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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'>Core show channels verbose provides this information. Just grep for the channel you need to hit.<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>[Digital^Dude] ®<br><b>Sent:</b> Friday, March 30, 2012 7:45 AM<br><b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br><b>Subject:</b> Re: [asterisk-users] Bridging an Answered call in Asterisk with another call<o:p></o:p></span></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal style='margin-bottom:12.0pt'><span style='font-family:"Trebuchet MS","sans-serif"'>Be it meetme or confbridge, asterisk 1.2.x or Asterisk 10.x. Is it possible to query a channel and get its conference number in return?</span><o:p></o:p></p><div><p class=MsoNormal>On Thu, Mar 22, 2012 at 11:09 AM, Satish Barot <<a href="mailto:satish4asterisk@gmail.com" target="_blank">satish4asterisk@gmail.com</a>> wrote:<o:p></o:p></p><p class=MsoNormal>Jayesh, Personally I haven't worked on Congbridge :). <br>Confbridge has evolved a lot in 10.X. So probably you should have no issues using it.<o:p></o:p></p><div><div><p class=MsoNormal style='margin-bottom:12.0pt'><o:p> </o:p></p><div><p class=MsoNormal>On Thu, Mar 22, 2012 at 11:04 AM, Jayesh Nambiar <<a href="mailto:jayesh.voip@gmail.com" target="_blank">jayesh.voip@gmail.com</a>> wrote:<o:p></o:p></p><p class=MsoNormal>Thank you Satish. I was also thinking on similar lines. I was just wondering if there was any mechanism with which we can bridge a new call with the existing running call if the Call-ID of the call is known !!<br>I can definitely use the confbridge application for the same right; as I am working on Asterisk10. What do you suggest??<br><br>Thanks again,<br><br>--- Jayesh<o:p></o:p></p><div><div><p class=MsoNormal style='margin-bottom:12.0pt'><o:p> </o:p></p><div><p class=MsoNormal>On Thu, Mar 22, 2012 at 10:33 AM, Satish Barot <<a href="mailto:satish4asterisk@gmail.com" target="_blank">satish4asterisk@gmail.com</a>> wrote:<o:p></o:p></p><p class=MsoNormal style='margin-bottom:12.0pt'>Make your user wait in a *Meetme* and then call your destination number through AMI and once he answers, place him in the same *Meetme*.<br><br>e.g. Assuming your destination is SIP extension, have something like...<br><br>Action: Originate<br>Channel: SIP/{your_destination_here}<br>Application: MeetMe<br>Data: {your_meetme_number_here}<br><br>Hope this helps. <br>--Satish Barot<o:p></o:p></p><div><div><div><p class=MsoNormal>On Wed, Mar 21, 2012 at 9:06 PM, Jayesh Nambiar <<a href="mailto:jayesh.voip@gmail.com" target="_blank">jayesh.voip@gmail.com</a>> wrote:<o:p></o:p></p></div></div><blockquote style='border:none;border-left:solid #CCCCCC 1.0pt;padding:0in 0in 0in 6.0pt;margin-left:4.8pt;margin-right:0in'><div><div><p class=MsoNormal style='margin-bottom:12.0pt'>Hello All,<br>I need to know a way of connecting an Answered call in Asterisk to another call which was triggered by an AMI. I have a scenario as follows:<br>1) User dials 123 from a touch screen Polycom phone.<br>2) Call comes to Asterisk and Asterisk answers the call and asks for PIN number.<br>3) Once the PIN is validated, Asterisk sends a User Event through AMI which invokes a browser in the Polycom phone.<br>4) The Browser will have a Text-Box to Enter the destination number where the caller wants to be bridged.<br>5) The caller enters this number in the browser which is sent as a Originate command to Asterisk through the AMI. Please note Asterisk does not get this number as DTMF events !!<br>6) Now, I need to BRIDGE this originated call from the AMI with the actual caller who is already present in Answered state in Asterisk probably listening to some music.<br><br>Is there any straightforward application or function to achieve this in Asterisk.<br><br>Any ideas or directions will be of great help !!<br><br>Thanks,<br><br>--- Jayesh<br><br><o:p></o:p></p></div></div><p class=MsoNormal><span style='color:#888888'>--<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></span><o:p></o:p></p></blockquote></div><p class=MsoNormal><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><p class=MsoNormal><o:p> </o:p></p></div></div><p class=MsoNormal><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><p class=MsoNormal><o:p> </o:p></p></div></div><p class=MsoNormal><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><p class=MsoNormal><o:p> </o:p></p></div></body></html>