Manu,
Perhaps it is possible to do this by running the call through an RTP
proxy (there are various) that supports inserts muxing outside audio
feeds into the media stream? Then you would just need to implement
some sort of middleware layer to map Asterisk SIP channels to corresponding
call flows in the RTP proxy and run outside commands to make this happen, I
assume.
That said, I can't find any information specifically on this subject from
a given source. But try looking at various open-source RTP proxies.
At present, it appears Asterisk only has the capability to "barge"
into a
channel on the software layer (or Zaptel kernelspace, in the case of
ZapBarge()) for the purpose of listening in.
-- Alex
On Fri, 27 Apr 2007, Manu Mehta said something to this effect:
> Hi,
>
> Is it possible to host call waiting service on Asterisk for a SIP device?
> What i am trying to achieve is that while a SIP user is busy on a call and
> a new call for that user comes in, asterisk should play the call waiting
> tone to that user.
> I have a vague idea that if i can get hold of the existing bridged channel
> when a subsequent call is received, i can then redirect that channel to
> play tone.
> The problem is how can i get hold of the bridged channel in the first
> place? Also is there a better way of accomplish call waiting.
>
> TIA,
>
> Manu Mehta
>
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