Re: [asterisk-users] custom automated meeting
on 11/02/2011 07:44 AM Sammy Govind wrote the following: core show application meetme Thanks! (I am new to asterisk, and just learning, so forgive my dumb questions) -- _ -- 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] custom automated meeting
I just want to make two specific sip phone sets to ring together, when someone dials a specific incoming extension. And then, when each of the ringed sets answers, to be placed immediately into meeting session with the caller together with the other phone set. Here is exactly what I mean: Person A dials 123456789. Asterisk routes the incoming call and rings sip phones B and C. Person B answers phone B and starts talking with person A, while phone C keeps ringing. A minute later, and while A and B are still talking together, person C answers phone C, and starts talking with A and B together (that is aromatically all being placed in the same conference session). Is that doable? -- _ -- 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] custom automated meeting
One way to do this (there are probably more and better ways). Incoming call to 123456789 launches meetme(1234,b(connecta.agi)) Connecta.agi calls lines B and C and connects them to meetme(1234). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thanasis Sent: Tuesday, November 01, 2011 1:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] custom automated meeting I just want to make two specific sip phone sets to ring together, when someone dials a specific incoming extension. And then, when each of the ringed sets answers, to be placed immediately into meeting session with the caller together with the other phone set. Here is exactly what I mean: Person A dials 123456789. Asterisk routes the incoming call and rings sip phones B and C. Person B answers phone B and starts talking with person A, while phone C keeps ringing. A minute later, and while A and B are still talking together, person C answers phone C, and starts talking with A and B together (that is aromatically all being placed in the same conference session). Is that doable? -- _ -- 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 -- _ -- 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] custom automated meeting
on 11/01/2011 03:25 PM Danny Nicholas wrote the following: One way to do this (there are probably more and better ways). Incoming call to 123456789 launches meetme(1234,b(connecta.agi)) Connecta.agi calls lines B and C and connects them to meetme(1234). Thanks, but could you be more elaborate please? Where can I find connecta.agi ? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thanasis Sent: Tuesday, November 01, 2011 1:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] custom automated meeting I just want to make two specific sip phone sets to ring together, when someone dials a specific incoming extension. And then, when each of the ringed sets answers, to be placed immediately into meeting session with the caller together with the other phone set. Here is exactly what I mean: Person A dials 123456789. Asterisk routes the incoming call and rings sip phones B and C. Person B answers phone B and starts talking with person A, while phone C keeps ringing. A minute later, and while A and B are still talking together, person C answers phone C, and starts talking with A and B together (that is aromatically all being placed in the same conference session). Is that doable? -- _ -- 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] custom automated meeting
There is only one way to find out connecta.agi in /var/lib/asterisk/agi-bin directory by developing it yourself. On Tue, Nov 1, 2011 at 6:57 PM, Thanasis thana...@asyr.hopto.org wrote: on 11/01/2011 03:25 PM Danny Nicholas wrote the following: One way to do this (there are probably more and better ways). Incoming call to 123456789 launches meetme(1234,b(connecta.agi)) Connecta.agi calls lines B and C and connects them to meetme(1234). Thanks, but could you be more elaborate please? Where can I find connecta.agi ? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thanasis Sent: Tuesday, November 01, 2011 1:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] custom automated meeting I just want to make two specific sip phone sets to ring together, when someone dials a specific incoming extension. And then, when each of the ringed sets answers, to be placed immediately into meeting session with the caller together with the other phone set. Here is exactly what I mean: Person A dials 123456789. Asterisk routes the incoming call and rings sip phones B and C. Person B answers phone B and starts talking with person A, while phone C keeps ringing. A minute later, and while A and B are still talking together, person C answers phone C, and starts talking with A and B together (that is aromatically all being placed in the same conference session). Is that doable? -- _ -- 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 -- _ -- 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] custom automated meeting
Although if you dig through the archives you can find a good cross-section of AGI samples. Check the Asterisk Cookbook wikis as well. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sammy Govind Sent: Tuesday, November 01, 2011 9:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] custom automated meeting There is only one way to find out connecta.agi in /var/lib/asterisk/agi-bin directory by developing it yourself. On Tue, Nov 1, 2011 at 6:57 PM, Thanasis thana...@asyr.hopto.org wrote: on 11/01/2011 03:25 PM Danny Nicholas wrote the following: One way to do this (there are probably more and better ways). Incoming call to 123456789 launches meetme(1234,b(connecta.agi)) Connecta.agi calls lines B and C and connects them to meetme(1234). Thanks, but could you be more elaborate please? Where can I find connecta.agi ? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thanasis Sent: Tuesday, November 01, 2011 1:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] custom automated meeting I just want to make two specific sip phone sets to ring together, when someone dials a specific incoming extension. And then, when each of the ringed sets answers, to be placed immediately into meeting session with the caller together with the other phone set. Here is exactly what I mean: Person A dials 123456789. Asterisk routes the incoming call and rings sip phones B and C. Person B answers phone B and starts talking with person A, while phone C keeps ringing. A minute later, and while A and B are still talking together, person C answers phone C, and starts talking with A and B together (that is aromatically all being placed in the same conference session). Is that doable? -- _ -- 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 -- _ -- 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] custom automated meeting
You need simple dialplan of four steps:
same =n,Set(conf_name=conf-${RAND(1,1000)})
same =n,Originate(SIP/dev1,app,MeetMe,${conf_name},dFI1x)
same =n,Originate(SIP/dev2,app,MeetMe,${conf_name},dFI1x)
same =n,MeetMe(${conf_name},dFI1xAC)
same =n,Noop(do post conference stuff)
2011/10/31 Thanasis thana...@asyr.hopto.org:
I need your help in implementing the following scenario:
A certain extension will ring two sip phones simultaneously and when one
of them answers, the other keeps ringing until it answers too, and then
all three (the caller and the other two) are immediately placed in a
conference room (same room for all three).
Can we do it?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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http://www.asterisk.org/hello
asterisk-users mailing list
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_
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Re: [asterisk-users] custom automated meeting
on 11/01/2011 05:41 PM Yaroslav Panych wrote the following:
You need simple dialplan of four steps:
same =n,Set(conf_name=conf-${RAND(1,1000)})
same =n,Originate(SIP/dev1,app,MeetMe,${conf_name},dFI1x)
same =n,Originate(SIP/dev2,app,MeetMe,${conf_name},dFI1x)
same =n,MeetMe(${conf_name},dFI1xAC)
same =n,Noop(do post conference stuff)
Thanks!
What is the meaning of the options dFI1xAC passed to
app,MeetMe,${conf_name} ?
Where can I find them described please?
2011/10/31 Thanasis thana...@asyr.hopto.org:
I need your help in implementing the following scenario:
A certain extension will ring two sip phones simultaneously and when one
of them answers, the other keeps ringing until it answers too, and then
all three (the caller and the other two) are immediately placed in a
conference room (same room for all three).
Can we do it?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] custom automated meeting
Type in asterisk CLIcore show application meetme
or google asterisk cmd meetme simple?
On Tue, Nov 1, 2011 at 10:33 PM, Thanasis thana...@asyr.hopto.org wrote:
on 11/01/2011 05:41 PM Yaroslav Panych wrote the following:
You need simple dialplan of four steps:
same =n,Set(conf_name=conf-${RAND(1,1000)})
same =n,Originate(SIP/dev1,app,MeetMe,${conf_name},dFI1x)
same =n,Originate(SIP/dev2,app,MeetMe,${conf_name},dFI1x)
same =n,MeetMe(${conf_name},dFI1xAC)
same =n,Noop(do post conference stuff)
Thanks!
What is the meaning of the options dFI1xAC passed to
app,MeetMe,${conf_name} ?
Where can I find them described please?
2011/10/31 Thanasis thana...@asyr.hopto.org:
I need your help in implementing the following scenario:
A certain extension will ring two sip phones simultaneously and when one
of them answers, the other keeps ringing until it answers too, and then
all three (the caller and the other two) are immediately placed in a
conference room (same room for all three).
Can we do it?
--
_
-- 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
--
_
-- 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
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Re: [asterisk-users] custom automated meeting
Hello, look at Page application Regards - Original Message - From: Thanasis thana...@asyr.hopto.org To: asterisk-users@lists.digium.com Sent: Monday, October 31, 2011 4:59 PM Subject: [asterisk-users] custom automated meeting I need your help in implementing the following scenario: A certain extension will ring two sip phones simultaneously and when one of them answers, the other keeps ringing until it answers too, and then all three (the caller and the other two) are immediately placed in a conference room (same room for all three). Can we do it? -- _ -- 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 -- _ -- 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
