Hi you can get some help using n-way dialplan example. Its generate new call and transfer current call in conference meetme. You can google to find its example On Nov 15, 2012 8:15 PM, "Michael" <voip.quest...@gmail.com> wrote:
> Hi Aldo, > > Thank you very much for answering my question. > > Can you kindly elaborate on how to do the following or at least where to > read about the way to do it? > >send both channels of the active call 111 - 22334455 to a context that > joins them in a conference room. > > >through AMI, I would originate the call to 22556677 and join it into the > conference. > > Thank you very much, > > Michael > > > On Thu, Nov 15, 2012 at 3:50 PM, Aldo Bergamini <aabe...@gmail.com> wrote: > >> 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 >> > > > -- > _____________________________________________________________________ > -- 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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