[Asterisk-Dev] Help needed for custom application

[Marc Haisenko]

Hi folks,
I've been tasked with writing a very special application for Asterisk which is 
needed for a very special project (I have no idea whether I may give out 
details, so I'll stay on the safe side and won't). Since I still have 
difficulties in understanding the Asterisk internals I thought I could try 
asking the people who probably know the most about Asterisk :-)

Just so you don't get me wrong, I don't want you to develop that application 
for me, I just need some hints at how to do it.

The application I need to develop (let's call it fdial) can be described as a 
forking dial application (which only needs to support SIP). It should act 
almost like a normal dial at first, but when a special control frame arrives 
from a channel it should dial to a second address, and when the call is 
established the voice streams should be sent to both destinations (and vice 
versa, but not between the two destinations. So the voice stream is forking 
to two destinations. One of the established destination channels may then be 
hung up (obviously at least one destination channel must be up, otherwise we 
have a normal hangup).

I know it sounds weird :-)

I've looked into app_dial.c, but that code is doing way too much to easily 
understand what is necessary to do a call... and especially having two active 
destinations channel at a time is something that no other applications seems 
to do, not even MeetMe (MeetMe seems to do conferencing/voice stream mixing 
via a Zaptel device, doesn't it ?).

Has anyone some suggestions for me ? Is it even possible without really nasty 
hacks to Asterisk itself ? Does anyone know of an application that does 
something remotely like the fdial I need to implement ?

Thanks for your help in advance,
	Marc Haisenko
-- 
Marc Haisenko
Linux Solutions
Be O.K. service group GmbH

Rüdesheimer Straße 7
D-80686 München
Tel:   +49 (0)89 - 548 43 33 21
Fax:   +49 (0)89 - 548 43 33 29
e-mail: haisenko at be-ok.com
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