On 15 Nov 2012, at 15:44, Michael 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?
Hi Michael, sure... I am sending you -by direct mail- a diagram that tries to illustrate what I would try to do. (I do not know if this list allows attachments; generally it's not permitted...). > >send both channels of the active call 111 - 22334455 to a context that joins > >them in a conference room. AMI has a useful command for that task: Redirect, see here: http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Redirect and here: http://www.voip-info.org/wiki/view/Asterisk+manager+Example:+Transfer If you are manipulating a call not from one of the connected terminals (e.g. your phone) you have to take care of both channels. This is what the Redirect command does. It lets you specify what to do with both channels: they can be sent to the same context or each to one context by itself. Finally you are able to make changes on one channel only... > >through AMI, I would originate the call to 22556677 and join it into the > >conference. So the plan would be to first send the two channels to a conference room (an ad hoc one), using a first redirect command. This is made to get the conf. room where the three way call will take place AND to be able to call the third party without losing the original call partner's channel... A second Redirect command should detach the user's channel from the conference and send it to a context that connects him/her to the third party, letting the original user offer the 3 way call. If the call is accepted, than a third redirect would send both channels to the conference room created at step 1, where the other party is waiting... The dynamic conference is closed either by the original call party hanging up his/her channel or with a direct AMI hangup command doing the same thing. Clearly this is logically equivalent to a manual transfer of the user's call party into a conference room. Then calling the second call party and transfer him/her to the conference and seeing the user finally dialing him/herself into the conference. You can do that with AMI, provided you have some means to make some sort of UI for the whole process... > Thank you very much, > > Michael You're welcome: 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
