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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Although if you dig through the archives you can find a good cross-section of AGI samples. Check the Asterisk Cookbook wikis as well.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>Sammy Govind<br><b>Sent:</b> Tuesday, November 01, 2011 9:08 AM<br><b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br><b>Subject:</b> Re: [asterisk-users] custom automated meeting<o:p></o:p></span></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal style='margin-bottom:12.0pt'>There is only one way to find out connecta.agi in /var/lib/asterisk/agi-bin directory by developing it yourself.<o:p></o:p></p><div><p class=MsoNormal>On Tue, Nov 1, 2011 at 6:57 PM, Thanasis <<a href="mailto:thanasis@asyr.hopto.org">thanasis@asyr.hopto.org</a>> wrote:<o:p></o:p></p><p class=MsoNormal>on 11/01/2011 03:25 PM Danny Nicholas wrote the following:<o:p></o:p></p><div><p class=MsoNormal style='margin-bottom:12.0pt'>> One way to do this (there are probably more and better ways). Incoming call<br>> to 123456789 launches meetme(1234,b(connecta.agi))<br>> Connecta.agi calls lines B and C and connects them to meetme(1234).<o:p></o:p></p></div><p class=MsoNormal>Thanks, but could you be more elaborate please?<br>Where can I find connecta.agi ?<o:p></o:p></p><div><div><p class=MsoNormal><br>><br>> -----Original Message-----<br>> From: <a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a><br>> [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>] On Behalf Of Thanasis<br>> Sent: Tuesday, November 01, 2011 1:58 AM<br>> To: Asterisk Users Mailing List - Non-Commercial Discussion<br>> Subject: Re: [asterisk-users] custom automated meeting<br>><br>> I just want to make two specific sip phone sets to ring together, when<br>> someone dials a specific incoming extension. And then, when each of the<br>> ringed sets answers, to be placed immediately into meeting session with the<br>> caller together with the other phone set.<br>><br>> Here is exactly what I mean:<br>><br>> Person A dials 123456789. Asterisk routes the incoming call and rings sip<br>> phones B and C. Person B answers phone B and starts talking with person A,<br>> while phone C keeps ringing. A minute later, and while A and B are still<br>> talking together, person C answers phone C, and starts talking with A and B<br>> together (that is aromatically all being placed in the same conference<br>> session).<br>><br>> Is that doable?<br><br>--<br>_____________________________________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>New to Asterisk? Join us for a live introductory webinar every Thurs:<br> <a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><o:p></o:p></p></div></div></div><p class=MsoNormal><o:p> </o:p></p></div></body></html>