[asterisk-dev] Manipulating audio in asterisk

[Slawek Sloma]

Hello All,

i am a student and new to asterisk. What i am trying to do is to implement
a steganographic method to hide data into the audio of a PSTN call.
To do that, on one end of the call I am trying to modify the audio that is
sent and on
the other end of a call I am trying to retrieve that data.

I have wrote a dialplan function (based on func_volume.c) that does this
using audiohooks.
Unfortunately i think that i don't understand how audiohooks work because I
think I can't
get only the audio from one end point of a call, modify it and send to
another end point - I am
modifying how I understand it the "audio in a channel" - that is mixed audio
from both end
points of a call. I hope you will understand me ;)


Here is a diagram what I have at home:

SIP-UA1 -> AST1 -> PSTN -> AST2 -> SIP-UA2

The call is between sip-ua1 and sip-ua2 - I would like to inject information
in
one asterisk server and retrieve them in another


Thanks,
Sławomir Słoma
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