[asterisk-dev] one way call bridge

[Tomáš Vrána]

Hi,

I am creating a special app based on asterisk and I need to separate
rx/tx strams in two different channels in order to eliminate possible
audio loops & feedback. 

I was thinking of using ChanSpy / whisper functionality to do that, but
I can't get it to work. Perhaps someone could tell me what the likely
problem is .... My setup is:

I have two SIP channels, extensions 103 and 502. Both extensions have
established calls to different meetme conferences.

I have these extensions set up:

[spy]
exten => 7502,1,ChanSpy(SIP/502,q) 
exten => 7103,1,ChanSpy(SIP/103,Wq)

The I try to originate call, which would spy on one channel (502) and
whisper the stream into the other one (103)

channel originate Local/7103 at spy extension 7502 at spy

Call is established, I get notice about starting to spy, but no sound
gets through... is that a bug or am I doing something wrong ? Would it
help to break that in two calls and join the with meetme conference ? Or
is there any easier way to do what I need ? 

I have also looked quickly through the bridging code, would it be
possible to hack something right there ? If yes, would someone
knowledgeable be so kind and point me in the right direction, pleas.
Before you start yelling at me, that my question doesn't belong here
please be advised that I have already asked the asterisk user audience
and I am posting here in order to get some pointers from people with
more insight... 

Thanks, Tom

-- 
 Tomas Vrana  <tomas.vrana at cemotel.cz>
 -------------------------------------
 Ceskomoravska telekomunikacni s.r.o.
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