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 <[email protected]> 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: [email protected] > > [mailto:[email protected]] 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 >
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