,1, Busy exten
|
=>
| Quote: | s-BUSY,1,Busy exten => s-CHANUNAVAIL,1,Playback(switchoff)
exten => s-CHANUNAVAIL,n,Read(number,,,sn) exten =>
s-CHANUNAVAIL,n,GotoIf($["${number}" = "9"]?106) exten =>
s-CHANUNAVAIL,106,SoftHangup(${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 |
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