Core show channels verbose provides this information. Just grep for the channel you need to hit.
From: [email protected] [mailto:[email protected]] On Behalf Of [Digital^Dude] R Sent: Friday, March 30, 2012 7:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Bridging an Answered call in Asterisk with another call 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? On Thu, Mar 22, 2012 at 11:09 AM, Satish Barot <[email protected]> wrote: Jayesh, Personally I haven't worked on Congbridge :). Confbridge has evolved a lot in 10.X. So probably you should have no issues using it. On Thu, Mar 22, 2012 at 11:04 AM, Jayesh Nambiar <[email protected]> wrote: 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 !! I can definitely use the confbridge application for the same right; as I am working on Asterisk10. What do you suggest?? Thanks again, --- Jayesh On Thu, Mar 22, 2012 at 10:33 AM, Satish Barot <[email protected]> wrote: 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*. e.g. Assuming your destination is SIP extension, have something like... Action: Originate Channel: SIP/{your_destination_here} Application: MeetMe Data: {your_meetme_number_here} Hope this helps. --Satish Barot On Wed, Mar 21, 2012 at 9:06 PM, Jayesh Nambiar <[email protected]> wrote: Hello All, 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: 1) User dials 123 from a touch screen Polycom phone. 2) Call comes to Asterisk and Asterisk answers the call and asks for PIN number. 3) Once the PIN is validated, Asterisk sends a User Event through AMI which invokes a browser in the Polycom phone. 4) The Browser will have a Text-Box to Enter the destination number where the caller wants to be bridged. 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 !! 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. Is there any straightforward application or function to achieve this in Asterisk. Any ideas or directions will be of great help !! Thanks, --- Jayesh -- _____________________________________________________________________ -- 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 -- _____________________________________________________________________ -- 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
-- _____________________________________________________________________ -- 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
