[asterisk-dev] Setting the context in a SIP channel

[Marc Haisenko]

On Friday 27 January 2006 18:34, Tilghman Lesher wrote:
> On Friday 27 January 2006 11:20, Marc Haisenko wrote:
> > The big problem is that I need to mix audio, so I can't just bridge
> > two channels as for a short (or not so short) period of time three
> > parties are involved... if this requirement wouldn't exist I'd had a
> > lot easier time and really could use all the stuff Asterisk already
> > provides :-) That's why I started off of app_conference.
>
> In what situation are you mixing three legs together?  Seems like
> the situation you've already described is separate from mixing three
> call legs together.  Isn't it the case that once you switch over the
> call, you want to completely replace one leg with another?

Yes, but for a short amount of time sound should be received/sent from both 
legs while the handover is in progress to avoid sound gaps and other stuff.

I also have the problem of handling custom SIP INFO messages to initiate the 
handover, so just using a normal bridge wouldn't work anyway, as far as I 
see.

C'ya,
	Marc

-- 
Marc Haisenko
Comdasys AG

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e-mail: haisenko at comdasys.com
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