On 15 Nov 2012, at 14:21, Michael wrote: > Hello, > > Does anyone know if it's possible to setup the following scenario? > > 1. A specific ext(let's say 111) is on active call with an external number > via SIP (let's say 22334455). > 2. Via a web GUI, send to asterisk another phone number (22556677) and the > ext number (111). > 3. Asterisk initiates a call to that number (22556677) and joins it to the > call in progress (between 111 and 22334455) in order to establish a 3-party > conf call. > > It's somewhat similar to ChanSpy, but with full conf capabilities and not > only whisper to one side. > > Thanks, > > Michael >
Hi Michael, I would use a combination of AMI & dialplan programming. Over AMI I would send both channels of the active call 111 - 22334455 to a context that joins them in a conference room. It is a matter of choice if it is better to create an ad hoc/ on the fly conference or use a set of predefined rooms. Next, again through AMI, I would originate the call to 22556677 and join it into the conference. You have to be aware that calling somebody and transferring the channel into a conference may leave the person on the other side of the wire WITHOUT means to exit the conference room and thus to close the call (I did it!!! embarrassing..). So one has to be sure (I am speaking of the old MeetMe app) that the "originator's" channel enters the conference room as the conference master. So, when that channel closes, all other channels are dumped out of the conference room and the whole thing closes down. HTH, Aldo -- _____________________________________________________________________ -- 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
