[asterisk-dev] Does Asterisk have any feature like this

[Dung, Nguyen Anh]

Hi All,

I'm diving into Asterisk source code to find the way to implement a feature
as described in the scenario below:

            1. A and B are on phone

            2. Asterisk calls C

            3. C answers and is able to get the RTP stream

            4. Asterisk connects A to C, drop B.

            5. A and C are on phone, the call duration of A is unchanged.

However, I have some questions:

1.       Does Asterisk have any feature like this before? As far as I know,
the closest feature is unattended transfer, but it still puts one side (B in
this case) on hold.

2.       I tried to use ast_bridge_call() to bridge A to C in the scenario
above, but I got a warning message from channel.c informed that they are
already bridged so that the bridging failed. Is there any suggestion for
doing this?

3.       Is conference code suitable for implementation of this feature? As
I know, when using conference call, we must dial to conference number (and
then Asterisk leads us to an available MeetMe room), it's not a call
through. The only good idea from conference call is it let us drop a channel
without having any impact on the existing call.

 

Thanks in advance.

Dung, Nguyen Anh.

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