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

[Nicholas Blasgen]

You may be able to use the Asterisk Manager interface to watch the
channels.  When C calls in, Asterisk could hangup on B and transfer to C.
Some mixture of AGI or the normal Dialplan (for after B is droped) and the
Manager (for handling control while in a call) should work.  And mix
everything together with a database somewhere so the manager and the
dialplan/AGI can exchange state information.

On 8/8/07, Dung, Nguyen Anh <nadung at tma.com.vn> wrote:
>
>  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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-- 
/Nick
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